id title reseller_price price brand product_type link image_link condition expiration_date description 29 Snom Power Supply 12.5000 12.50 Snom VoIP Accessories http://www.voipon.co.uk/snom-power-supply-p-29.html http://www.voipon.co.uk/images/no_image_available.jpg new Availability: In Stock Snom 190/300/320/360/370 power supplies 31 Snom 360 IP Telephone 134.0000 147.00 Snom Snom IP Telephones http://www.voipon.co.uk/snom-360-ip-telephone-p-31.html http://www.voipon.co.uk/images/snom_360_sml.jpg new Availability: In Stock The snom 360 has been designed for the highest levels of efficiency and productivity in the business environment. The 360's dedicated keys provide users with direct access to functions for audio & call control whilst context-sensitive menus offer the additional functionality that may be needed at any given moment. The display can be tilted for the most user-friendly reading angle. The snom 360 mini browser The snom mini browser is embedded in this top-of-the-range 360 executive SIP phone; it allows users and developers to create web-driven, screen-based telephone applications. Examples include custom contact-center apps, web-based phone directories, messaging, posting of news and other info on telephone screens, and more. Voice over the Internet Protocol (VoIP) has finally made it. Today we have solutions that fulfill the requirements of modern businesses and offer new services known from traditional telephony solutions. Declaration of Independence Choosing the Session Initial Protocol (SIP) is your personal declaration of independence. SIP is like HTTP a simple, pragmatic protocol for establishing sessions specified by the Internet Engineering Task Force (IETF). SIP has been chosen by most vendors as the communication protocol of the future. SIP components can be combined into a complete system without making you dependent on a single vendor. This declaration of independence should not be underestimated. It already changed the mainframe world into the IT landscape as we know it today telephony is next on the list. Don't miss out on it by building on a dinosaur business model. Making Your Work Easier Labor cost is the most expensive production factor in many companies. In these environments, simple and fast access to the most common Snom 360 functions increases the productivity and gives your investment in telephony infrastructure a positive ROI. CONNECTORS: Network: RJ45 (Ethernet) PC: RJ45 (Ethernet) Power: 5 V DC (stabilized) 2 port switch included (802.3 10/100 BT half duplex/full duplex with autosense) Power over LAN (IEEE 802.3af) on network port Handset: RJ14 Standard HandsetConnector Headset: RJ-Connector Headset Extension Board: Proprietary snom connector USER INTERFACE Display: Graphical 128 x 64 pixels 47 keys, 13 LEDs Last calls (100 entries) Address book (100 entries) Address book Import/Export Number guessing, speed dialing Missed calls, dialed calls Call waiting indication Clock, daylight saving, call-timer Call blocking (Deny List) Programmable function keys Menu-driven user interface Selectable ringing melodies National language support for selected languages (NLS) Asian language support (ALS) Downloadable ringing melodies URL Dialing support Do not disturb Speakerphone (Full Duplex) Auto answer mode UTF8-encoded Caller-ID (ALS) Multi-Line registration Snom 360 SECURITY https (server/client) SIPS SRTP Certificates can be loaded into devic CALL FEATURES Hold Blind transfer, Attended transfer Music on hold support Divert Call intrusion Conferencing (3-way conference bridge on phone) Call park, Call pick-up Call completion CMC (Client Matter Code) WEB SERVER Embedded web server Easy configuration of the phone, remote configuration Dial from web interface Password protection Diagnostics (tracing, logging) CODECS G.711 aLaw, uLaw G.729A, G.723.1, GSM 6.10 (Full rate) G.722 (16 kHz) SIP RFC3261 compliant UDP, TCP, TLS support Digest authentication Loose routing and strict routing support Error-information support within Snom 360 Reliability of provisional responses (RFC3262) DNS SRV (RFC3263), redundant server support Offer/answer (RFC3264) Message waiting indication reception (RFC3842), subscription for MWI events (RFC3265) Dialog-state In-band DTMF/Out-of-band DTMF (RFC2833) STUN client (NAT traversal) ENUM (RFC3261) NAPTR (RFC2915) rport (RFC3581) REFER (RFC3515) Many other SIP features INSTALLATION Static IP provisioning, DHCP http client for configuration Automatic software update Completely automatic installation from web Snom IP Phone Comparison Table Phone Power over ethernet Display SIP Identities Programmable Keys with LEDs Large LED for incoming calls Extension Keyboard Available Wireless headset adaptor Snom 300 Y 2 Line 16 Characters 4 6 N N N Snom 320 Y Hinged 2 line character display with graphical field 12 12 N Y Y Snom 360 Y Hinged, backlit graphical display (128x64 pixels) 12 12 N Y Y Snom 370 Y Hinged, backlit, high-definition graphical display (240x128 pixels) 12 12 Y Y Y 32 Elmeg IP290 IP Telephone 88.0000 90.00 Elmeg Elmeg IP Telephones http://www.voipon.co.uk/elmeg-ip290-ip-telephone-p-32.html http://www.voipon.co.uk/images/elmeg_ip290.jpg new Availability: Discontinued  For a high quality similiar alternative please see the Snom 300 The Elmeg IP290 encompasses the important aspects of demanding business customers using the latest VoIP technology. Fitted with a two-line graphical display, the IP290 supports alphanumeric caller ID and a feature-rich user interface. An extra Ethernet port for the enables users to "daisy chain" the phone with their PCs without the necessity for an extra LAN socket. For hands-free operation a standard PC headset can be used. Using SIP, the IP290 interoperates with equipment with standard compliance from other vendors. Advanced features like call pickup and parking or automatic call-back are implemented using latest standards. The programmable keys make the IP290 quick and easy to find numbers and initiate calls. Support for SIPS and SRTP allow voice conversations on a very high security level. The various support avenues of NAT standards allows the operation in all environments where VoIP is possible. Features Dual Ethernet port (network/PC) 3 softkeys and 5 programmable function keys Headset connection Hands-free calling (full duplex) Telephone directory with 100 entries List of accepted calls (100 entries) Putting on hold and transferring of calls Integrated music on hold (only with SIP) Downloadable ringing tones Caller-specific ringing tones Call forwarding Completion of call on busy subscriber Block and single number dialing Automatic adding to numbers Speed dialing Caller list Compatible with different languages Locking of anonymous callers possible List of accepted calls (100 entries) Display of calls in queue "Presence" display Receiving of "instant messages" Compatible with Microsoft® Windows Messenger® VMail and SMS (RFC3265) Display with 2x 24 characters and graphic panel RFC 326-compatible SIP stack UDP and TCP Digest authentication Static IP and DHCP http and https for configuration files Automatic software updates Integrated HTTP server Completely configurable via Web link 33 Hitachi Wireless IP 5000 162.0000 169.00 Hitachi Hitachi IP Telephone http://www.voipon.co.uk/hitachi-wireless-ip-5000-p-33.html http://www.voipon.co.uk/images/hitachi_wireless_ip_5000.jpg new Availability: Due to a component shortage Hitachi have advised the IP 5000 will be available Late September 08. Painlessly introduce wireless IP phones using your existing LAN, and watch the efficiency of your office communications soar. WirelessIP5000 is an all-around wireless IP phone supporting the Session Initiation Protocol (SIP), which is essential for multimedia communications. With flexible support for the latest IP-PBX systems and existing PBX systems, including IP-Centrex offered by telecommunications service providers, you can painlessly and flexibly introduce the WirelessIP5000 making use of your existing office communications environment. Reduce communications costs, while at the same time greatly reducing the administrative cost and burden of your telephony equipment. Make your communications efficiency soar, both inside the company and out. Office Telephone Issues It is too expensive to change wiring/equipment, and the administrative burden is too great You need to reduce communications costs for the Hitachi Wireless IP 5000 You want to leverage existing telephony equipment and networks You want to stop double investment/administration for IT network and telephone lines You™ve introduced IP phones, but are unable to answer calls because your people are often out of the office or in other parts of the building for the Hitachi Wireless IP 5000 You want to set up an IP telephony solution in the future Implement a flexible wireless IP phone solution scaled to the size of your office and existing systems. Inexpensively build an in-house wireless telephone system. Improve the efficiency of in-house communications by moving your existing IP phone system to wireless. Easily support the move to an in-house IP phone system by introducing VoIP devices. Use of wireless LAN (IP network) reduces capital investment. Painlessly and inexpensively create an ubiquitous network supporting the diverse process requirements and system environments of your business. Wireless IP 5000 provides unrivaled scalability capable of supporting all types of system environments and telephony requirements. From an office primarily using analog phones to a leading-edge office that has introduced IP-PBX and migrated largely to VoIP, WirelessIP5000 flexibly supports a wide range of user environments. Create a truly ubiquitous network for all your business situations. Prevent information leaks through security-aware services The Wireless IP 5000 is equipped with powerful security features conforming to such standards as WEP (64/128/256) and IEEE 802.IX (MD5/EAP-TLS), in order to prevent information leaks in a wireless LAN environment. Authentication via 802.IX (EAP-TLS) achieves secure communications using electronic certificates issued by VeriSign Japan. Communications encryption and prevention of terminal spoofing allows you to access communications comfortably and securely in any environment. Support for WPA, PEAP, and SecureRTP is also planned. Support for Dynamic Network Binding (DNB) Feature Support is available in keeping with your network environment, from environments that use Wireless IP 5000 exclusively, to locations in which wireless LAN access points from multiple networking environments are installed. This is called dynamic network binding (DNB). Offering a new communications environment using SIP and IEEE 802.11b, with an open concept only possible with wireless IP phones. Support for 802.11b allows you to leverage your existing wireless LAN equipment This wireless IP phone is 802.11b-compliant, allowing you to leverage your existing wireless LAN environment. Quickly assess your interlocutor's status with the Presence feature Use the Wireless IP5000's Presence feature to quickly ascertain the situation of the people you communicate with (e.g. whether they are out of the office, in a meeting, or out of town on business). No longer must you call to find out, or head to their office to check. Support for instant messaging Support for instant messaging is available, enabling you to send and receive short messages in real time. Quickly send required information any time, without worrying about the other person's availability for the Hitachi Wireless IP 5000. Handy built-in site-scan feature A built-in site scan feature enables you to measure signal strength. Use this information to find the best access point, and plan your access-point layout for the Hitachi Wireless IP 5000. A wide range of features is offered through combination with PCs Connect to your PC's USB interface using an optional USB cable to recharge your WirelessIP5000's battery. You can also edit the address book stored in your phone's memory, or configure and upgrade your Wireless IP 5000. Use the Ping feature to check communication latency A Ping feature is provided in order to enable you to check whether you can connect to a given person's network or terminal. Quickly perform network diagnostics and other administrative tasks. Configure and view terminal information from a PC browser Use a PC browser to configure or view the IP settings and other information from your Wireless IP 5000. More efficiently manage the terminals in your office. SIP redundancy feature eliminates worry about the unexpected Support for SIP redundancy is offered, enabling you to communicate via an SIP server installed at a branch office if your home-office SIP server should ever go down. Installing secondary SIP servers where appropriate ensures a secure communications path even in the presence of failure for the Hitachi Wireless IP 5000. 44 Vegastream Vega 400 - 1 Fractional E1 (15 Channels) 1599.0000 1699.00 Vegastream Vegastream Vega 400 http://www.voipon.co.uk/vegastream-vega-400-1-fractional-e1-15-channels-p-44.html http://www.voipon.co.uk/images/vegastream_400.jpg new Availability: In Stock The Vega 400 digital gateway connects digital telephony equipment to IP networks. All Vega 400 gateways have four E1 / T1 interfaces and can be purchased with various VoIP capacities up to 120 VoIP channels. The Vega 400 has been designed with future expansion in mind and consists of: Base Vega 400 unit with expansion slots Field-installable expansion modules to provide additional VoIP channel capacity A Vega 400 unit can therefore be installed with one VoIP channel capacity and then later additional capacity can be provisioned as and when it is needed. Each E1 / T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can, therefore, be connected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Open, Non-Proprietary Interfaces The Vega 400 gateway supports the following signaling schemes: ETSI ISDN NI1, NI2, AT&T 5ESS, DMS100 ISO QSIG Basic Call and QSIG feature transparency Channel Associated Signaling (CAS) All VegaStream gateways can support SIP, H.323 and T.38 fax. The Vega 400 gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment. Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.723.1 (5.3/6.4 kbps) G.729a (8 kbps) G.711 (a-law/µ-law) (64 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 120 VoIP channels Telephony Interface Primary Rate ISDN (User configurable NT/TE): 4 x E1 Euro-ISDN ISO QSIG QSIG Feature Transparency (H.323) CAS R2MFC 4 x T1 NI1/NI2 AT&T 5ESS DMS100 CAS (RBS) - E&M wink start - Loop start - Ground start ISO QSIG QSIG Feature Transparency (H.323) LAN Interface 2 x 10 BaseT / 100 BaseTX, full or half duplex Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Certification Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power ISDN: NT/TE and Link up > LAN: Speed, Activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Power 100..240 VAC, 47..63 Hz, 1..0.5 A -48V DC also available, 1.2A (Max) 45 Mediatrix 1102 115 115.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1102-p-45.html http://www.voipon.co.uk/images/mediatrix_1102.jpg new Availability: In Stock Mediatrix 1102 overview: The Mediatrix 1102 is a VoIP access device equipped with two FXS ports and two 10/100 BaseT Ethernet ports. It can connect analog phones or fax machines and legacy PBX and Key Systems to an IP telephony network. Its second Ethernet port connects a PC or a LAN to a wide area network. The Mediatrix 1102 serves as an ideal CPE platform for integration with an existing IP telephony architecture deployed by service providers, carriers or system integrators. For enterprise, residential and SOHO end-users, the Mediatrix 1102 provides a simple, transparent and cost-effective migration to an IP-based telephony infrastructure without the need to discard existing analog devices. Additionally, by providing IP connectivity to analog CO trunk ports of a PBX, or analog CO line ports of a Key System, the Mediatrix 1102 IP-enables legacy telephone systems and makes IP telephony applications available. For service providers, carriers and system integrators, the Mediatrix 1102 creates additional revenue-generating opportunities by immediately bringing existing enterprise, residential and SOHO users online to new high-value IP telephony services. Quite simply, the Mediatrix 1102 is an ideal, cost-effective solution for bringing VoIP to the desktop, connecting SOHOs to an existing enterprise IP telephony network and for connecting residential users to a carrier's or service provider's IP telephony offering. The Mediatrix 1102 makes use of existing broadband access equipment to connect to any standards-based VoIP network. In addition to interfacing with legacy telephone systems, the Mediatrix 1102 also interfaces with IP-based telephone systems by connecting analog phones and fax machines to an IP-PBX or IP-Key System. Plus, the Mediatrix 1102 offers full integration with the Mediatrix IP Communication Server to create an end-to-end converged enterprise solution. Mediatrix 1102 Key Benefits Voice Functionalities Carrier-grade voice quality T.38 support High compression Codecs support PSTN bypass option available for emergency calls Ease of configuration Automatic firmware and configuration file download SNMP and web management TFTP, HTTP or HTTPS auto-provisioning Security Support for SNMPv3 Encrypted configuration files support HTTP Digest authentication HTTPS support Network functionalities Transparent IP Address Sharing PPPoE and DHCP client Interoperable with equipment from leading industry vendors STUN Support Enhanced Telephony Features Multiple country tone support Call Forward / Call Transfer / Conference Call / Call Waiting support Optional PSTN Bypass on power failure / loss of WAN / special number dialing 46 Mediatrix 4104 (1104) 4 port FXS SIP/MGCP Gateway 298 298.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-4104-1104-4-port-fxs-sipmgcp-gateway-p-46.html http://www.voipon.co.uk/images/mediatrix_1104.jpg new Availability: In Stock Mediatrix 1104 overview: The Mediatrix 1104 is a high-quality and cost efficient VoIP gateways connecting small offices to an IP network, while preserving investment in analog telephones and faxes. It allows Services Providers to deploy rapidly and economically their solutions in SOHOs and it is the ideal solution for small branch office connection to larger private networks. The Mediatrix 1104 connects up to 4 analog phones and/or faxes to a broadband modem. The Mediatrix 1104 enables cost-effective VoIP deployments in small offices for both IP Centrex and private network applications. The Mediatrix 1104 has the additional benefit of supporting high compression codecs simultaneously on each analog voice ports, thus saving valuable bandwidth. As all other Mediatrix devices, the 1104 provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible. In addition, an optional intelligent PSTN bypass allows Mediatrix 1104 users to make emergency calls and maintain their phone service in the event of a power outage or network failure. Mediatrix 1104 Key Benefits Voice Functionalities Carrier-grade voice quality T.38 support PSTN bypass option available for emergency calls Ease of configuration and management Automatic firmware and configuration file download Encrypted configuration files support SNMPv3 and web management TFTP or HTTP auto-provisioning Network functionalities QoS Support DHCP client STUN Cient Interoperable with equipment from leading industry vendors Enhanced Telephony Features Call Forward on Busy or on No Answer / Call Transfer / 3-way Conference Call / Call Waiting / Caller ID on Call Waiting / Call on Hold H.245 and Out-of-Band DTMF H.450 basic services Dynamic De-Jitter Manager Voice Activity Detection Mediatrix 1104 Technical Specifications IP Telephony Protocol SIP - RFC3261, RFC3262, RFC3263, H323, MGCP/NCS Real-Time Transport Protocol RTP/RTCP per RFC 1889 and RFC 1890, RFC 2833, RFC 3389 Vocoders G.711 (a-law, U-law), G.723.1, G.726, G.729a, ab Echo Cancellation G.168 Silence Suppression Silence detection / suppression and Comfort Noise Generation level software adjustable. Real-Time Transport Protocols RTP/RTCP ?RFC1889, RFC1890, RFC2833, RFC3389 Network Management Protocols SNMPv3, HTTP (SIP only), TFTP, DHCP WAN Connection 1 10/100 Base T Ethernet RJ-45 connector Analog Connection 4 RJ-11 connectors, analog phone/fax (FXS) interface. Bypass Connection 1 RJ-11 connector, PSTN bypass QoS ToS, DiffServ, 802.1p, 802.1Q, WFQ Real-Time Fax Group 3/Super G3 Fax real-time FoIP over clear channel (G.711), G.726 or T.38 Operating Environment Operating temperature: 0C to 40C Storage temperature: -20C to 70C Humidity: up to 85%, non-condensing Power Supply Internal off-line power supply connected to the AC main with a standard IEC-320 power cord Unit Dimensions and Weight 5.5 cm x 26.0 cm x 17.7 cm (2.2 in. x 10.2 in. x 7 in) Unit Weight 833 g (1.8 lbs) 47 Mediatrix 1124 1598 1598.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1124-p-47.html http://www.voipon.co.uk/images/mediatrix_1124.jpg new Availability: In Stock Mediatrix 1124 overview The Mediatrix 1124 is a high-quality and cost efficient VoIP gateway connecting larger branch offices and multi-tenant buildings to an IP network, while preserving investment in analog telephones and faxes. It allows Services Providers to deploy rapidly and economically their solutions in medium-size premises and it is the ideal solution for branch office connectivity to larger private networks. The Mediatrix 1124 connects up to 24 analog phones and/or faxes to a broadband modem or LAN. The Mediatrix 1124 enables cost-effective VoIP deployments in medium-size branch offices and multi-tenant applications. The Mediatrix 1124 has the additional benefit of supporting high compression codecs simultaneously on each analog voice ports, thus saving valuable bandwidth. As all other Mediatrix devices, the 1124 provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible. In addition, an optional intelligent PSTN bypass allows Mediatrix 1124 users to make emergency calls and maintain their phone service in the event of a power outage or network failure Mediatrix 1124 Key Benefits Voice Functionalities Carrier-grade voice quality T.38 support PSTN bypass option available for emergency calls Ease of configuration and management Automatic firmware and configuration file download SNMP and web management TFTP or HTTP auto-provisioning Security Support for SNMPv3 Encrypted configuration files support HTTP Digest authentication Network functionalities QoS Support DHCP client Interoperable with equipment from leading industry vendors STUN Client Enhanced Telephony Features Call Forward / Call Transfer / 3-way Conference Call /Call Waiting / Caller ID on Call Waiting / Call on Hold H.245 and Out-of-Band DTMF (H.323) H.450 basic services (H.323) Dynamic De-Jitter Manager Voice Activity Detection Mediatrix 1124 Technical Specifications IP Telephony Protocol SIP V RFC3261,H323, MGCP/NCS Real-Time Transport Protocol RTP/RTCP per RFC 1889 and RFC 1890, RFC 2833, RFC 3389 Vocoders G.711 (a-law, U-law), G.723.1, G.726, G.729a, ab Echo Cancellation G.168 Silence Suppression Silence detection / suppression and Comfort Noise Generation level software adjustable. Real-Time Transport Protocols RTP/RTCP V RFC1889, RFC1890, RFC2833, RFC3389 Network Management Protocols SNMPv3, HTTP (SIP only), TFTP, DHCP WAN Connection 1 10/100 Base T Ethernet RJ-45 connector Analog Connection 1 RJ-21X TELCO 25 pairs connector, analog phone/fax (FXS) interface Bypass Connection 1 RJ-11 connector, PSTN bypass QoS ToS, DiffServ, 802.1p, 802.1Q Real-Time Fax Group 3/Super G3 Fax real-time FoIP over clear channel (G.711), G.726 or T.38 Operating Environment Operating temperature: 0°C to 40°C Storage temperature: -20°C to 70°C Humidity: up to 85%, non-condensing Power Supply AC: Standard power cord receptacle (IEC 320-C14) for universal AC input internal SMPS Unit Dimensions and Weight 4.4 cm x 43.0 cm x 21.0 cm (1.74 in. x 17.9 in. x 8.4 in) Unit Weight 1781 g (3.9lbs) 48 Mediatrix 1204 333.3300 358.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1204-p-48.html http://www.voipon.co.uk/images/mediatrix_1204.jpg new Availability: In Stock Mediatrix 1204 overview: The Mediatrix 1204 is a high-quality and cost efficient VoIP gateway connecting IP networks to the PSTN. It is the ideal solution to deploy private or small hosted toll bypass networks. For enterprise end-users, the Mediatrix 1204 provides a simple, transparent and cost-effective way of maintaining a connection to the PSTN while migrating to an IP-based telephony infrastructure. The Mediatrix 1204 connects up to 4 FXO trunks to an IP Ethernet access The Mediatrix 1204 provides PSTN access for various VoIP endpoints such as IP phones, FXS devices, softphones and IP-based PBX and Key Systems. It is an efficient solution to maintain local PSTN breakout in remote locations that are converted to IP. The Mediatrix 1204 provides a gateway to the PSTN for IP-based PBX and Key Systems. Thereby, it allows the deployment of VoIP Remote Line Extension and Branch Office Connectivity solutions without sacrificing any local PSTN access points. By connecting CO lines from selected sites to a VoIP network, the Mediatrix 1204 enables service providers and enterprises to use a VoIP connection between pre-determined local networks. When used in conjunction with Mediatrix' Communication Server, routing schemes and calling rights can be programmed in order to optimize the use of resources and minimize long distance fees. As all other Mediatrix devices, the 1204 provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP or HTTP server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible. Mediatrix 1204 Key Benefits Voice Functionalities Carrier-grade voice quality Ease of configuration and management Automatic firmware and configuration file download SNMP and web management TFTP or HTTP auto-provisioning Security Support for SNMPv3 Encrypted configuration files support HTTP Digest authentication Mediatrix 1204 Technical Specifications IP Telephony Protocol SIP V RFC3261, RFC3262, RFC3263, H323 Real-Time Transport Protocol RTP/RTCP per RFC 1889 and RFC 1890, RFC 2833, RFC 3389 Vocoders G.711 (a-law, U-law), G.723.1, G.726, G.729a, ab Echo Cancellation G.168 Silence Suppression Silence detection / suppression and Comfort Noise Generation level software adjustable. Real-Time Transport Protocols RTP/RTCP ¡V RFC1889, RFC1890, RFC2833, RFC3389 Network Management Protocols SNMPv3, HTTP (SIP only), TFTP, DHCP WAN Connection 1 10/100 Base T Ethernet RJ-45 connector Analog Connection 4 RJ-11 connectors, analog line (FXO) interface QoS ToS, DiffServ, 802.1p, 802.1Q, WFQ Real-Time Fax Group 3/Super G3 Fax real-time FoIP over clear channel (G.711), G.726 or T.38 Enhanced Telephony Features Call Forward on Busy or on No Answer / Call Transfer / 3-way Conference Call / Call Waiting / Caller ID on Call Waiting / Call on Hold H.245 and Out-of-Band DTMF H.450 basic services Dynamic De-Jitter Manager Voice Activity Detection Operating Environment Operating temperature: 0C to 40C Storage temperature: -20C to 70C Humidity: up to 85%, non-condensing Power Supply Internal off-line power supply connected to the AC main with a standard IEC-320 power cord Unit Dimensions and Weight 5.5 cm x 26.0 cm x 17.7 cm (2.2 in. x 10.2 in. x 7 in) Unit Weight 742.4 g (1.6lbs) 49 Mediatrix 2102 85.0000 92.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-2102-p-49.html http://www.voipon.co.uk/images/mediatrix_2102.jpg new Availability: In Stock Mediatrix 2102 overview: The Mediatrix 2102 is a high-quality and cost efficient VoIP gateway connecting SOHOs to an IP network, while preserving investment in analog telephones and faxes. It allows Services Providers to deploy rapidly and economically their solutions in smaller premises and it is the ideal solution for remote line connection to larger private networks. The Mediatrix 2102 connects up to two analog phones and/or faxes, as well as a PC or a home router to a broadband modem. The Mediatrix 2102 enables cost-effective VoIP deployments in residential and SOHO applications. It connects up to two analog phones and/or faxes, as well as a PC or a home router to an IP network over a single broadband connection. With an embedded PPPoE client and its innovative Transparent IP Address Sharing technology, the Mediatrix 2102 and the PC or router connected to the second Ethernet port have the same public IP address, eliminating the need for private IP addresses or address translations. The Mediatrix 2102 has the additional benefit of supporting high compression codecs simultaneously on both analog voice ports, thus saving valuable bandwidth. As all other Mediatrix devices, the 2102 provides web interface, giving users a convenient access to the unit for initial set-up. The devices can also auto-provision by fetching their encrypted configuration from a TFTP, HTTP or HTTPS server making installation secure and transparent to the end-users. To further facilitate deployments, factory loaded configurations are possible. In addition, an optional intelligent PSTN bypass allows Mediatrix 2102 users to make emergency calls and maintain their phone service in the event of a power outage or network failure. Mediatrix 2102 Key Benefits Voice Functionalities Carrier-grade voice quality T.38 support High compression Codecs support PSTN bypass option available for emergency calls Ease of configuration Automatic firmware and configuration file download SNMP and web management TFTP, HTTP or HTTPS auto-provisioning Security Support for SNMPv3 Encrypted configuration files support HTTP Digest authentication HTTPS support Network functionalities Transparent IP Address Sharing PPPoE and DHCP client Interoperable with equipment from leading industry vendors STUN Support Enhanced Telephony Features Multiple country tone support Call Forward / Call Transfer / Conference Call / Call Waiting support Optional PSTN Bypass on power failure / loss of WAN / special number dialing 52 Digium Wildcard TE410P PCI ISDN PRI Card 647.0000 719.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te410p-pci-isdn-pri-card-p-52.html http://www.voipon.co.uk/images/digium_te410p.gif new Availability: In Stock Digium Wildcard TE410P Quad span T1/E1 PCI Card for VoIP/SIP/IAX/Asterisk The TE410P is part of the next generation of Digium hardware designed to improve performance and scalability via bus mastering architecture. The Digium TE410P supports both E1 and T1 environments and is selectable on a per-card/per-port basis. This enables signaling translation between E1 and T1 equipment, allowing inexpensive T1 channel banks to connect with E1 circuits. The TE410P improves I/O speed dramatically (up to 10x); the result is lower CPU usage and increased card density per server. The TE410P has been designed by Digium to be totally compatible with existing applications and it is fully Asterisk integrated. Furthermore, the TE410P's driver is open source and supports an API interface for custom application development. The close links between Digium and the Asterisk open-source PBX make building voIP solutions simpler and more flexible. Further Information The TE410P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE410P supports a 3.3v PCI slot only - typically available on newer motherboards and in 64-bit PCI bus architectures. The TE405P is for use only with a 5.0 volt PCI slot. Documentation Data Modes: for the TE410P Cicso HDLC HDLC PPP Multilink PPP Frame Relay Voice Modes: PRI CPE & PRI NET NI1 NI2 EuroISDN for the TE410P 4ESS (AT&T) 5ESS (Lucent) DMS100 E&M Wink Feature Group B for the TE410P Feature Group D FXO & FXS Ground Start Loop Start for the TE410P Loop Start with Disconnect Detect Certifications: Telecom FCC Part 68 (TIA-968-A), IC CS-03, TBR 4, TBR 12, TBR 13, AS/ACIF S038, AS/ACIF S016 for the TE410P Safety UL/CSA 60950, IEC 60950, EN60950, AS/NZS 60950 Note: Finland, Norway and Sweden require that equipment using this product must be located in a Restricted Access Location (RAL). Emissions: EN55022 Class B Radiated & Conducted Immunity: EN55024 ITE, EN61000 53 Digium Wildcard TE405P PCI ISDN PRI Card 647.0000 719.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te405p-pci-isdn-pri-card-p-53.html http://www.voipon.co.uk/images/digium_te405p.jpg new Availability: In Stock E1/T1 Quad Port Digium TE405P Asterisk Card The Digium TE405P maximises performance and scalability through bus mastering architecture. Supporting both E1 and T1 environments, the TE405P is selectable on a per-port or per-card basis. This enables signaling translation between E1 and T1 equipment and allows cost-efficient T1 channel banks to connect with E1 circuits. The Digium TE405P improves I/O speed by up to 10 times - resulting in reduced CPU load and increased card density per server. Digium has designed the TE405P to be fully compatible with existing software applications and fully Asterisk integrated. The open source driver also supports an API interface for custom application development. The TE405P supports industry standard data and telephony protocols, including Primary Rate ISDN (both North American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE405P is for use only with a 5.0 volt PCI slot. The TE410P is for use only with a 3.3 volt PCI slot - typically available on newer motherboards and in 64-bit PCI bus architectures. Data Modes: Cisco HDLC HDLC PPP Multilink PPP Frame Relay Voice Modes: PRI CPE & PRI NET - NI1 - NI2 - EuroISDN - 4ESS (AT&T) - 5ESS (Lucent) - DMS100 E&M - Wink - Feature Group B - Feature Group D FXO & FXS - Ground Start - Loop Start - Loop Start with Disconnect Detect Certifications: Telecom: FCC Part 68 (TIA-968-A), IC CS-03, TBR 4, TBR 12, TBR 13, AS/ACIF S038, AS/ACIF S016 Safety: UL/CSA 60950, IEC 60950, EN60950, AS/NZS 60950 Emissions EN55022 Class B Radiated & Conducted Immunity EN55024 ITE, EN61000 Note : Finland, Norway and Sweden require that equipment using this product must be located in a Restricted Access Location (RAL). 54 Digium Wildcard TE110P PCI ISDN PRI Card 249.3000 283.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te110p-pci-isdn-pri-card-p-54.html http://www.voipon.co.uk/images/digium_te110p.jpg new Digium Wildcard TE110P VoIP Interface Card   Availability: In stock The Digium Wildcard TE110P brings a high-performance, cost-efficient and flexible single span togglable T1/PRI or E1/PRI interface to the Digium range of IP telephony interface devices. The TE110P is a single span, selectable T1 (24-channel) or E1 (32-channel) card that supports all the functionality of our quad T1/E1 card. This card supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice and 12 to data while passing all traffic through to the Asterisk Open Source PBX, which reliably routes the channels to their designated locations. This eliminates the need for an external router. By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, one can easily connect multiple PCs equipped with the TE110P and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE110Ps to each individual PC. Add addition cards as you need them for your expanding applications. The TE110P supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Data Modes: Cicso HDLC HDLC PPP Multilink PPP Frame Relay Voice Modes: PRI CPE & PRI NET NI1 NI2 EuroISDN 4ESS (AT&T) 5ESS (Lucent) DMS100 E&M Wink Feature Group B Feature Group D FXO & FXS Ground Start Loop Start Loop Start with Disconnect Detect Certifications: Telecom FCC Part 68 (TIA-968-A), IC CS-03, TBR4, TBR12, TBR13, AS/ACIF S038, AS/ACIF S016 for the TE110P Safety UL/CSA 60950, IEC 60950, EN60950, AS/NZS 60950 Note: Finland, Norway and Sweden require that equipment using this product must be located in a Restricted Access Location (RAL). Emissions EN55022 Class A, IEC 61000 Immunity EN55024 ITE, EN61000 65 Ingate SIParator 60 5089 5089.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-siparator-60-p-65.html http://www.voipon.co.uk/images/ingate_siparator.jpg new Your company is communications intensive and you want access to the best in realtime person-to-person IP communications. But your attempts at using Voice over IP (VoIP), presence or instant messaging (IM) have failed because your firewall blocks SIP-based traffic. The Ingate SIParator® is a product that seamlessly works with your existing firewall, allowing employees to utilize SIP-based applications. Internet, the LAN and Firewalls Every business has a Local Area Network (LAN) with Internet access. To maintain privacy and security, the LAN is protected and separated from the public Internet by a firewall. Unfortunately, most firewalls are not SIP-capable, and do not allow SIP-based communications, like VoIP, IM or presence, to go through. The Ingate SIParator is the only product of its kind that allows you to add SIP transparency to your network. The SIParator is designed to support a full range of capacity requirements and is compatible with all commercially available firewalls. The Ingate SIParator performs all SIP proxy and registrar functions. The registrar holds the private IP addresses of the users inside your network, allowing the SIParator® to relay SIP signalling. Once the session has been initiated, the agreed UDP or TCP ports are opened and the SIParator relays the media streams. This functionality enables SIP communications also to and from NAT:ed networks. In gate offers SIParators for enterprises of all sizes. The SIParator can be configured as a part of the DMZ or in a standalone mode. In both cases, the benefits of SIP-based communications can be added to the network simply and easily. SIP Transparency Existing commercially available firewalls prevent delivery of realtime SIP-based communications. Some existing firewalls will allow SIP messages through port 5060, but media streams are not supported. Using Ingate's SIP technology, this problem is solved. The Ingate SIP proxy negotiates between the two end points and dynamically opens the media ports in the SIParator necessary to allow the traffic to flow. Enterprises that want the time and cost saving benefits of interoperable, universal realtime communications without replacing their existing firewall can purchase a one-box solution: the Ingate SIParator. Ingate Remote SIP Connectivity For businesses that want to connect with home office workers and road warriors, Ingate® Systems offers Remote SIP Connectivity, an applications module that enables far-flung users to leverage the benefits of VoIP, instant messaging, presence, video conferencing and other forms of communications based on the SIP protocol that are already integrated into the company's network. 67 Ingate Firewall 1600 4585 4585.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-firewall-1600-p-67.html http://www.voipon.co.uk/images/ingate_firewall_1600.jpg new Availability: In Stock A Leader in the next-generation firewall technology, Ingate systems proudly offers the ingate Firewall 1600 - a powerful appliance that fits the needs of organisations with high demands for capacity, throughput and reliability. Like all firewalls from Ingate, the Firewall 1600 offers the ability to use realtime communications such as Voice over IP (VoIP) and instant messaging (IM) based on the session initiation protocol (SIP). Today many companies are looking to use VoIP and other realtime IP communications. Ingate products meet the special requirements for such usage by fully supporting SIP - the standardized Internet Protocol for live person-to-person IP communications. With ingate, enterprises can use VoIP and other live communications on the LAN and globally via the Internet or Private IP networks. The Ingate Firewall 1600 is only 1 U high, fitting easily into an ever-more-crowded network design. It boasts six interfaces, two of which can be used at Gigabit speed. The Firewall 1600 supports up to 360 simultaneous RTP sessions (e.g. VoIP calls). In addition, the Firewall 1600 includes a new display that communicates its status. Included with the Ingate Firewall 1600 are ten SIP user licenses that can be used for the registration of SIP users on the built-in SIP registrar. Five SIP traversal licenses are also included, allowing up to five calls to traverse the firewall simultaneously. Additional SIP user licenses and SIP traversal licenses can be purchased at any time. The Ingate Firewall 1600 is fully featured, supporting stateful inspection and packet filtering with rules defined and maintained by the network security administrator utilizing the graphical html based user interface. Ingate Firewalls include an encrypted Virtual Private Network (VPN) termination module. In addition to the complete SIP support, Ingate Firewalls have a proxy for all standard protocols, including TCP, UDP, FTP, and DHCP. 74 Digium TDM40B - 4 FXS 151.7000 176.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm40b-4-fxs-p-74.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM40B 4 phone extension (fxs) and 3 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM40B 4 phone extension (fxs) and 3 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 75 Digium TDM02B - 2 FXO 101.3000 119.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm02b-2-fxo-p-75.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM02B 2 incoming line (fxo) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM02B 2 incoming line (fxo) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 76 Digium TDM03B - 3 FXO 134.2000 157.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm03b-3-fxo-p-76.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM03B 3 incoming line (fxo) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM03B 3 incoming line (fxo) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 77 Digium TDM04B - 4 FXO 167.2000 195.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm04b-4-fxo-p-77.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM04B 4 incoming line (fxo) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM04B 4 incoming line (fxo) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 78 Digium TDM10B - 1 FXS 64.4000 76.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm10b-1-fxs-p-78.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM10B 1 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM10B 1 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 79 Digium TDM11B - 1 FXO 1 FXS 97.5000 112.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm11b-1-fxo-1-fxs-p-79.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM11B 1 incoming line (fxo) and 1 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM11B 1 incoming line (fxo) and 1 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 80 Digium TDM13B - 3 FXO 1 FXS 163.2000 190.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm13b-3-fxo-1-fxs-p-80.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM13B 3 incoming line (fxo) and 1 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM13B 3 incoming line (fxo) and 1 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 81 Digium TDM12B - 2 FXO 1 FXS 130.4000 147.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm12b-2-fxo-1-fxs-p-81.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM12B 2 incoming line (fxo) and 1 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM12B 2 incoming line (fxo) and 1 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 82 Digium TDM01B - 1 FXO 68.4000 81.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm01b-1-fxo-p-82.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM01B 1 incoming line (fxo) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM01B 1 incoming line (fxo) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 83 Digium TDM20B - 2 FXS 93.5000 109.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm20b-2-fxs-p-83.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM20B 2 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM20B 2 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 84 Digium TDM21B - 1 FXO 2 FXS 126.5000 142.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm21b-1-fxo-2-fxs-p-84.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM21B 1 incoming line (fxo) and 2 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM21B 1 incoming line (fxo) and 2 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 85 Digium TDM22B 2 FXO 2 FXS 159.4000 180.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm22b-2-fxo-2-fxs-p-85.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM01B 1 incoming line (fxo) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM01B 1 incoming line (fxo) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 86 Digium TDM30B - 3 FXS 122.7000 142.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm30b-3-fxs-p-86.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM30B 3 phone extension (fxs)serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM30B 3 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 87 Digium TDM31B - 1 FXO 3 FXS 155.6000 176.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm31b-1-fxo-3-fxs-p-87.html http://www.voipon.co.uk/images/digium_analog_card.jpg new Availability: In Stock The TDM31B 1 incoming line (fxo) and 3 phone extension (fxs) serves as an interface with analogue telephones or analogue lines for integrating into Asterisk PBX. Used in conjunction with Asterisk a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX platform. The TDM31B 1 incoming line (fxo) and 3 phone extension (fxs) takes the place of an expensive channel bank and brings the system price point to the lowest in the industry. To scale this solution simply add additional modules or TDM cards as required. Until now, this low cost and revolutionary solution had yet to be filled in the telephony space. 88 Epygi QuadroCS Conference Server 2187.75 2187.75 Epygi Quadro Conference http://www.voipon.co.uk/epygi-quadrocs-conference-server-p-88.html http://www.voipon.co.uk/images/quadrocs.jpg new Availability: In Stock Epygi Technologies, Ltd. extends its product line with the Conference Server QuadroCS, a low cost, feature rich alternative to costly conferencing services or expensive conferencing systems. The Conference Server unifies voice communications between the PSTN, isolated SIP islands and commercial IP gateway services provided by Internet Telephone Service Providers (ITSP). One Conference Server can be shared across multiple global workgroups. The unifying value of the Conference Server accelerates the adoption - cycle of SIP voice communications allowing disparate installations to conveniently interconnect in a QuadroCS conference. 89 Epygi Quadro E1 / T1 Gateway 2132.75 2132.75 Epygi Quadro E1 / T1 Gateway http://www.voipon.co.uk/epygi-quadro-e1-t1-gateway-p-89.html http://www.voipon.co.uk/images/quadro_e1_t1.jpg new Availability: In Stock Your business already owns its own PBX, but now you want to move into the World of IP telephony for its higher performance and lower cost. Look to the QuadroE1/T1 and move quickly to Voice over IP. The QuadroE1/T1 VoIP gateway completely unites both PSTN and VoIP technologies, even in legacy equipment configurations. Just install the QuadroE1/T1 gateway between your Internet access router and your existing PBX to start making IP phone calls anywhere in the world. In addition, the QuadroE1/T1 can connect to a PSTN service provider via E1/T1 to provide access and virtual local phone numbers to foreign users. Of course, the QuadroE1/T1 may be connected directly either to the Main Office via E1/T1 trunk and to the Internet using its integrated access router. It's Flexible If the QuadroE1/T1 is communicating with another Quadro device or IP phone on the far-end, the calls will be nearly free. If you connect to a VoIP service provider, you can make long-distance calls to any phone number on the public phone network for a fraction of the regular cost. The QuadroE1/T1 combines the cost-reducin benefits of IP telephony with the ubiquity of the PSTN, which opens many scenarios for free phone calls. Connect PBXs of geographically disperse departments of a company wherever they are in the world over IP. Free calls between the departments can quickly increase communication and effectiveness. Since the QuadroE1/T1 contains call routing information, employees who are out of the office can still be reached automatically. The world becomes your virtual office. In addition, a pair of QuadroE1/T1 units can be deployed to provide virtual local numbers to parties anywhere in the world. Communication has never been so flexible. 90 Epygi Quadro 2x 464.1 464.10 Epygi Quadro IP PBX Systems http://www.voipon.co.uk/epygi-quadro-2x-p-90.html http://www.voipon.co.uk/images/quadro_2x.jpg new Availability: In Stock The Quadro2x is the fully-featured IP PBX for home offices and teleworkers. Connected by one FXO line to the PSTN and over 10Base-T Ethernet to the Internet, the Quadro2x allows attaching 2 FXS phones and two fully-featured LAN IP phones (SIP/ MGCP). Up to 24 additional virtual extension allow e.g., registering at various SIP servers. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro2x offers a multitude of features including extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. <SPAN class=maintextbold> Integrated Internet Access Router The Quadro2x IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. <SPAN class=maintextbold> VoIP Carriers The Quadro2x IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro2x to work according to individual requirements 91 Epygi Quadro 4x 1575 1575.00 Epygi Quadro IP PBX Systems http://www.voipon.co.uk/epygi-quadro-4x-p-91.html http://www.voipon.co.uk/images/quadro_4x.jpg new Availability: In Stock The Quadro4x is the fully-featured IP PBX for small businesses. Connected by two FXO lines to the PSTN and over 10Base-T Ethernet to the Internet, the Quadro4x provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully-featured SIP phones may be attached to Quadro4x's LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro4x offers a multitude of features including extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. <SPAN class=maintextbold> Integrated Internet Access Router The Quadro4x IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. <SPAN class=maintextbold> VoIP Carriers The Quadro4x IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro4x to work according to individual requirements. 92 Epygi Quadro 16x 2180.25 2180.25 Epygi Quadro IP PBX Systems http://www.voipon.co.uk/epygi-quadro-16x-p-92.html http://www.voipon.co.uk/images/quadro_16x.jpg new Availability: In Stock The Quadro16x is the fully-featured IP PBX for enhanced small businesses. Connected by four FXO lines to the PSTN and over 10Base-T Ethernet to the Internet, the Quadro16x provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully-featured SIP phones may be attached to Quadro16x's LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro16x offers a multitude of features including extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16x IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro 16x IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro16x to work according to individual requirements. 93 Epygi Quadro 4xa 1494 1494.00 Epygi Quadro IP PBX ADSL http://www.voipon.co.uk/epygi-quadro-4xa-p-93.html http://www.voipon.co.uk/images/quadro_4xa.jpg new Availability: In Stock The Quadro4a is the fully-featured IP PBX for small businesses that prefer ADSL as their uplink connectivity. Connected by two FXO lines to the PSTN and over ADSL to the Internet, the Quadro4xa provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully-featured SIP phones may be attached to Quadro4xa's LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro4xa offers a multitude of features including extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro4xa IP PBX embodies a fully-fledged Internet access router with fi rewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro4xa IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro4xa to work according to individual requirements. 94 Epygi Quadro 4xia 1494 1494.00 Epygi Quadro IP PBX ADSL http://www.voipon.co.uk/epygi-quadro-4xia-p-94.html http://www.voipon.co.uk/images/quadro_4xia.jpg new Availability: In Stock The Quadro4xia is the fully featured IP PBX for small businesses that prefer ISDN to connect to the PSTN and ADSL for Internet connectivity. With one ISDN line and one ADSL port Quadro4xia can enable up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully featured SIP phones may also be attached to Quadro4xia's LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Feature Rich IP PBX The Quadro4xia offers the same multitude of features as all Quadro IP PBXs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro4xia IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro4xia IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented. 95 Epygi Quadro 16xa 2180.25 2180.25 Epygi Quadro IP PBX ADSL http://www.voipon.co.uk/epygi-quadro-16xa-p-95.html http://www.voipon.co.uk/images/quadro_16xa.jpg new Availability: In Stock The Quadro16xa is the fully-featured IP PBX for enhanced small businesses prefering ADSL Internet connectivity. Connected by four FXO lines to the PSTN and over ADSL to the Internet, the Quadro16xa provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully-featured SIP phones may be attached to Quadro16xa's LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro16xa offers the multitude of features that is characteristic for Quadro IP PBXs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16xa IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro 16xa IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro16xa to work according to individual requirements. 96 Epygi Quadro 16xia 2180.25 2180.25 Epygi Quadro IP PBX ADSL http://www.voipon.co.uk/epygi-quadro-16xia-p-96.html http://www.voipon.co.uk/images/quadro_16xia.jpg new Availability: In Stock The Quadro16xia is the fully featured ISDN IP PBX for growing small businesses that prefer ISDN connectivity for the PSTN and ADSL for the Internet. With its three ISDN lines and three ADSL lines the Quadro16xia provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully featured SIP phones may be attached to Quadro16xia's LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Feature Rich IP PBX The Quadro16xia offers the same multitude of features as all Quadro IP PBXs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16xia IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. Data security is also enhanced by an extensive IPSec VPN functionality with DES and 3DES encryption. VoIP Carriers As with all Quadro IP PBXs, the Quadro16xia is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the confi guration of Quadro16xi to work according to individual requirements. 97 Epygi Quadro 4xs 2068 2068.00 Epygi Quadro IP PBX SDSL http://www.voipon.co.uk/epygi-quadro-4xs-p-97.html http://www.voipon.co.uk/images/quadro_4xs.jpg new Availability: In Stock The Quadro4xs is the fully featured IP PBX for small businesses that prefer G.SHDSL as their uplink connectivity. Connected by two FXO lines to the PSTN and over G.HDSL to the Internet, the Quadro4xs provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully featured SIP phones may also be attached to Quadro4xs&trade; LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Feature Rich IP PBX The Quadro4xs offers a multitude of features including extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro4xs IP PBX is a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. Data security is also enhanced by an extensive IPSec VPN functionality with DES and 3DES encryption. VoIP Carriers The Quadro4xs IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the confi guration of Quadro4xs to work according to individual requirements. 98 Epygi Quadro 4xis 2068 2068.00 Epygi Quadro IP PBX SDSL http://www.voipon.co.uk/epygi-quadro-4xis-p-98.html http://www.voipon.co.uk/images/quadro_4xis.jpg new Availability: In Stock The Quadro4xis is the fully featured ISDN IP PBX for small businesses that prefer ISDN connectivity to the PSTN and G.SHDSL for the Internet. With one ISDN line and one G.SHDSL port Quadro4xis can enable up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully featured SIP phones may also be attached to Quadro4xis&trade; LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Feature Rich IP PBX The Quadro4xis offers the same multitude of features as all Quadro IP PBxs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro4xis IP PBX is a fully-fl edged Internet access router with fi rewall security, including IDS, NAT, policy and service based fi ltering as well as stateful stateful inspection. Data security is also enhanced by an extensive IPSec VPN functionality with DES and 3DES encryption. VoIP Carriers The Quadro4xis IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of the Quadros to work according to individual requirements. 99 Epygi Quadro 16xs 3448 3448.00 Epygi Quadro IP PBX SDSL http://www.voipon.co.uk/epygi-quadro-16xs-p-99.html http://www.voipon.co.uk/images/quadro_16xs.jpg new Availability: In Stock The Quadro16xs is the fully featured IP PBX for growing small businesses prefering G.SHDSL Internet connectivity. Connected by four FXO lines to the PSTN and over G.SHDSL to the Internet, the Quadro16xs provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully featured SIP phones may also be attached to Quadro16xs&trade; LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your offi ce. Feature Rich IP PBX The Quadro16xs offers the multitude of features that is characteristic for Quadro IP PBXs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16xs IP PBX is a fully-fledged Internet access router with fi rewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. Data security is also enhanced by an extensive IPSec VPN functionality with DES and 3DES encryption. VoIP Carriers The Quadro 16xs IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro16xs to work according to individual requirements. 100 Epygi Quadro 16xis 3448 3448.00 Epygi Quadro IP PBX SDSL http://www.voipon.co.uk/epygi-quadro-16xis-p-100.html http://www.voipon.co.uk/images/quadro_16xis.jpg new Availability: In Stock The Quadro16xis is the fully featured ISDN IP PBX for growing small businesses that prefer ISDN connectivity to the PSTN and G.SHDSL for the Internet. With three ISDN lines and one G.SHDSL port Quadro4xis can enable up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully featured SIP phones may also be attached to Quadro16xis&trade; LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your offi ce. Feature Rich IP PBX The Quadro16xis offers the same multitude of features as all Quadro IP PBXs: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16xis IP PBX is a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. Data security is also enhanced by an extensive IPSec VPN functionality with DES and 3DES encryption. VoIP Carriers As with all Quadro IP PBXs, the Quadro16xis is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of Quadro16xis to work according to individual requirements. 101 Epygi Quadro 2xi 464.1 464.10 Epygi Quadro IP PBX ISDN http://www.voipon.co.uk/epygi-quadro-2xi-p-101.html http://www.voipon.co.uk/images/quadro_2xi.jpg new Availability: In Stock The Quadro2xi is an as fully-featured IP PBX for home offices and teleworkers as the Quadro2x is. The important difference: Quadro2xi offers you an ISDN line instead of FXO and connects you via two B-channels of 64 kbit/s each to the digital PSTN. A 10Base-T Ethernet line provides access to the Internet. Alltogether, two standard phones and two fully featured LAN IP phones (SIP/MGCP) may be attached. Up to 24 additional virtual extension allow e.g., registering at various SIP servers. In addition, all extensions assigned to the ISDN line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were at your home office. Fully-Featured IP PBX The Quadro2xi offers the same multitude of features as the Quadro2x: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro2xi IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro2xi IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the confi guration of Quadro2xi to work according to individual requirements. 102 Epygi Quadro 4xi 1575 1575.00 Epygi Quadro IP PBX ISDN http://www.voipon.co.uk/epygi-quadro-4xi-p-102.html http://www.voipon.co.uk/images/quadro_4xia.jpg new Availability: In Stock The Quadro4xi is the fully-featured ISDN IP PBX for small businesses. Connected by one ISDN line to the PSTN and over 10Base-T Ethernet to the Internet, the Quadro4xi provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides four FXS phones, up to 16 fully-featured SIP phones may be attached to Quadro4xi's LAN port. In addition, all SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro4xi offers the same multitude of features as Quadro4x: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro4xi IP PBX embodies a fully-fledged Internet access router with fi rewall security, including IDS, NAT, policy and service based filtering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers The Quadro4xi IP PBX is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the configuration of the Quadros to work according to individual requirements. 103 Epygi Quatro 16xi 2180.25 2180.25 Epygi Quadro IP PBX ISDN http://www.voipon.co.uk/epygi-quatro-16xi-p-103.html http://www.voipon.co.uk/images/quadro_16xi.jpg new Availability: In Stock The Quadro16xi is the fully-featured ISDN IP PBX for enhanced small businesses. Connected by three ISDN lines to the PSTN and over 10Base-T Ethernet to the Internet, the Quadro16xi provides up to 70 physical and virtual extensions to make up to 20 calls in parallel. Besides 16 FXS phones, 48 fully-featured SIP phones may be attached to Quadro16xi's LAN port. In addition, the SIP extensions assigned to a line may be mirrored and act as remote IP extensions. Wherever you are, calls will reach you, and you will be able to make calls as if you were in your office. Fully-Featured IP PBX The Quadro16xi offers the same multitude of features as the Quadro16x: extensive PBX functionality, integrated voicemail, cost saving call routing functionality, IP routing, SIP and H.323 call signaling and much more. Integrated Internet Access Router The Quadro16xi IP PBX embodies a fully-fledged Internet access router with firewall security, including IDS, NAT, policy and service based fi ltering as well as stateful inspection. An extensive IPSec VPN functionality with DES and 3DES encryption makes an additional contribution to data security. VoIP Carriers As all Quadro IP PBXs, Quadro16xi is easily applicable for VoIP Carriers as well as small businesses. For VoIP carriers, a special wizard has been implemented that facilitates the confi guration of Quadro16xi to work according to individual requirements. 104 Digium Asterisk Business Edition 424.8000 498.00 Digium Digium Business Edition http://www.voipon.co.uk/digium-asterisk-business-edition-p-104.html http://www.voipon.co.uk/images/business_edition.gif new Availability: In Stock Digium, the leader in open source telephony, announces Asterisk Business Edition, a professional-grade version of its acclaimed open source PBX for the Linux operating system. This version provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications. An all-new Asterisk technical manual and quick-start documentation supplements the package, making Asterisk even easier to install, configure, and use. Asterisk Business Edition is backed by Digium's professional support team with a full one year limited warranty. This provides enterprise environments with a PBX and telephony platform suitable for critical business applications. Digium's comprehensive test program ensures Asterisk Business Edition's reliability, performance, and interoperability with key hardware, software, and protocols. Digium hardware cards are tested for full compatibility with Asterisk Business Edition, as are several select models of servers, VoIP, and TDM devices. All major software features in Asterisk Business Edition are thoroughly tested for functionality and reliability. Test bed systems are also subjected to extreme stress conditions using Empirix® test equipment to simulate hundreds of thousands of calls in various real-world combinations and configurations. As a result, customers can rely on their combination of proven Asterisk software and Digium hardware to work together to provide a feature-rich PBX or VoIP system. 105 Digium Wildcard TE411P PCI ISDN PRI Card 998.0000 998.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te411p-pci-isdn-pri-card-p-105.html http://www.voipon.co.uk/images/te411p.gif new Availability: This product is now obsolete Digium TE411P Asterisk Card with Echo Cancellation The TE411P is the next generation of Digium hardware that offer on-board echo cancellation. They supports both E1 and T1 environments and is selectable on a per-card or per-port basis. Echo cancellation and DTMF detection is supported for four full T1 (96 channels) or E1 (124 channels) and improves voice quality in situations where software echo cancellation is not sufficient, where echo cancellation is not done at the CO, or where CPU utilization must be minimized. By supporting 16ms of echo cancellation over 128 channels, 32ms over 64 channels, or 64ms over 32 channels, this card will perform in the most difficult of environments while providing capacity/length tradeoff. It provides echo cancellation to analog channel banks or for E&M signaling with echo from the far end. The TE411P reduces CPU overhead required for software echo cancellation and frees resources for other processes, such as codec translation. Digium has designed the TE411P to be fully compatible with existing software applications and it is fully integrated with the Asterisk open source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. The TE411P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE411P is for use with a 3.3-volt PCI slot. Target Applications Legacy PBX/IVR services Voice-over Internet Protocol (VoIP) services. Complex IVR Trees Meet-me-Bridge conferencing VoIP Gateways (supports SIP, H.323 and IAX) PRI Switch Compatibility EuroISDN (PRI or PRA) Q.931/Q.921 AT&T 4ESS DMS 100 Lucent 5E Network or CPE National ISDN 2 CAS Voice Modes Feature Group D E&M Wink A-law, u-law and linear modes supported Data Modes SyncPPP (Both Fixed and Dialup) Framr Relay Cisco HDLC Multi-link PPP Echo Cancellation G.168 compliant 128 taps over 128 channels (16ms over 128 channels) 106 Digium Wildcard TE406P PCI ISDN PRI Card 998.0000 998.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te406p-pci-isdn-pri-card-p-106.html http://www.voipon.co.uk/images/te406p.gif new Availability: Please note this product has been discontinued The TE406P is the next generation of Digium hardware that offer on-board echo cancellation, supporting both E1 and T1 environments - selectable on a per-card or per-port basis. Echo cancellation and DTMF detection is supported for four full T1 (96 channels) or E1 (124 channels) and improves voice quality in situations where software echo cancellation is not sufficient, where echo cancellation is not done at the CO, or where CPU utilization must be minimized. By supporting 16ms of echo cancellation over 128 channels, 32ms over 64 channels, or 64ms over 32 channels, this card will perform in the most difficult of environments while providing capacity/length tradeoff. It provides echo cancellation to analog channel banks or for E&M signaling with echo from the far end. The TE406P reduces CPU overhead required for software echo cancellation and frees resources for other processes, such as codec translation. Digium has designed the TE406P to be fully compatible with existing software applications and it is fully integrated with the Asterisk open source PBX/IVR platform. Also, the open source driver supports an API interface for custom application development. With the combination of Digium Hardware and Asterisk software, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications. The TE406P support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE406P is for use only with a 5.0-volt PCI slot. Target Applications Legacy PBX/IVR services Voice-over Internet Protocol (VoIP) services. Complex IVR Trees Meet-me-Bridge conferencing VoIP Gateways (supports SIP, H.323 and IAX) PRI Switch Compatibility EuroISDN (PRI or PRA) Q.931/Q.921 AT&T 4ESS DMS 100 Lucent 5E Network or CPE National ISDN 2 CAS Voice Modes Feature Group D E&M Wink A-law, u-law and linear modes supported Data Modes SyncPPP (Both Fixed and Dialup) Framr Relay Cisco HDLC Multi-link PPP Echo Cancellation G.168 compliant 128 taps over 128 channels (16ms over 128 channels) 107 Digium Wildcard TE210P PCI ISDN PRI Card 384.0000 405.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te210p-pci-isdn-pri-card-p-107.html http://www.voipon.co.uk/images/te210p.gif new Availability: In Stock Digium TE210P Dual T1/E1/J1 Scaleable PCI Interface Card The TE210P is part of the newer generation of Digium hardware designed to improve performance and scalability through bus mastering architecture. The Digium TE210P supports E1, T1, and J1 environments - selectable on a per-card or per-port basis. This allows signaling translation between E1, T1, J1 equipment and allows inexpensive T1/J1 channel banks to connect with E1 circuits. Because the TE210P improves I/O speed by up to 10 times, the result is lower CPU usage and increased card density per server. Digium has designed the TE210P to be fully compatible with existing applications and it is fully integrated with the Asterisk PBX/IVR. The open source driver also supports an API interface for custom application development. The TE210P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE210P supports a 3.3v PCI slot only - typically available on newer motherboards and in 64-bit PCI bus architectures. The TE205P is for use only with a 5.0 volt PCI slot. Data Modes: Cisco HDLC HDLC PPP Multilink PPP Frame Relay Voice Modes: PRI CPE & PRI NET NI1 NI2 EuroISDN 4ESS (AT&T) 5ESS (Lucent) DMS100 E&M Wink Feature Group B Feature Group D FXO & FXS Ground Start Loop Start Loop Start with Disconnect Detect 108 Digium Wildcard TE205P PCI ISDN PRI Card 384.0000 405.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te205p-pci-isdn-pri-card-p-108.html http://www.voipon.co.uk/images/te205p.gif new Availability: In Stock Digium TE205P E1 & T1/J1 Asterisk Card The Digium TE205P is the next generation of hardware that improves performance and scalability through bus mastering architecture. Both E1 and T1/J1 environments are supported on the TE205P - selectable on a per-card or per-port basis. This enables signaling translation between E1 and T1/J1 equipment and allows inexpensive T1/J1 channel banks to connect with E1 circuits. Because the TE205P improves I/O speed over slave-only architecture, the result is reduced CPU usage and increased card density per server. The Digium TE205P has been designed to be fully compatible with existing software applications and is fully integrated with the open-source Asterisk PBX/IVR platform. In addition, the open source driver supports an API interface for custom application development. Using Digium hardware and Asterisk, numerous combinations of telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications. The TE205P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE205P is for use only with a 5.0-volt PCI slot. The TE210P is for use only with a 3.3-volt PCI slot. Data Modes: Cisco HDLC HDLC PPP Multilink PPP Frame Relay Voice Modes: PRI CPE & PRI NET NI1 NI2 EuroISDN 4ESS (AT&T) 5ESS (Lucent) DMS100 E&M Wink Feature Group B Feature Group D FXO & FXS Ground Start Loop Start Loop Start with Disconnect Detect 109 Digium TDM400P FXO Module 37.8000 42.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm400p-fxo-module-p-109.html http://www.voipon.co.uk/images/tdm400p_fxo.gif new Availability: In Stock The FXO module allows the TDM400P card to terminate analog telephone lines (POTS). Because of the modular design, a user can activate additional ports at any time with more FXS or FXO daughter cards. The FXO module passes all the call features any standard analog telephone line will support. Worldwide certifications are pending. 110 Digium TDM400P FXS Module 36.0000 40.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm400p-fxs-module-p-110.html http://www.voipon.co.uk/images/tdm400p_fxs.gif new Availability: In Stock The FXS module allows the TDM400P card to terminate analog telephones. Because of the modular design, a user can activate additional ports at any time with more FXS or FXO daughter cards. The FXS module passes all the call features any standard analog telephone will support. 111 Vegastream Vega 50 6x4 - 4 fxs + 2 fxo 711.0000 790.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-4-fxs-2-fxo-p-111.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 4 FXS + 2 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 112 Vegastream Vega 50 6x4 - 8 FXS + 2 FXO 873.0000 970.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-8-fxs-2-fxo-p-112.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 8 FXS + 2 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 113 Vegastream Vega 50 6x4 - 24 x FXS + 2 x FXO 1323.0000 1470.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-24-x-fxs-2-x-fxo-p-113.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 24 FXS + 2 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 114 Vegastream Vega 50 6x4 - 4 x FXO 648.0000 720.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-4-x-fxo-p-114.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 4 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 115 Vegastream Vega 50 6x4 - 4 x BRI 945.0000 1050.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-4-x-bri-p-115.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 4 BRI interfaces (8 channels) Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 116 Vegastream Vega 50 6x4 - 4 x BRI + 4 x FXO 1278.0000 1420.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-4-x-bri-4-x-fxo-p-116.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 4 BRI interfaces (8 channels) + 4 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 117 Grandstream GXP 2000 IP Phone 58.9000 62.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp-2000-ip-phone-p-117.html http://www.voipon.co.uk/images/gxp_2000.gif new Availability: In Stock The Grandstream GXP-2000 is a modern business IP phone that utilises open technology standards. Created with leading edge technologies, the GXP2000 provides innovative and excellent sound quality, extensive features and great configurability at amazing prices. The Grandstream GXP2000 is expandable, secure and easy to manage, offering 4 individual SIP accounts, 7 programmable keys, visual message indicator, full duplex hands-free speakerphone, dual 10M/100Mbps Ethernet ports, intuitive user interfaces, large back-lit graphical LCD display, security and privacy protection, XML screen content customisation, automated phone book synchronisation with directory server using XML, as well as broad interoperability with most 3rd party SIP products. The GXP2000 is an ideal IP phone for both the small business and the enterprise customer. Supports SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP, PPPoE, TFTP, NTP & TLS (pending) Supports NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), LDAP and SOAP (pending) Advanced jitter buffer control, packet delay and loss concealment technology Support for popular codecs including G.711, G.726, G.728, G.723, G.729, GSM, G.722 (wideband) and ILBC with dynamic negotiation of codec type and packet time Supports 4 independent SIP accounts or SIP Server platforms with 11 line indicators (expandable to a 123 more through expansion key-module) Supports 7 programmable keys Large 131x64 Graphical LCD to display up to 8 lines and 22 characters per line Full duplex speakerphone with advanced echo cancellation Dual 10/100Mbps Ethernet ports Headset jack Supports Caller ID display or block, per call or permanent Supports Call waiting, Hold, Mute, Transfer (blind or attended), Forward (on busy, unconditional or no answer) , and more Supports Multi-party conferencing, Intercom and various DTMF options (RFC2833, SIP INFO, in audio) Supports Integrated Power-over-Ethernet (802.3af and Cisco style) Message Waiting Indicator for voicemail Call Log Phone Book Syslog for monitoring and 128-bit AES encryption of downloadable configuration Support for Layer 2 QoS (802.1Q VLAN and 802.1p) and Layer 3 QoS (DiffServ, IP ToS) And many more enterprise grade features 118 Snom 320 IP Telephone 106.0000 115.00 Snom Snom IP Telephones http://www.voipon.co.uk/snom-320-ip-telephone-p-118.html http://www.voipon.co.uk/images/snom_320_sml.jpg new Availability: In Stock The Snom 320 is perfect for general office and knowledge-worker environments. It is an affordable, yet powerful SIP business voIP phone with built-in, full-duplex speakerphone as well as three-party conference bridging. A 2 x 24 semi-graphic LCD display and menu-driven user interface offers custom branding and easy feature management. The Snom 320 has 12 programmable keys with LEDs to support flexible trunk-access/busy lamp configuration. The 320 also offers a 100-number call memory, 100-number onboard address book (to which data may easily be uploaded), custom call blocking, configurable/downloadable ringtones, auto-answer mode, DND and other sophisticated features to insure maximum convenience and productivity. In addition, the Snom 320's built-in web server supports even simpler end-user configuration, screen dialing and access to call history. The snom 320 is remotely-manageable with upgradeable firmware, is uniquely easy to install and is largely self-configuring. Broad codec support and full compatibility with current SIP recommendations insures interoperability; support for STUN (for NAT traversal), ENUM (for dialed-number resolution) and other state-of-the-art features enables flexible deployment, behind local proxies, IP PBXs or hosted VoIP services. The snom 320 supports the SRTP security standard (a current specification from the Internet Engineering Task Force (IETF)) - for protection against eavesdropping - and TLS for protection against sniffing of signaling and authentication data. By limiting the need for external conference bridges/media server capacity or the use of conference services for routine multiparty calls, the snom 320's built-in three-party conference bridge helps limit total cost of ownership, while also insuring high audio quality and low latency. Key Features two-line display with graphical field 47 keys, 13 LEDs 12 programmable function keys Speakerphone Dual Ethernet connection Power over Ethernet Headset connection SIP RFC3261 Security: SIPS/SRTP STUN, ENUM, NAT, UPnP, ICE Compression: G.723.1 and others National Language Support Snom IP Phone Comparison Table Phone Power over ethernet Display SIP Identities Programmable Keys with LEDs Large LED for incoming calls Extension Keyboard Available Wireless headset adaptor Snom 300 Y 2 Line 16 Characters 4 6 N N N Snom 320 Y Hinged 2 line character display with graphical field 12 12 N Y Y Snom 360 Y Hinged, backlit graphical display (128x64 pixels) 12 12 N Y Y Snom 370 Y Hinged, backlit, high-definition graphical display (240x128 pixels) 12 12 Y Y Y 119 Polycom SoundPoint IP301 77.0000 84.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip301-p-119.html http://www.voipon.co.uk/images/polycom_ip301_sml.jpg new Availability: In Stock Polycom SoundPoint IP 301 - Cost Effective voIP Telephony The Polycom SoundPoint IP301 is a two-line desktop IP phone that provides remarkable value. Similar in sound quality and ease-of-use to the Polycom SoundPoint IP500 and SoundPoint IP600, the SoundPoint IP 301 offers users an entry-level, high quality choice for their IP telephony requirements. Working jointly with Polycom's IP Telephony Technology Partners, the SoundPoint IP301 supports core enterprise functionality. The Polycom SoundPoint IP 301 offers an easy transition from traditional PBX features and functionality into the world of voIP with 10 dedicated hard keys and 3 context sensitive soft keys. The IP301's intuitive user interface offers one button access to common telephony features whilst an information-rich display delivers content for messaging, directory access, call information and applications. Suitable for everyday users, the SoundPoint IP301 offers excellent voice quality in handset or headset mode. This enhances the productivity of business phone calls, with less time spent trying to understand what people are saying - and more time communicating with them.   Soundpoint IP301 Benefits & Features Standards based 2-line entry-level phone 4 line x 20 character LCD 2 port 10/100 Ethernet switch Flexible powering options Intuitive user interface with single-button access to telephony features offers easy transition from legacy PBX and Key Systems to Voice over IP Interoperability with leading IP PBX and Softswitch platforms SIP or MGCP Dedicated RJ-9 headset port Support of IEEE 802.3af or Cisco PoE with optional cable accessory Reversible base stand / wall mount 120 Polycom SoundPoint IP501 115.0000 123.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip501-p-120.html http://www.voipon.co.uk/images/polycom_ip501_sml.jpg new Availability: In Stock Polycom Soundpoint IP 501 - 3-line desktop IP Phone Delivering Exceptional Sound Quality The power of a network. The simplicity of a phone. Whether deploying MGCP or SIP standards, Polycom offers a solution that fits all of your business communication needs. The SoundPoint IP 501 is a voIP telephone that seamlessly integrates with IP PBX and Softswitch vendors' IP solutions. With an intuitive user interface, the IP 501 offers dedicated, one-button access to common telephony features, whilst an information-rich display delivers content for messaging, directory access, call information and applications. The SoundPoint IP 501 provides superb voice quality when used in handset, headset or speakerphone mode. In addition, as protocols develop and standards evolve, it's easy to update the phones in the field via a software download, enabling new features and functionality as they become available. Polycom Soundpoint IP 501 Features & Benefits Field upgradeable Message alert indicator Hands-free speakerphone Superb voice quality when used with a headset, handset or in hands-free speakerphone mode Freedom to choose multiple protocols and platforms Dual-port 10/100 Ethernet switch Direct port for headset Large, content-rich 160 x 80 display 4 context-sensitive soft keys Multiple call appearances Full-duplex speakerphone with Polycom Acoustic Clarity Technology Dedicated feature keys offer one-button access to essential telephony features Interoperable with leading IP PBX and Softswitch platforms SIP or MGCP Dedicated RJ-9 headset port Support of IEEE 802.3af or Cisco PoE with optional cable accessory 121 Polycom SoundPoint IP601 162.0000 179.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html http://www.voipon.co.uk/images/polycom_ip601_sml.jpg new Availability: In Stock Polycom Soundpoint IP 601 - 6-lines, advanced functionality & Expansion Module support Six-line convenience with unsurpassed hands-free voice quality and a high-resolution display. The SoundPoint IP 601 is a great choice for a standards-based Voice over IP telephone. It is ideal for users requiring multiple lines, and delivers both traditional telephone capabilities and new, converged applications to your business desktop. Incorporating a two-port Ethernet switch and auto-sensing Power over Ethernet circuitry, the SoundPoint IP 601 installs easily. It provides a flexible IP telephony solution delivering excellent voice quality. An intuitive user interface offers dedicated, single-button access to common telephony features. The IP 601's high-resolution display supplies content for call information, directory access, system status, and future applications.   Polycom Soundpoint IP 601 Features & Benefits Six-line phone ideal for users requiring multiple lines and / or advanced features Excellent voice quality with Polycom Acoustic Clarity Technology Backlit 320 x 160-pixel grayscale graphical LCD Expandability to support up to three SoundPoint® IP Expansion Modules Intuitive user interface with single-button access to common telephony features High-resolution graphical LCD with support of Asian characters Flexible powering options (PoE or AC adapter) Standards based, Interoperability with leading IP PBX and Softswitch platforms SIP, MGCP Full-duplex speakerphone with Polycom Acoustic Clarity Technology Advanced functionality, including shared lines, busy lamp field, presence, and XHTML applications Support of up to three SoundPoint IP Expansion Modules Integrated IEEE 802.3af Power over Ethernet (PoE) support 122 Aastra 480I IP Telephone 101.0000 114.95 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-480i-ip-telephone-p-122.html http://www.voipon.co.uk/images/aastra_480i.gif new Availability: In Stock Based on the popular Screen phone platform, the Aastra 480i is an advanced, fully featured, IP screen phone that provides a flexible IP solution designed with interoperability and ease of use in mind. Improved Productivity Maximise productivity while minimising training time with the 480i's user-friendly visual prompts. Know instantly whether someone is available to answer a call with the screen-based Busy Lamp Field on their phone. Manage up to 4 calls with the 480i's call appearance light. The Aastra 480i Gain provides advanced telephone features at your finger-tips, and helps you manage your calls with confidence. Quality Business Feature Set The Aastra 480i offers the same reliability and durability found in all Aastra business telephones. Its sleek, ergonomic design and large screen makes this cost-effective phone a smart addition to any business environment. The built-in speakerphone offers the excellent sound quality demanded by today's businesses. With the 480i, intercom (set paging)* is also available, connecting you and your peers quickly and accurately. Aastra 480i Main Features Enhanced BLF (Busy Lamp Field) - View a phones status (onhook, offhook, ringing, connected) on the 480i's large 8-line backlit display. Go even further with presence information (out to lunch, busy, DND and more). 4 Call Appearance Lights - manage up to 4 calls at the touch of a button. Automatically place callers on hold as you jump between calls Remarkable audio quality - speaker phone with excellent voice delivery. Intercom ability allows set paging within the office Multiple Protocol Support - Tightly integrated to work with a wide variety of IP telephony systems / PBXs Less Wiring - Built in two-port 10/100 Ethernet switch lets you share a connection with your computer. Inline power support eliminates power adapters Protect Your Investment - Firmware upgrades can be installed in the field as standards develop and protocols evolve Technical Specifications Size: 11.43 cm W X 17.78 cm L X 11.43 cm H»Weight: 0.91 kg Headset compatibility: Modular RJ9 connector, compatible with amplified business headsets Handset: Hearing Aid compatible Power: Supports IEEE 802.3af inline power. Universal wall adapter with power is also available for networks that don't supply power Feature Keys 6 customizable softkeys 8 Predefined hard keys for most common call functions such as Transfer, Conference, Intercom etc. Networking Dual 10/100 Mbps switched ethernet ports XML programmable API available for 3rd party applications Sample applications include visual voicemail, directory services Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup Time & date synchronization using SNTP Server provisioned XML based user configuration files User programmable Interactive Services Protocol & Feature Support XML Open XML standard interface for screen based applications H.323MGCP SIP Codecs: G.711 m/A and G.729A (Annex B) configuration. ITU-T H.323 with Business feature phone extensions Gatekeeper routed Support for gatekeeper discovery Configurable up to 4 lines varies by IP PBX or Softswitch solution E. 164 dialing H.450 (hold, message waiting) PSTN call progress tones Presence or BLF information IETF MGCP with business package extensions GUI support Configurable up to 4 lines varies by IP PBX or Softswitch solution Extensions for feature key and display support Automatic Call Agent Address on some systems. Presence of BLF information IETF SIP with Business Feature phone extensions Local feature-rich GUI Call transfer, hold, divert (forward) Called, calling, connected party identification/information Conferencing Speed dial, redial Up to 4 lines/call appearances Presence or BLF information The Aastra 480i no longer requires a PoE converter as it is now shipped with a power supply.   123 Aastra 9112I IP Telephone 51.0000 59.95 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-9112i-ip-telephone-p-123.html http://www.voipon.co.uk/images/aastra_9112i_sml.jpg new Availability: In Stock The Aastra 9112i is a value priced, basic single line IP terminal with speaker phone that offers a flexible, interoperable IP solution which meets VoIP budgets for most applications. Main Features Remarkable Audio - Quality speaker phone with excellent voice delivery. Enhanced BLF (Busy Lamp Field) - View a phones status (onhook,offhook,ringing,connected) on the 9112i screen display.Go even further with presence information (out to lunch,busy,DND and more) Multiple Protocol Support - Tightly integrated to work with a wide variety of IP telephony systems Protect Your Investments - Firmware upgrades can be installed in the field as standards develop and protocols evolve Single 10/100 Ethernet Port Technical Specifications Size: 19.1 cm W X 19.1 cm L X 8.9 cm H Power: Universal wall adapter with power is supplied. Weight: 1.1 kg Headset compatibility: Modular RJ9 connector,compatible with amplified business headsets Handset: Hearing Aid compatible Feature Keys 7 Programmable keys Networking 10/100 Mbps switched Ethernet port Time and date synchronization using SNTP Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup Server provisioned XML-based user configuration files User programmable Interactive Services XML programmable - API available for 3rd party applications Sample applications include - visual voicemail,directory services Protocol and feature support for Aastra 9112i XML SIP H.323 MGCP Codecs: G.711 m/A and G.729A (Annex B) configuration. Open XML standard interface for screen based applications ITU-T H.323 with Business feature phone extensions Gatekeeper routed Support for gatekeeper discovery H.450 (hold,message waiting) E. 164 dialing Single Line Configurable varies by IP PBX or Softswitch solution PSTN call progress tones Presence or BLF information IETF MGCP with business package extensions Aastra GUI support Single Line Configurable varies by IP PBX or Softswitch solution Extensions for feature key and display support Automatic Call Agent Address on some systems. Presence of BLF information IETF SIP with Business Feature phone extensions Local feature-rich GUI Call transfer,hold,divert (forward) Called, calling, connected party identification/information Conferencing Speed dial, redial Single line call appearance Presence or BLF information 9112i Safety and Emissions North American,Australian and European approvals including: FCC Part 15,Subpart B EN55024 and EN55022,Class B AS/NZS CISPR 22 TIA 968-A and CS-03,Issue 8 AS/ACIF S004 CSA 60950-1 NRTL/C TS001 AS/NZS3260 EN60950 124 Aastra 9133I IP Telephone 74.5000 87.95 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-9133i-ip-telephone-p-124.html http://www.voipon.co.uk/images/aastra_9133i_sml.jpg new Availability: In Stock The Model 9133i is an advanced feature multi-line IP Telephone that takes full advantage of VoIP technology by offering a flexible,interoperable solution at anaffordable price. Key Features: Enhanced BLF (Busy Lamp Field) - View a phones status (onhook,offhook,ringing,connected) on the 9133i screen display. Go even further with presence information (out to lunch,busy,DND and more) 3 Call Appearance Lights - Juggle up to 3 calls at the touch of a button. Place callers on hold automatically as you jump between calls Remarkable Audio - Quality speaker phone with excellent voice delivery. Multiple Protocol Support - Tightly integrated to work with a wide variety of IP telephony systems Protect Your Investments - Firmware upgrades can be installed in the field as standards develop and protocols evolve Less Wiring - Built in two-port 10/100 Ethernet switch lets you share a connection with your computer. Inline power support eliminates power adapters Technical specifications Size: 19.1 cm W X 19.1 cm L X 8.9 cm H Weight: 1.1 kg Power: Aastra Supports IEEE 802.3af inline power. Universal wall adapter with power is also available for networksthat don't supply power Headset compatibility: Modular RJ9 connector,compatible with amplified business headsets Handset: Hearing Aid compatible Display: Three Line Adjustable Backlit display Feature Keys: 11 Programmable BLF keys 5 Predefined Keys for Speakerphone,Mute,Transfer,Conference and Redial Networking: Dual 10/100 Mbps switched Ethernet ports Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup Time and date synchronization using SNTP Server provisioned XML based user configuration files User programmable Interactive Services XML programmable API available for 3rd party applications Sample applications include visual voicemail,directory services Available Protocol and feature support: XML H.323 MGCP SIP Aastra 9133i Codecs: G.711 m/A and G.729A (Annex B) configuration. Open XML standard interface for screen based applications ITU-T H.323 with Business feature phone extensions Gatekeeper routed Support for gatekeeper discovery Configurable up to 3 lines varies by IP PBX or Softswitch solution H.450 (hold,message waiting) E. 164 dialing PSTN call progress tones Presence or BLF information IETF MGCP with business package extensions GUI support Configurable up to 3 lines varies by IP PBX orSoftswitch solution Automatic Call Agent Address on some systems. Extensions for feature key and display support Presence of BLF information IETF SIP with Business Feature phone extensions Local feature rich GUI Call transfer,hold,divert (forward) Called,calling,connected party identification/information Speed dial, redial Conferencing - Up to 3 lines/ call appearances Presence or BLF information Safety and Emissions North American,Australian and European approvals including: EN55024 and EN55022,Class B FCC Part 15,Subpart B AS/NZS CISPR 22 TIA 968-A and CS-03,Issue 8 AS/ACIF S004 CSA 60950-1 NRTL/C TS001 AS/NZS3260 EN60950 The Aastra 9133i no longer requires a PoE converter as it is now shipped with a power supply.   125 AVM B1 ISDN Controller 254.6000 268.00 AVM AVM BRI Cards http://www.voipon.co.uk/avm-b1-isdn-controller-p-125.html http://www.voipon.co.uk/images/avm_b1.gif new Availability: In Stock The new AVM ISDN-Controller B1 PCI v4.0 sets new standards for basic-rate ISDN in professional server and workstation applications. Equipped with a high-performance, 50-MIPS StrongT processor, a completely new development by AVM, the ISDN-Controller B1 PCI v4.0 provides a CAPI 2.0 interface for a great number of current operating systems, and features universal compatibility through support for numerous D and B-channel protocols, including Group 3 fax. The active design and the use of bus-mastering DMA ensure a minimal load on the host system and maximum reliability. Technical specifications ISDN plug-in adapter for the PCI bus (PCI 2.1) For BRI lines and PBX extensions (S0 interface) Multitasking StrongT RISC-Processor (50 MIPS at 50 MHz) 1 MB of fast static RAM on board PCI bus-mastering DMA Throughput of 2 x 64 kbit/s and 1 x 16 kbit/s Supported operating systems: Windows Server 2003, Windows XP, XP 64Bit Edition, 2000/NT/Me/98/95/3.x, Novell NetWare 6.x/5.x/4.x./3.12, Linux, OS/2, DOS Loadable D-channel protocols DSS1 [Euro-ISDN], 1TR6, NI-1, 5ESS, and B-channel protocols X.75, T.30, T.70NL, T.90, X.31, V.110, V.120, and more Group 3 fax integrated in the Controller (send at up to 14,400 bit/s per V.17; receive at up to 9600 bit/s per V.29, on both B-channels simultaneously); incl. two-dimensional MR/MMR compression and ECM (Error Correction Mode) Also available: fax polling, DTMF, GSM and support for CAPI 1.1 applications OEM B channel software also loadable Standardized applications interface Common ISDN API (CAPI) 2.0 Full backward compatibility with B1 family hardware and driver software Short card: about 146 x 120 mm; power consumption: about 2 watts Install up to four Controllers in one PC, even in mixed configurations with the AVM ISDN-Controller B1 PCI; B1 v1.3, v2.0, v3.0; B1 PCMCIA and B1 USB CE certification, conformance with the R&TTE Directive (1999/5/EEC) 126 AVM C2 ISDN Controller 448 448.00 AVM AVM BRI Cards http://www.voipon.co.uk/avm-c2-isdn-controller-p-126.html http://www.voipon.co.uk/images/avm_c2.jpg new Please note the AVM ISDN-Controllers C2 has been discontinued. For alternatives please see the quadBRI and B410P . The AVM ISDN-Controllers C2 is the right hardware for server installations with up to two basic-rate ISDN lines. Equipped with a StrongARM SA-110 processor for 270 MIPS and 16 MB of on-board memory, the C2 supports all the ISDN communication services. Its design ensures unrestricted communication with digital or analog stations over all four channels simultaneously. The controller software for the C2 is available for Windows 2000/NT, NetWare and Linux. Technical specifications AVM ISDN-Controller for BRI lines; PCI bus interface One to two BRI lines Active ISDN-Controller with on-board CPU and memory High-performance StrongARM SA-110 CPU with 270 MIPS at 233 MHz 16 MB of SDRAM State-of-the-art technologies, including BGA devices, SDRAM, 3.3 V supply voltage, and more Low power consumption: about 3 W PCI bus-mastering DMA for minimum system load and maximum throughput Suitable for all ISDN services: Internet access Internetworking Remote access File transfer Fax Voice Video Digital communication with other networks Application interface Common ISDN API 2.0: CAPI SoftCompression X.75/V.42bis and channel bundling per CAPI specification CAPI SoftFax: telefax at 2400, 4800, 9600 and 14,400 bit/s CAPI SoftModem: analog modem connections at 1200/75, 2400, 4800, 9600 and 14,400 bit/s Digital protocols: X.75, HDLC transparent, bit-transparent, X.25, ISO 8208 (X.25 DTE-DTE), X.31 Case a/b, T.70, T.90, Mobile ISDN (ISO 3309), V.110, V.120 Analog protocols: Group 3 fax (T.30, V.17, V.29, V.27ter), Group 3 Annex A (Error Correction Mode) and Group 3 F (telefax with two-dimensional data compression) Modem connections with V.21, V.22, V.22bis, V.32, V.32bis and DTMF detection Support for point-to-point lines and leased lines Supplementary Services (such as Hold & Retrieve, Suspend & Resume (TP), 3PTY, ECT, and more) All protocol software downloaded to the controller AVM CoNDIS WAN, AVM TAPI, and AVM ISDN CAPI Port Driver for all supported Windows operating systems Up to four active AVM ISDN-Controllers in one system International approvals Package contents AVM ISDN-Controller C2 CAPI 2.0 driver Software for Windows XP (64-Bit Edition), Server 2003/XP/2000/NT, Linux (inklusive 64-Bit), OS/2, and Novell NetWare 4.x/5.x/6x 2 ISDN cables with RJ-45 plugs, 6 m 1 CD-ROM with driver software, middleware, applications, and more ISDN-Controller C2 manual ISDN-Tools manual 127 AVM C4 ISDN Controller 848 848.00 AVM AVM BRI Cards http://www.voipon.co.uk/avm-c4-isdn-controller-p-127.html http://www.voipon.co.uk/images/avm_c4.jpg new Please note the AVM ISDN-Controllers C4 has been discontinued. For alternatives please see the quadBRI and B410P . The AVM ISDN-Controller C4 is the right hardware for server installations with more than one basic-rate ISDN line. Equipped with a StrongARM SA-110 processor for 270 MIPS and 16 MB of on-board memory, the C4 supports all the ISDN communication services. Its design affords exclusive digital and analog communication features. The controller software for the C4 is available for Windows NT, Windows 2000, Novell NetWare and Linux. Technical specifications AVM ISDN-Controller for BRI lines; PCI bus interface One to four BRI lines Active ISDN-Controller with on-board CPU and memory High-performance StrongARM SA-110 CPU with 270 MIPS at 233 MHz 16 MB of SDRAM State-of-the-art technologies, including BGA devices, SDRAM, 3.3 V supply voltage, and more Low power consumption: about 3 W PCI bus-mastering DMA: minimum system load and maximum throughput Suitable for all ISDN services: Internet access Internetworking Remote access File transfer Fax Voice Video Digital communication with other networks Application interface Common ISDN API 2.0 Digital protocols: X.75, HDLC transparent, bit-transparent, X.25, ISO 8208 (X.25 DTE-DTE), X.31 Case a/b, T.70, T.90, Mobile ISDN (ISO 3309), V.110, V.120 CAPI SoftCompression X.75/V.42bis and channel bundling in conformance with CAPI specification CAPI SoftFax: fax at 2400, 4800, 9600 and 14400 bit/s, including ECM and MR/MMR; CAPI SoftModem: analog modem connections at 1200/75, 2400, 4800, 9600 and 14,400 bit/s Analog protocols: Group 3 fax (T.30, V.17, V.29, V.27ter), Group 3 Annex A (Error Correction Mode) and two-dimensional data compression (MR/MMR) Modem connections with V.2, V.21, V.22bis, V.22, and V.32bis DTMF detection, support for point-to-point lines and leased lines Supplementary Services (such as Hold & Retrieve, Suspend & Resume (TP), 3PTY, ECT, and more) All protocol software downloaded to the controller AVM CoNDIS WAN, AVM TAPI, and AVM ISDN CAPI Port Driver for all supported Windows operating systems Up to four ISDN-Controller C4s in one system International approvals Package contents AVM ISDN-Controller C4 CAPI 2.0 driver Software for Windows XP (64-Bit Edition), Server 2003/XP/2000/NT, Linux (inklusive 64-Bit), OS/2, and Novell NetWare 4.x/5.x/6x ISDN cables with RJ-45 plugs, 6 m CAPI 2.0 drivers for Windows XP, 2000, NT; NetWare and Linux Configuration and diagnostics software AVM ISDN CAPI Port Driver AVM NDIS WAN CAPI Driver AVM ISDN Tools (file transfer programs) 128 Eicon Diva Server V-BRI 327.0000 344.98 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-vbri-p-128.html http://www.voipon.co.uk/images/eicon_v_bri.jpg new Availability: In Stock Diva Server V-BRI Eicon Networks' Diva Server V-BRI is a high-performance, PC-based telephony adapter that provides rich media processing capabilities for two voice channels over an ISDN base rate interface. It is the perfect choice for enterprises looking for small size voice, speech and conferencing platforms to get started, yet allow for easy expansion. Powerful Digital Signal Processors (DSP) - one dedicated to each communication channel, ensures real-time voice processing reducing latency and improving overall system performance. Thus Diva Server V-BRI enables both legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as; voice portals and speech-enabled contact centers. Features Up to 2 voice channels over ISDN BRI interface World-class media processing - voice activity detection, DTMF and tone handling, echo cancellation Efficient integration with speech (ASR and TTS) engines Enhanced Switching and Conferencing support Robust Voice over IP (VoIP) technology built in Powerful onboard CPU and DSPs ensure high performance, offloading server processing Extensive, open and well documented API Tightly integrated with most recent Microsoft Windows 2000, XP and Server 2003 Full Linux support - latest SuSE and RedHat versions Ease of installation guaranteed by Plug and Play conformance Easy upgrade with free software downloads from Eicon 129 Eicon Diva Server V-4BRI 868 868.00 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-v4bri-p-129.html http://www.voipon.co.uk/images/eicon_v_4bri.jpg new Availability: In Stock Diva Server V-4BRI Eicon Networks' Diva Server V-4BRI is a high-performance, PC-based telephony adapter that provides rich media processing capabilities for up to eight voice channels over four ISDN base rate interfaces. It is the perfect choice for enterprises looking for a medium size voice, speech and conferencing platform that can easily scaled to larger configurations. Powerful Digital Signal Processors (DSP) - one dedicated to each communication channel, ensures real-time voice processing reducing latency and improving overall system performance. Thus Diva Server V-4BRI enables both legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as; voice portals and speech-enabled contact centers. Features: Up to 8 voice channels over 4 ISDN BRI interfaces World-class media processing - voice activity detection, DTMF and tone handling, echo cancellation Efficient integration with speech (ASR and TTS) engines Enhanced Switching and Conferencing support Robust Voice over IP (VoIP) technology built in Powerful onboard CPU and DSPs ensure high performance, offloading server processing Extensive, open and well documented API Tightly integrated with most recent Microsoft Windows 2000, XP and Server 2003 Full Linux support - latest SuSE and RedHat versions Ease of installation guaranteed by Plug and Play conformance Easy upgrade with free software downloads from Eicon 130 Junghanns quadBRI PCI ISDN 305.0000 350.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-quadbri-pci-isdn-p-130.html http://www.voipon.co.uk/images/quadbri.jpg new Availability: In Stock The quadBRI PCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world. Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The quadBRI PCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. Multiple quadBRI PCI ISDN cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge B channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. B channels on different cards can be actively switched). Target applications ISDN BRI PBX ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking Requirements CPU 500+ Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 4 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge PCM bus connectors daisy chaining of max. 8 cards 4 dual-color LEDs (layer 1 state indicators) active channel switching (across multiple cardsover the external PCM bus) Point-to-Point (TE / NT) and Point-to-Multipoint(TE / NT) euroISDN protocol stack suitable for 3.3 volts and 5.0 volts 32 bit PCI slots Optional S0 bus power feeding module available for feeding up to 4 cards 131 Junghanns octoBRI PCI ISDN 460.0000 530.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-octobri-pci-isdn-p-131.html http://www.voipon.co.uk/images/octobri.jpg new Availability: In Stock The octoBRI PCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world. Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The octoBRI PCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. Multiple octoBRI PCI ISDN cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge B channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. B channels on different cards can be actively switched). Target applications ISDN BRI PBX ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking Requirements CPU 500+ Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 4 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge PCM bus connectors daisy chaining of max. 4 cards 4 dual-color LEDs (layer 1 state indicators) Active channel switching (across multiple cards over the external PCM bus) Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) EuroISDN protocol stack suitable for 3.3 volts and 5.0 volts 32 bit PCI slots Optional S0 bus power feeding module available for feeding up to 2 cards 132 Sangoma A101 PRI ISDN Card 241.9600 305.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html http://www.voipon.co.uk/images/sangoma_a101_pcix_small.gif new Availability: In Stock Sangoma A101 (AFT card with one channelised T1 port) AFT card with one T1/E1 or fractional T1/E1 port, supporting multiple DS0 channels of HDLC or non HDLC data. Used to support WANPIPE® in multichannel hub configurations, and as T1 voice gateways for PBX systems. The A101 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Operational Modes Data only: T1/E1 and fractional T1/E1, single channel HDLC per line. Can be used as a hub for sub-DS1 remotes. The A101 and A102 can support any configuration of up to 62 DS0s carrying Frame Relay, PPP or HDLC data. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes: Supports Robbed Bit Channel Associated Signalling (CAS) and ISDN PRI. Block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform. Channelised mode supporting individual DMA into voice timeslots plus onboard HDLC support of PRI channel for soft PBX implementations that can use these features. Mixed Voice/Data mode: Combination of router/PBX functions in one server - Asterisk as an option. Both 8 bit (64kbps per channel) and 7 bit (56kbps per channel) board-level HDLC support. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack providing certified Frame Relay, PPP, HDLC and Technical Specifications Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes: T1/E1 and fractional T1/E1, single channel HDLC per line HDLC data Power: 520mA at +5v PCI 32 bit (5v and 64 bit (3.3v) compatible. Temperature range: 0 - 45C. All set-up and configuration is in software or by machine BIOS. DSU/CSU set up entirely in software. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal , Master. Software controlled DSU/CSU test modes. Remote monitoring of card and CSU/DSU operation. Dimensions: 2U Form factor: 120mm x 55 mm. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up), FreeBSD, Open BSD. Higher level protocols Asterisk Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty Three years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CE., Declaration of Conformity Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 RJ 45 Pin-out 133 Sangoma A102D PRI ISDN Card 645.9000 814.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a102d-pri-isdn-card-p-133.html http://www.voipon.co.uk/images/sangoma_a102_pcix_small.gif new Availability: In Stock The A102D PCIx has hardware based echo cancellation across 60 channels The A102 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Operational modes Data only T1/E1 and fractional T1/E1, single channel HDLC per line. Can be used as a hub for sub-DS1 remotes. The A101c and A102c can support any configuration of up to 62 DS0s carrying Frame Relay, PPP or HDLC data. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform. Channelized mode supporting individual DMA into voice timeslots plus on-board HDLC support of PRI channel for soft PBX implementations that can use these features. Mixed Voice/Data mode Combination of router/PBX functions in one server.Both 8 bit (64kbps per channel) and 7 bit (56kbps per channel) board-level HDLC support.WAN data connection is supported by Sangoma's standard WANPIPE® routing stack providing certified Frame Relay, PPP, HDLC and X.25. RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Technical Specification Available as Single T1/E1 port (A101) or Dual T1/E1 port (A102) with daughterboard (as shown in photograph). Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Power: 800mA peak, operational 300mA max at +3.3v or 5v. MTBF: > 1 Million hours. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. Certification FCC Part 15 Class A, FCC Part 68, CE. T echnical certifications in Russia, Malaysia Production quality ISO 9002 Warranty Five years parts and labour. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/ 2000/ XP, FreeBSD, Open BSD,NetBSD, Solaris Voice applications Asterisk, Yate, OPAL Open PBX/IVR Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame Diagnostic tools WANPIPEMON, SNMP, System logs 134 Sangoma A104D PCI PRI ISDN Card 1090.2300 1374.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a104d-pci-pri-isdn-card-p-134.html http://www.voipon.co.uk/images/sangoma_a104_pcix_small.gif new Availability: In Stock The A104D PCIx has hardware based echo cancellation across 120 channels The sandwich DSP card on the "d" model features Octasic's certified carrier-grade algorithms providing carrier grade echo cancellation and Voice Quality Enhancement (VQE) functions. Supporting 32-672 channels G.168-2002 echo cancellation, a minimum of 1024 taps for a 128ms tail/channel on all channel densities, the system also supports Octasic music protection, acoustic echo control and adaptive noise reduction. The "d" model also features on-board DTMF decoding and tone recognition. The A104 is the, quad port version of Sangoma's range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A104 provides full speed 132 Mbps PCI bus transfer with FPGA and DSP based processing to unload the host CPU in demanding environments such as soft PBX/IVR voice applications. Compatible with both the 3.3v and 5v PCI bus, A104 cards operate in all commercially available motherboards sharing IRQs properly with themselves and all other PCI compatible devices, so you never have to worry about hardware compatibility issues. Like all the Sangoma AFT Series, the A104 is field upgradeable to take advantage of the hardware and software improvements as they become available. Optional Modes The A104 and drivers fully support TDM voice gateways for the Asterisk® , Yate® , OPAL® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. The A104 and drivers fully support TDM voice gateways for the Asterisk® , Yate® , OPAL® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDACT (patent pending) technology is integrated to drastically reduce the load due to software echo cancellation, which is the largest component of CPU load in a typical soft PBX system. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Full channelized mode to act as major network hub for sub-DS1 remotes. The A104 can support any configuration of up to 124 remote 64kbps connections carrying Frame Relay, PPP or HDLC data. Timeslots can be concatenated to support remote fractional T1/E1/J1 sites in any combination. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Mixed Voice/Data mode: Robust combination of router/PBX functions in one server. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack , completely independently of TDM voice application for total system reliability. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Technical Specifications Quad port T1/E1/J1 card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features related to voice and/or data to be added when they become available. DSP card on the A104d: o G.168-2002 echo cancellation in hardware o 1024 taps/128ms tail per channel on all channel densities o DTMF decoding and tone recognition Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up). Higher level protocols Asterisk Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty FIVE years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 135 Sangoma A104 PCI PRI ISDN Card 686.2900 865.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a104-pci-pri-isdn-card-p-135.html http://www.voipon.co.uk/images/sangoma_a104_pcix_small.gif new Availability: In Stock Sangoma A104 (AFT card with four channelised T1 ports) AFT card with four T1/E1 or fractional T1/E1 port, each supporting a single channel data stream. A general purpose card supporting all applications including WANPIPE?. The A104 is an updated, quad port version of Sangoma's range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. Based on bus mastering PCI technology supported by a ring-buffer DMA architecture, the A104 provides full speed 132 Mbps PCI bus transfer with minimal real-time processor load. This provides optimal performance in demanding environments such as soft PBX/IVR voice applications (eg. Asterisk?). Operational Modes Data only: T1/E1 and fractional T1/E1, single channel HDLC per line. Full channelised mode to act as major network hub for sub-DS1 remotes. The A104 can support any configuration of up to 124 remote 64kbps connections carrying Frame Relay, PPP or HDLC data. Timeslots can be concatenated to support remote fractional T1/E1/J1 sites in any combination. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes: The A104 and drivers fully support TDM voice gateways for the Asterisk? PBX project, as well as other Open Source and proprietary PBX/Switch/IVR applications. Mixed Voice/Data mode: Robust combination of router/PBX functions in one server. WAN data connection is supported by Sangoma's standard WANPIPE? routing stack , completely independently of TDM voice application for total system reliability. WANPIPE? supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Technical Specifications Quad port T1/E1/J1 card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features related to voice and/or data to be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up) Higher level protocols Asterisk Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty Three years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CE., Declaration of Conformity Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 RJ 45 Pin-out 143 Digium IAXY Analog Adapter 45.0000 54.99 Digium IAXY Analog Adaptor http://www.voipon.co.uk/digium-iaxy-analog-adapter-p-143.html http://www.voipon.co.uk/images/iaxy.jpg new Availability: In Stock The Digium S101I, affectionately known as the IAXy, takes Asterisk from the PC to the CPE. The IAXy provides a single, fully featured FXS interface with an Ethernet back-end, speaking the Asterisk-native IAX protocol, at a highly competitive price. The IAXy is aimed at Voice Over Broadband and Internet Telephone Service Providers. The IAX protocol provides complete NAT transparency, enabling full operation behind NAT and PAT firewalls. This includes the ability to robustly transfer calls between endpoints, allowing on-net calls to be moved off of a service provider's network for better quality and lower cost. Target Applications Internet Telephony Service Provider Remote PBX Extensions Wireless Phone Service with External Bridge Features Auto Upgrade Remote Reprovisioning Caller ID Call Waiting Cancel Call Waiting (*70) Caller ID on Call Waiting Caller ID Disable, Enable (*67, *82) Three-way Calling Call Transfer Blind Transfer Call Parking VMWI (Voice Mail Waiting Indicator) Mute Rx on-HookPulse DialCall Hold Environmental Conditions Operation Range: 0° to 50° C, 32° to 122° F Storage Range: -20° to 65° C, 4° to 149° F Humidity: 10-90% non-condensing VoIP Codecs &#956;law (G.711) ADPCM VoIP Protocol Inter-Asterisk eXchange (IAX) Telephone Connector: RJ11 Ringer Equivalence Number (REN): 5 at 1500 ft. Power Requirements 6V DC, 1000mA Regulated Switching Tip Positive 3-3.8mm outer- diameter, 1-1.3mm inner- diameter connector, locking tip, 11.5mm length 145 Linksys PAP2T Analog Telephone Adapter 29.0000 34.00 Linksys Linksys Analog Adaptors http://www.voipon.co.uk/linksys-pap2t-analog-telephone-adapter-p-145.html http://www.voipon.co.uk/images/spa_2002.jpg new Availability: In Stock The Linksys PAP2T Internet Phone Adapter enables high-quality feature-rich VoIP (voice over IP) service through your broadband Internet connection. Just plug it into your home Router or Gateway and use the two standard telephone ports to connect analog phones or use one of the ports for a fax machine. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and a reliable fax connection, even while using the Internet at the same time. With Internet telephony, along with low domestic and international phone rates, an impressive array of special telephone features are available. Choose your preferred free local dialing area code, regardless of where you live. Or add a virtual telephone number in any area code, forwarded to your Internet phone. You can even add a toll-free number. The Linksys Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephony service provider, such as Caller ID, Call Waiting, Voicemail, Call Forwarding, Distinctive Ring, and much more. Features Benefits Enables feature-rich telephone service over your broadband Internet connection Two standard telephone ports for analog phones or use one of the ports for a fax machine, each with an independent phone number High quality, clear sounding voice service simultaneous with Internet use Compatible with all common telephone features: Caller ID, Call Waiting, Voicemail, etc. Telephony Two voice ports (RJ-11) for analog phones or Fax machines Impedance Agnostics - 8 Configurable Settings Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Caller ID with Name/Number (Multi-national Variants) Caller ID Blocking Call Forwarding: No answer, Busy, All Do Not Disturb Call Transfer Three-way Conference Calling with Local Mixing Message Waiting Indication - Visual and Tone Based Call Return Call Back on Busy Call Blocking with Toll Restriction Delayed Disconnect Distinctive Ringing - Calling and Called Number Off-hook Warning Tone Selective/Anonymous Call Rejection Hot line and Warm Line Calling Speed Dialing of 8 Numbers/Addresses Music on Hold Package Contents PAP2T Phone Adapter Unit Power Adapter RJ-45 Ethernet Cable Quick Installation Guide Linksys SPA Series Analogue Telephone Adapter (ATA) Comparison Chart Model Service Lines Active Calls 3-Way Call Conferences Network Ports PSTN (FXO) Ports Phone (FXS) Ports SPA1001 2 2 1 1 0 1 PAP2T 2 4 2 1 0 2 SPA2102 2 4 2 2 0 2 SPA3102 2 3 1 2 1 1 SPA400 N/A N/A N/A 1 4 0 146 Grandstream Handytone 286 Analog Adaptor 25.6500 27.00 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-286-analog-adaptor-p-146.html http://www.voipon.co.uk/images/handytone_286.jpg new Availability: In Stock Grandstream HandyTone 286 ATA-286 is an award-winning next generation VoIP analog telephone adapter based on industry open standards. Built upon innovative technology, Grandstream HandyTone ATA-286 features market leading superb sound quality, compact size, and rich functionalities at highly-affordable price. Main Features of the Grandstream HandyTone 286: Full Feature Voice/Fax-over-IP Ultra-Compact and Lightweight Universal Plug and Dial Extremely Cost Affordable Features Support SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols Support NAT traversal using IETF STUN and symmetric RTP (compatible with Cisco's ATA-186, etc) Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality Provides 1 LAN port and 1 FXS interface for any analog telephones, cordless phones, and fax machines Support transparent Fax pass-through and in the future T.38 (pending) Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products Advanced and patent pending adaptive jitter buffer control, packet delay and loss concealment technology Support popular codecs including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, and G.728. Dynamic negotiation of codec and voice payload length Support standard voice features such as Caller ID Display or Block, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, early dial, click-to-dial Support acoustic echo cancellation, voice mail with indicator, downloadable ring tone (pending) Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) Support DIGEST authentication and encryption using MD5 and MD5-sess. Provide easy configuration thru manual operation (attached analog phone keypad and voice prompt, Web interface) or personalized automated provisioning via central configuration file for mass deployment. Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs. Support for fail-over SIP server and DNS server (pending) Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 147 Grandstream Handytone 386 Analog Adaptor 31.5000 32.95 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-386-analog-adaptor-p-147.html http://www.voipon.co.uk/images/handytone_386.jpg new Availability: In Stock Grandstream HandyTone 386 is an "All-In-1" VoIP IAD based on SIP standard. Built upon Grandstream's innovative technology, HandyTone 386 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability. Main Features of the Grandstream HandyTone 386: Full Feature Voice/Fax-over-IP Ultra-Compact and Lightweight Universal Plug and Dial Extremely Cost Affordable 2 FXS Ports (Phone or Fax) PSTN Pass-Through port Compact, Lightweight and Highly Affordable Features Support SIP (RFC3216), TXP/UDP/IP, HTTP, ICMP, ARP/RARP, DNS (SRV AND A-record), DHCP, PPPoe, STUN, TFTP, Advanced Digital Signal Processing (DSP) to ensure superb high fidelity voice quality Support advanced jitter buffer control, packet delay and loss concealment technology Support various popular voice codecs including G.711, G723, G.729A/B, G.726, G.728, and ILBC Support integrated routers, DHCP server, and NAT with DMZ and port forwarding capabilities on all HandyTone 4xx models. HandyTone 486 also supports bridge mode Support caller ID display or block, hold, call waiting, Flash, call transfer (blind or attended, call forward (unconditional, on busy, or no answer), 3 way conference Support dial plan, In-band and out-of-band DTMF (RFC2833, SIP INFO, In-audio, polarity reversal, T.38 fax, VAD/CNG/AGC, and G.168 line echo cancellation (32 ms tall length Support digest encryption and authentication using MD5 and MD5-sess Support layer 2 (802.1Q VLAN, 802.1p) and layer 3 (DiffServ, Tos) QoS Support automated intellifent NAT traversal without end user manipulating firewall configuration Support remote and automated provisioning and monitoring, as well as remote firmware upgrade using tftp/http Support device configuraton using standard telephone and voice prompt, Web Browser, or secure (128-bit AES encrypted) central configuration file or mass deployment Stylish and compact design )size of a credit card or wallet), with small universal supply Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 148 Grandstream Handytone 486 Analog Adaptor 35.1500 37.00 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-486-analog-adaptor-p-148.html http://www.voipon.co.uk/images/handytone_486.jpg new Availability: In Stock Grandstream HandyTone 486 is an "All-In-1" VoIP IAD based on SIP standard. Built upon Grandstream's innovative technology, HandyTone 486 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability. Key Features Support SIP2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, TFTP, etc. Built-in router, NAT and Gateway Powerful digital signal processing (DSP) to ensure superb audio quality; advanced adaptive jitter control and packet loss concealment technology Ultra compact (wallet size) and lightweight design, great companion for travelers Support various vocoders including G.711 A-/U-law, G.723.1, G.729A/B, G.728, G.726, iLBC. Support Caller ID/Name display or block, Hold, Call Waiting, Call Transfer, Call Forward, Flash, Three-Way Conferencing (pending) Support customizable ring tone (pending) Fax pass through and T.38 (pending) Support silence suppression and VAD, AGC, and echo cancellation (G.168) Support standard encryption and authentication (DIGEST using MD5 and MD5-sess) Support layer-2 (802.1Q VLAN, 802.1p) and layer-3 (DiffServ, ToS) QoS Support automated NAT traversal without manual manipulation of firewall/NAT Support remote automated provisioning and software upgrade even through firewall/NAT to enable "zero configuration" and "plug-and-dial" for end users Support device configuration via built-in IVR, Web browser or central configuration file Support DNS SRV Look Up SIP Server Fail Over (pending) Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 149 Grandstream Handytone 488 Analog Adaptor 45.6000 48.00 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-488-analog-adaptor-p-149.html http://www.voipon.co.uk/images/handytone_488.jpg new Availability: In Stock Grandstream HandyTone 488 is an "All-In-1" VoIP IAD based on SIPstandard. Built upon Grandstream's innovative technology, HandyTone 488 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability. Main Features of the Grandstream HandyTone 488: All-In-1 VoIP IAD Built-in Router, NAT and Gateway 1 FXO Gateway 1 FXS Port Compact, Lightweight and Highly Affordable * FXO port allows local and remote PSTN-to-VoIP call origination and VoIP-to-PSTN call termination, automated call routing through PSTN, and power fail-over. PSTN Pass-through allows local PSTN line service for income/outbound call and failover. Featurs Support SIP (RFC3216), TXP/UDP/IP, HTTP, ICMP, ARP/RARP, DNS (SRV AND A-record), DHCP, PPPoe, STUN, TFTP, Advanced Digital Signal Processing (DSP) to ensure superb high fidelity voice quality Support advanced jitter buffer control, packet delay and loss concealment technology Support various popular voice codecs including G.711, G723, G.729A/B, G.726, G.728, and ILBC Support integrated routers, DHCP server, and NAT with DMZ and port forwarding capabilities on all HandyTone 4xx models. HandyTone 486 also supports bridge mode Support caller ID display or block, hold, call waiting, Flash, call transfer (blind or attended, call forward (unconditional, on busy, or no answer), 3 way conference Support dial plan, In-band and out-of-band DTMF (RFC2833, SIP INFO, In-audio, polarity reversal, T.38 fax, VAD/CNG/AGC, and G.168 line echo cancellation (32 ms tall length Support digest encryption and authentication using MD5 and MD5-sess Support layer 2 (802.1Q VLAN, 802.1p) and layer 3 (DiffServ, Tos) QoS Support automated intellifent NAT traversal without end user manipulating firewall configuration Support remote and automated provisioning and monitoring, as well as remote firmware upgrade using tftp/http Support device configuraton using standard telephone and voice prompt, Web Browser, or secure (128-bit AES encrypted) central configuration file or mass deployment Stylish and compact design )size of a credit card or wallet), with small universal supply Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 150 Grandstream Handytone 496 Analog Adaptor 42.0000 43.95 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-496-analog-adaptor-p-150.html http://www.voipon.co.uk/images/handytone_496.jpg new Availability: In Stock Grandstream HandyTone 496 is an "All-In-1" VoIP IAD based on SIP standard. Built upon Grandstream's innovative technology, HandyTone 496 features superb sound quality, rich functions, high degree of integration, ease of use, compact size, and ultra-affordability. Main Features of the Grandstream HandyTone 496: All-In-1 VoIP IAD Built-in Router, NAT and Gateway 2 FXS Port Compact, Lightweight and Highly Affordable Typical Application of a Grandstream HandyTone-496 * FXO port allows local and remote PSTN-to-VoIP call origination and VoIP-to-PSTN call termination, automated call routing through PSTN, and power fail-over. PSTN Pass-through allows local PSTN line service for income/outbound call and failover. Features Support SIP (RFC3216), TXP/UDP/IP, HTTP, ICMP, ARP/RARP, DNS (SRV AND A-record), DHCP, PPPoe, STUN, TFTP, Advanced Digital Signal Processing (DSP) to ensure superb high fidelity voice quality Support advanced jitter buffer control, packet delay and loss concealment technology Support various popular voice codecs including G.711, G723, G.729A/B, G.726, G.728, and ILBC Support integrated routers, DHCP server, and NAT with DMZ and port forwarding capabilities on all HandyTone 4xx models. HandyTone 486 also supports bridge mode Support caller ID display or block, hold, call waiting, Flash, call transfer (blind or attended, call forward (unconditional, on busy, or no answer), 3 way conference Support dial plan, In-band and out-of-band DTMF (RFC2833, SIP INFO, In-audio, polarity reversal, T.38 fax, VAD/CNG/AGC, and G.168 line echo cancellation (32 ms tall length Support digest encryption and authentication using MD5 and MD5-sess Support layer 2 (802.1Q VLAN, 802.1p) and layer 3 (DiffServ, Tos) QoS Support automated intellifent NAT traversal without end user manipulating firewall configuration Support remote and automated provisioning and monitoring, as well as remote firmware upgrade using tftp/http Support device configuraton using standard telephone and voice prompt, Web Browser, or secure (128-bit AES encrypted) central configuration file or mass deployment Stylish and compact design )size of a credit card or wallet), with small universal supply Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 151 Grandstream Budgetone 101 IP Telephone 29.7000 33.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-budgetone-101-ip-telephone-p-151.html http://www.voipon.co.uk/images/budgetone_101.gif new Availability: In Stock The Grandstream BudgeTone 101 IP Phone is an award-winning VoIP network phone based on industry open standards. Built upon innovative technology, Grandstream IP phones offer market-leading superb sound quality and rich functionality at a very affordable price. The Grandstream BudgeTone 101 MKII, released in November 2005 now includes a headset port. Key Features & Specification Supports SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, PPPoE protocols Supports symmetric RTP and NAT traversal via STUN and external media proxy Interoperable with various 3rd party SIP end user devices, Proxy/Registrar/Server and gateway products Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality Advanced adaptive jitter buffer control, packet delay & loss concealment technology Supports popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, wide-band G.722 and iLBC. Dynamic negotiation of codec and voice payload length Supports standard voice features such as Caller ID Display or Block, Call Waiting, Hold, Mute, Transfer, Forward, FLASH, in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), Dial Plans, off-hook auto dial, early dial, click-to-dial Supports 3-way conferencing, full duplex hands-free speakerphone, redial, call log, volume control, voice mail indicator Provides easy configuration through manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment. Supports for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Supports NAT-friendly remote software upgrade capability (via tftp and http) even from behind firewalls/NATs. Downloadable music ringing tones Support for fail-over SIP server and DNS server Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) Supports DIGEST authentication and encryption using MD5 and MD5-sess Headset jack Open Standards Compatible Superb Audio Quality World's Most Cost Affordable Incredible Functionality 152 Grandstream Budgetone 102 IP Telephone 46.0000 48.95 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-budgetone-102-ip-telephone-p-152.html http://www.voipon.co.uk/images/budgetone_102.gif new Availability: The Grandstream Budgetone 102 has been replaced by the Grandstream Budgetone 200 The Grandstream BudgeTone 102 IP Phone is an award-winning next generation VoIP network phone based on industry open standards. Built upon innovative technology, Grandstream VoIP phones feature market leading superb sound quality and rich functionality at an ultra-affordable price. The Grandstream BudgeTone 102 MKII, released in November 2005 now includes a headset port. Key Features Supports SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, PPPoE protocols Supports symmetric RTP and NAT traversal via STUN and external media proxy Interoperable with various 3rd party SIP end user devices, Proxy/Registrar/Server and gateway products Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality Advanced adaptive jitter buffer control, packet delay & loss concealment technology Supports popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G .726, G.728, wide-band G.722 and iLBC. Dynamic negotiation of codec and voice payload length Supports standard voice features such as Caller ID Display or Block, Call Waiting, Hold, Mute, Transfer, Forward, FLASH, in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), Dial Plans, off-hook auto dial, early dial, click-to-dial Supports 3-way conferencing, full duplex hands-free speakerphone, redial, call log, volume control, voice mail indicator Supports Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) Supports DIGEST authentication and encryption using MD5 and MD5-sess Provides easy configuration through manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment. Supports for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Supports NAT-friendly remote software upgrade capability (via tftp and http) even from behind firewalls/NATs. Downloadable music ringing tones Support for fail-over SIP server and DNS server Headset jack Open Standards Compatible Superb Audio Quality Rich Functionality World's Most Cost Affordable 153 Utstarcom F1000G Wireless IP Telephone 79.0000 79.99 UTstarcom UTstarcom IP Telephone http://www.voipon.co.uk/utstarcom-f1000g-wireless-ip-telephone-p-153.html http://www.voipon.co.uk/images/f1000.gif new Availability: In stock The residential F1000G Wireless IP Telephone is a revolutionary device that expands the reach of VoIP communications. It provides consumers a new cost effective way to communicate, with great features such as 3-way Calling, Call Waiting, Call Transfer and many popular features. Key Features Bar Type Design 802.11b/g Wireless SIP Messaging TFTP/HTTP Provsioning VoIP Supported: - SIP / DHCP 3-Way Calling Call Waiting / Call Transfer / Call Forwarding Technical Profile: Dimensions: 11 x 4.5 x 2.2cm Weight: 111g Battery Li-ion DC 3.6V 1500mAh Talk time Up to 4 hours (Approx.) Charging time 2 ~ 3 hours Standby time 80 ~ 100 hours Voice Processing* G.711u,G.711a,G.729a/b, G.726 Comfort noise generation (CNG) Voice activity detection (VAD) Adaptive jitter buffer Echo cancellation *Requires network support 154 Power over Ethernet Converter 17.5 17.50 Aastra IP Telephones http://www.voipon.co.uk/power-over-ethernet-converter-p-154.html http://www.voipon.co.uk/images/aastra_poe-sml.jpg new Availability: In Stock POE (Power-Over-Ethernet) Converter The POE (Power-Over-Ethernet) Converter is no longer required for the Aastra 480i and the 9133i. However these are still available for purchase in the event one might be needed. Features include: - Self-contained Power Injector up to 15W - 100-250 VAC Universal Input - Desktop Style, Single Output - Regulated Output With Low Ripple - Detection Collision Avoidance - CE Compliant 155 Redfone foneBRIDGE T1/PRI-to-Ethernet Bridge 1374.95 1374.95 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge-t1pritoethernet-bridge-p-155.html http://www.voipon.co.uk/images/redfone_bridge.gif new Please note the foneBRIDGE has been discontinued. For an alternative please see the foneBRIDGE2 Quad T1/E1 foneBRIDGE is a quad T1/PRI-to-Ethernet Bridge. It is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP systems. Designing, building, and installing an Asterisk system has never been this easy! foneBRIDGE eliminates the need to install proprietary TDM (PRI/T1) hardware cards in approved/compatible server configurations. Instead, foneBRIDGE terminates T1 and/or PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities. foneBRIDGE provides an economical means of buliding and maintaining a redundant Asterisk solution. Instead of purchasing a TDM board for each redundant server, multiple redundant servers can share a single foneBRIDGE. If the foneBRIDGE itself fails, replacing it is far simpler that replacing a TDM card in an active server.   Promotes Hardware Independance Install Asterisk on commodity (white box) servers. Reduce CPU and resource load. Reduces Wiring Install foneBRIDGE(s) at trunk termination and supply a single Ethernet connection to server location(s). Uses Power over Ethernet (PoE) - no AC adapter or brick required. Enables Cost-Effective Redundant Solutions Share trunk resources among multiple redundant servers. Install multiple foneBRIDGEs on a network. 156 Phone Line cable RJ11 to BT Plug 0.95 0.95 VoIP Accessories http://www.voipon.co.uk/phone-line-cable-rj11-to-bt-plug-p-156.html http://www.voipon.co.uk/images/rj11.jpg new Availability: In Stock Phone Line cable RJ11 to BT @ 1.5m 157 Draytek Vigor2800 0.0000 130.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2800-p-157.html http://www.voipon.co.uk/images/draytek_2800.jpg new Availability: Discontinued The Draytek 2800 series has been discontinued, and replaced by the 2820 series. The replacement for the 2800 is the Draytek Vigor 2820 .  The DrayTek Vigor2800 is DrayTek's flagship ADSL router- a combined ADSL router for Internet access that also houses a firewall, VPN device and an Ethernet switch. The DrayTek Vigor 2800 includes features such as VLAN, QoS, content filtering, VPN support for up to 16 tunnels and a USB printer port and support for all current ADSL technologies. Feature Highlights: Easy Internet-sharing via ADSL, ADSL 2 connection Robust firewall to help protect your network from external attacks Powerful VPN facilities provide deployment of linked branch offices and teleworkers Comprehensive business features such as VLAN, Bandwidth Control, Advanced Content Filtering Wireless LAN data rate up to 108Mbps with Super G technology Routing Features Internet Protocols : PPPoA, PPPoE Static routing and Dynamic Routing with RIP v1 and v2 DHCP Server, DHCP Client, DHCP Relay, DHCP over IPSec DNS Cache/Proxy Configurable MTU Size Dynamic DNS Firewall & Security Features Stateful Packet Inspection Firewall Packet filtering based on port, source IP address, destination IP address, MAC address, (ICMP/TCP/UDP) DoS, DDos attacks prevention (IP Spoofing, Land Attack, Smurf Attack, Ping of Death, TCP SYN flooding) E-mail alert and logging via syslog Disable Firewall Option Content Filtering Instand Messenger/P2P blocking URL key word blocking Java applet, cookies, Active X, compressed, executable, multimedia file blocking Time schedule VPN (Virtual Private Network) Features Support up to 32 simultaneous tunnels High-performance IPSec 3DES encryption MPPE, DES (56-bit) and 3DES (168-bit) encryption, AES (256-bit) encryption Authenication MD5 & SHA-1 Diffie-Hellman Group 768-bit & 1024-bit Key Management: Auto Internet Key Exchange (IKE) w/ Perfect Forward Secrecy (PFS) PKI (X.509) certificate support Operation Mode: Main & Aggressive Single-session Virtual Private Network (VPN) Pass-through (IPSec, L2TP), PPTP Radius Support for dial-in teleworker profiles NAT (Network Address Translation) Features Simultanous Non-NAT & NAT mode Many-to-One (NAT) Many-to-Many (Multi-NAT) Full Routing (Non-NAT) Applications & Gaming Features Port Forwarding UPnP Support Single DMZ Support Management Features Web Based Interface (HTTP/HTTPS) Simple Installation Wizard CLI (Command Line Interface, Telnet/SHH) System performance and status monitoring Built-in Diagnostic Function Administration access control Remote Management via Web services Syslog Support Configuration backup/restore Firmware upgrade via TFTP/FTP SNMP management MIB-II Support Bandwidth Control and Quality of Service Wireless rate-control Wired & Wireless VLAN Support 4 VLANs Class-based bandwidth guarantee by user-defined traffic categories Support of four priority-levels Support of DiffServ Codepoint classifying Printer Server USB Printer Server USB Port Version 1.1 Compatible with most printers with a USB Port Unidirectional Only LAN Ports 4 RJ-45 10/100 Ethernet switch Auto Sensing / Manual Selection Auto Uplink WAN Ports RJ-11 ADSL Port Built-in ADSL 2/2+ Modem G.dmt: 8Mbps downstream, 832Kbps upstream G.lite: 1.5Mbps downstream, 512Kbps upstream ADSL2: 12Mbps downstream, 1Mbps upstream ADSL2+: 24Mbps downstream, 1Mbps upstream ISDN Bri Port Package Contents Draytek Vigor 2800 Firewall VPN ADSL 2 Router UK Power adapter RJ11 Telephone Cable RJ45 3m Ethernet cable Installation guide 158 Draytek Vigor2800G 155 155.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2800g-p-158.html http://www.voipon.co.uk/images/draytek_2800G.jpg new Availability: Discontinued The Draytek 2800 series has been discontinued, and replaced by the 2820 series. The replacement for the 2800G is the Draytek Vigor 2820n . Product Highlights Combination ADSL Modem, router, firewall, print-server and Wireless Access Point Compatible with all UK ADSL Lines Virtual LAN (VLAN) Facility Ethernet Port Throttling Print Server Capability Internet Firewall Internet Content Filtering VPN Facilities Microsoft uPnP Compliant SNMP & Syslog Dynamic DNS RIP & Static Routing 802.11g Compliant Wireless Access The Vigor 2800G router builds on the great specification of the Vigor2600 with the addition of a built-in Wireless Access Point. This wireless interface enabled wireless connection of PCs and supports the new 802.11g high-speed wireless protocol, which is up to 6 times faster than the previous 802.11b protocol, providing a total wireless bandwidth of up to 54Mb/s. Twin extra-gain aerials ensuring maximum coverage range and signal diversity (high-gain aerials are available as an optional extra). The Vigor2600G also has a built-in 4-port Ethernet switch (expandable) for wired devices so that you can connect wired 10/100BaseT PCs too. The Vigor 2800G comes with the same highly-regarded pedigree as the Vigor2600We ADSL router, but with the higher speed wireless access point and other new features now available. Content Filtering The Vigor 2800G also helps protect against internal Internet 'abuse' with its content filter which can block specified sites, e.g. common email sites or keywords within URLs. Additionally, you can block Java/ActiveX applet download, as well as HTML download of specific file types (e.g. ZIP, EXE, multimedia etc.) Virtual LAN (VLAN) & Bandwidth Throttling The Vigor 2800G's VLAN facility enables you to segment each of the router's four RJ45 Ethernet ports, so that each is a separate virtual LAN. You can create VLAN groups which include or exclude any of the ports so that groups, departments and companies can communicate with each other, or not. For example, two companies could share the same broadband feed, without having access to each other's networks. For more details of VLAN, see here. The 'Bandwidth Throttling' feature lets you set a maximum throughput for each of the Vigor's four Ethernet ports, which can prevent a particular user (or segment) from taking all of your bandwidth. Print Server The USB port on the back of the router allows you to connect most standard USB based printers and then print to them from any Windows98SE/XP/2000 PC, using built-in O/S support from any application, thus not needing to have a particular PC provide the printer sharing to its peers. Enhanced Wireless Security The Vigor 2800G supports industry standard WEP encryption. WEP uses a shared key (common to all users) so WEP alone, should someone find out your common key, can be compromised. Therefore, the Vigor2600G allows you to select additional security methods for wireless clients. 802.1x is an authentication method now supported by WindowsXP whereby a wireles client must enter a unique username and password before wireless access is granted. You can add and delete individual users as required. In addition, the Vigor2600G can add "VPN over WLAN" to increase the level of wireless encryption, using DES/3DES encryption and 256-bit WPA encryption will be supported later (feature est. early 2004, subject to change). Finally, you can lock the router down further so if the unique hardware ('MAC') address of the wireless client is not in the 'allow' list, the client is also denied access.     159 Linksys SPA-941 IP Telephone 56.5500 58.90 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa941-ip-telephone-p-159.html http://www.voipon.co.uk/images/spa-941.jpg new Availability: In Stock This is a SIP based IP phone from Linksys/Sipura. The SPA-941 is suitable for use with SIP based IP PBX systems, or together with a SIP service provider. The SPA-941 phone is the first VoIP Telephone from Linksys since they bought Sipura. The SPA-941 is an affordable, full featured, stylish IP telephone with multiline appearances, graphical display, speakerphone, headset jack and message waiting indicator with a dedicated message retrieval button. Key features Stand alone SIP Business Class Telephone Supports 2 SIP lines, upgradable to 4 Cost Effective Call Features With an intuitive, easy to operate user interface SPA-941 supports standard telephone features, including: Do not disturb Call hold Call transfer Three party conferencing Call History Missed Calls Dialed Calls Received Calls Address book Off-hook dialing Speed dialing Stylish and functional in design, the SPA- 941 can be used in SOHO, enterprise, small to medium businesses together with service offerings including IP PBX, hosted IP telephony and IP Centrex. The SPA-941 is simple to use but as with all Sipura/Linksys products, there are a complete set of configuration options for the advanced user.   Linksys SPA Series VoIP Telephone (SIP) Comparison Chart SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernt Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 160 Atcom AT-320 IAX IP Telephone 40.0000 49.00 Atcom Atcom IAX IP Telephone http://www.voipon.co.uk/atcom-at320-iax-ip-telephone-p-160.html http://www.voipon.co.uk/images/atcom-320.jpg new Please note the Atcom AT320PD has been discontinued. For an alternative please see the Atcom AT530. The Atcom AT-320PD IP Phone, one of the Atcom AT-320 series IP Phones, is an advanced, fully featured IP Phone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable price with support for all the major protocols including IAX2 through interchangeable and user upgradeable firmware releases. Product Features IAX2, H.323 V4, MGCP, SIP and Net2phone private protocol Power over Ethernet Audio codec G.711, G.723, G.729 DHCP support for LAN or Cable modem PPPoE support for ADSL or Cable modem Set phone by HTTP web browser (IE6.0) or Telnet Upgrade by FTP VAD (Voice active detect) CNG (Comfort noise generation) Dynamic voice jitter buffer G.168/165 compliant 16ms echo cancellation Tone generation and Local DTMF re-generation according with ITU-T E.164 dial plan and customized dial rules 100 entries for quick dial 80 entries each for missed calls, answered calls and dialled calls Adjustable volume for both handset and speaker Voice prompt Hotline Electrical Voltage: DC 12V~24V Power adapter: DC 12V/450mA Network interface: 2 RJ-45 Ethernet connector Power over Ethernet: 802.3af Dimensions 200×195×87mm (L × W × H) CE Approved 161 Rhino 24 Port FXO T1 Channel Bank 1165.5000 1295.00 Rhino Rhino Channel Banks http://www.voipon.co.uk/rhino-24-port-fxo-t1-channel-bank-p-161.html http://www.voipon.co.uk/images/rhino_24_fxo.jpg new Availability: In Stock Managing your telecommunication needs has never been easier than with Rhino products. Rhino satisfies the needs of any T1 channel bank application, no matter how stringent the requirement. Unique Rhino features like real-time T1 status on our four line by 40 character (4x40)LCD display, or automatic, hands-off configuration utilizing artificial intelligence software, and crystal clear audio quality proves that Rhino products are in a top class of their own. Knowing that the Rhino is ready to perform means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost T1 channel banks including FXS,FXO, or mixed mode FXS/FXO analog interfaces. As a bonus, every system comes with our standard fractional V.35 data port. Add the Rhino modular, internal power supply system to the list and Rhino crushes the competition. Using Asterisk? Rhino channel banks allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your T1 applications. Features Asterisk* soft PBX tested and ready! 4x40 LCD real-time status display shows independent AB bit signaling Automatic T1 configuration using arti-ficial intelligence software, no config-uration switches CSU auto detects T1modes: D4 or ESF, AMI or B8ZS T1 CSU via RJ48C, with line build out programmability Internal power supply module, no external components FLASH based system configuration V.35 fractional data interface, user selectable to 56K or 64K Self diagnostics, verbose error reporting, loop back modes Single channel configuration, each channel may have a different analog start mode, signaling protocol, line current, or gain Telco network uptime and downtime for T1 history Remote connection via RS-232c or Ethernet (optional) via Windows based graphical user interface Field software upgradable 19", 23" or wall mounting kit standard T1 and analog cables included Immediate, Wink, Loop, Ground, RevPol (Loop), and three DID start protocols Distinctive ring in Loop start mode Caller ID enabled in Loop start mode Low cost fully populated FXS, fully populated FXO, or modular FXS/FXO mixed mode capable 110VAC, 220VAC and -48VDC modular power supply models 5-year limited warranty Benefits Rhino Equipment Corp. channel banks provide a flexible and reliable product line that can satisfy any T1 need. Our products will beat your expectations, or your money back - guaranteed. T1 CSU All Rhino channel banks feature a single CSU that is software controlled,software programmable and will self configure to the proper mode right out of the box. Analog Interfaces The Rhino channel bank provides either FXS or FXO analog interfaces, both with digital technology that utilizes a dedicated Digital Signal Processor(DSP) for monitoring real-time line characteristics. FXS - The Rhino FXS analog inter-face provides standard battery and ringing voltages to each of up to 24 uoloop or ground start analog channels. Each channel provides: Digital DSP Echo cancellation Line current limit from 18ma to 45ma Tx and Rx gain control of -5db to+6db Power down control from 1 to 3 secs. Ring cadence control FXO - The Rhino FXO analog inter-face requires a standard POTs analog telephone line up to 24 inputs Each channel provides: Digital DSP Dynamic impedance matching Tx and Rx gain control of -15db to+12db Ring cadence monitoring timer FXS / FXO Modules - The modular Rhino channel bank can be ordered with four to 24 channels of either FXS or FXO analog channels installed in increments of four channels up to sixmodules.When using the Rhino in fractional T1 applications, there is no need to install a 4-channel analog card in those slot positions. 162 Rhino 24 Port FXS T1 Channel Bank 741.6000 824.00 Rhino Rhino Channel Banks http://www.voipon.co.uk/rhino-24-port-fxs-t1-channel-bank-p-162.html http://www.voipon.co.uk/images/rhino_24_fxs.jpg new Availability: In Stock Managing your telecommunication needs has never been easier than with Rhino products. Rhino satisfies the needs of any T1 channel bank application, no matter how stringent the requirement. Unique Rhino features like real-time T1 status on our four line by 40 character (4x40)LCD display, or automatic, hands-off configuration utilizing artificial intelligence software, and crystal clear audio quality proves that Rhino products are in a top class of their own. Knowing that the Rhino is ready to perform means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost T1 channel banks including FXS,FXO, or mixed mode FXS/FXO analog interfaces. As a bonus, every system comes with our standard fractional V.35 data port. Add the Rhino modular, internal power supply system to the list and Rhino crushes the competition. Using Asterisk? Rhino channel banks allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your T1 applications. Features Asterisk* soft PBX tested and ready! 4x40 LCD real-time status display shows independent AB bit signaling Automatic T1 configuration using arti-ficial intelligence software, no config-uration switches CSU auto detects T1modes: D4 or ESF, AMI or B8ZS T1 CSU via RJ48C, with line build out programmability Internal power supply module, no external components FLASH based system configuration V.35 fractional data interface, user selectable to 56K or 64K Self diagnostics, verbose error reporting, loop back modes Single channel configuration, each channel may have a different analog start mode, signaling protocol, line current, or gain Telco network uptime and downtime for T1 history Remote connection via RS-232c or Ethernet (optional) via Windows based graphical user interface Field software upgradable 19", 23" or wall mounting kit standard T1 and analog cables included Immediate, Wink, Loop, Ground, RevPol (Loop), and three DID start protocols Distinctive ring in Loop start mode Caller ID enabled in Loop start mode Low cost fully populated FXS, fully populated FXO, or modular FXS/FXO mixed mode capable 110VAC, 220VAC and -48VDC modular power supply models 5-year limited warranty Benefits Rhino Equipment Corp. channel banks provide a flexible and reliable product line that can satisfy any T1 need. Our products will beat your expectations, or your money back - guaranteed. T1 CSU All Rhino channel banks feature a single CSU that is software controlled,software programmable and will self configure to the proper mode right out of the box. Analog Interfaces The Rhino channel bank provides either FXS or FXO analog interfaces, both with digital technology that utilizes a dedicated Digital Signal Processor(DSP) for monitoring real-time line characteristics. FXS - The Rhino FXS analog inter-face provides standard battery and ringing voltages to each of up to 24 uoloop or ground start analog channels. Each channel provides: Digital DSP Echo cancellation Line current limit from 18ma to 45ma Tx and Rx gain control of -5db to+6db Power down control from 1 to 3 secs. Ring cadence control FXO - The Rhino FXO analog inter-face requires a standard POTs analog telephone line up to 24 inputs Each channel provides: Digital DSP Dynamic impedance matching Tx and Rx gain control of -15db to+12db Ring cadence monitoring timer FXS / FXO Modules - The modular Rhino channel bank can be ordered with four to 24 channels of either FXS or FXO analog channels installed in increments of four channels up to sixmodules.When using the Rhino in fractional T1 applications, there is no need to install a 4-channel analog card in those slot positions. 163 Rhino Modular Channel Bank FXO Module 198.95 198.95 Rhino Rhino Channel Banks http://www.voipon.co.uk/rhino-modular-channel-bank-fxo-module-p-163.html http://www.voipon.co.uk/images/rhino_fxo_module.jpg new Availability: In Stock FXO Module for the Modular Chassis which takes up to 6 4-port FXO or FXS modules in any combination. Providing reliable, flexible, and leading-edge solutions for a demanding industry Managing your telecommunication needs has never been easier than with Rhino products. Rhino satisfies the needs of any T1channel bank application, no matter how stringent the requirement. Unique Rhino features like real-time T1 status on our four line by 40 character (4x40) LCD display, or automatic, hands-off configuration utilizing artificial intelligence software, and the crystal clear audio quality proves that Rhino products are in a top class of their own. Knowing that the Rhino is ready to out perform means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost T1channel banks including FXS, FXO, or mixed mode FXS/FXO analogue interfaces. As a bonus, every system comes with our standard fractional V.35 data port. Add the Rhino modular, internal power supply system to the list and Rhino crushes the competition. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your T1 application. 164 Rhino Modular Channel Bank FXS Module 98.95 98.95 Rhino Rhino Channel Banks http://www.voipon.co.uk/rhino-modular-channel-bank-fxs-module-p-164.html http://www.voipon.co.uk/images/rhino_fxs_module.jpg new Availability: In Stock FXS Module for the Modular Chassis which takes up to 6 4-port FXO or FXS modules in any combination. Providing reliable, flexible, and leading-edge solutions for a demanding industry Managing your telecommunication needs has never been easier than with Rhino products. Rhino satisfies the needs of any T1channel bank application, no matter how stringent the requirement. Unique Rhino features like real-time T1 status on our four line by 40 character (4x40) LCD display, or automatic, hands-off configuration utilizing artificial intelligence software, and the crystal clear audio quality proves that Rhino products are in a top class of their own. Knowing that the Rhino is ready to out perform means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost T1channel banks including FXS, FXO, or mixed mode FXS/FXO analogue interfaces. As a bonus, every system comes with our standard fractional V.35 data port. Add the Rhino modular, internal power supply system to the list and Rhino crushes the competition. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your T1 application. 166 Epygi Quadro 4 Expansion Key 400 400.00 Epygi Quadro IP PBX Systems http://www.voipon.co.uk/epygi-quadro-4-expansion-key-p-166.html http://www.voipon.co.uk/images/key.jpg new Availability: In Stock Expansion key for 12 additional local IP extensions 167 Snom Asterisk Kit 238.95 238.95 Asterisk PBX Kits http://www.voipon.co.uk/snom-asterisk-kit-p-167.html http://www.voipon.co.uk/images/snom_asterisk_kit.gif new Availability: In Stock These great value Asterisk kits allow you to setup a small IP PBX at a minimal price and come with: 1 x TDM01B - The TDM01B 1 incoming line (fxo) allows for 1 incoming line from your telco provider (BT/NTL) 2 x Snom 300 - The snom 300 addresses demanding businness customers using the latest VoIP technology. The 3 items above will allow for a small 2 extension IP PBX. If however you need more incoming lines or wish to upgrade the type of Snom telephone please choose from the options below. Snom 300 As the basic model of the SNOM business telephone family, the SNOM 300 fulfils the most important requirements of VoIP telephony and additionally offers numerous functions that are indispensable in the business world. SNOM 320 Ideal for general office and knowledge-worker environments, the SNOM 320 is an affordable, yet powerful SIP business telephone with built-in, full-duplex speakerphone and three-party conference bridging. SNOM 360 The SNOM 360 is designed for maximum productivity and efficiency in the everyday business environment. Dedicated keys provide direct access to audio and call control functions. The graphical display can be tilted for optimum reading angle. 168 Grandstream Asterisk Kit 173.95 173.95 Asterisk PBX Kits http://www.voipon.co.uk/grandstream-asterisk-kit-p-168.html http://www.voipon.co.uk/images/grandstream_asterisk_kit.gif new Availability: In Stock These great value Asterisk kits allow you to setup a small IP PBX at a minimal price and come with: 1 x TDM01B - The TDM01B 1 incoming line (fxo) allows for 1 incoming line from your telco provider (BT/NTL) 2 x Grandstream BT200 - SIP based IP Telephone The 3 items above will allow for a small 2 extension IP PBX. If however you need more incoming lines or wish to upgrade the type of Grandstream telephone please choose from the options below. Grandstream BT101 Basic SIP Phone. Grandstream BT200 The Grandstream BudgeTone-200 series IP phones are award-winning, next generation IP telephones based on SIP industry open standards. Grandstream GXP2000 The GXP-2000 is a next-generation enterprise IP telephone based on open industry standards. 169 Aastra Asterisk Kit 209.95 209.95 Asterisk PBX Kits http://www.voipon.co.uk/aastra-asterisk-kit-p-169.html http://www.voipon.co.uk/images/aastra_asterisk_kit.gif new Availability: In Stock These great value Asterisk kits allow you to setup a small IP PBX at a minimal price and come with: 1 x TDM01B - The TDM01B 1 incoming line (fxo) allows for 1 incoming line from your telco provider (BT/NTL) 2 x Aastra 9112i - Value priced, basic single line IP Telephone The 3 items above will allow for a small 2 extension IP PBX. If however you need more incoming lines or wish to upgrade the type of Aastra telephone please choose from the options below. Aastra 9112i The Model 9112i is a value priced, basic single line IP Telephone with speakerphone offering a flexible, interoperable IP solution that meets VoIP budgets for most applications. Aastra 9133i The Model 9133i is an advanced, fully featured multi-line IP Telephone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable price. Aastra 480i The Model 480i is an advanced, fully featured, IP screen Telephone that provides a flexible IP solution designed with interoperability and ease of use in mind. Please note: The Aastra 480i and the Aastra 9133i no longer require a PoE converter as they are both shipped with a power supply. 170 Polycom Asterisk Kit 275 275.00 Asterisk PBX Kits http://www.voipon.co.uk/polycom-asterisk-kit-p-170.html http://www.voipon.co.uk/images/polycom_asterisk_kit.gif new Availability: In Stock These great value Asterisk kits allow you to setup a small IP PBX at a minimal price and come with: 1 x TDM01B - The TDM01B 1 incoming line (fxo) allows for 1 incoming line from your telco provider (BT/NTL) 2 x Polycom SoundPoint IP300 - basic single line IP Telephone with speakerphone offering a flexible, interoperable IP solution that meets VoIP budgets for most applications. The 3 items above will allow for a small 2 extension IP PBX. If however you need more incoming lines or wish to upgrade the type of Polycom telephone please choose from the options below. Polycom SoundPoint IP300 The Polycom SoundPoint IP300 is a two-line desktop IP telephone that delivers remarkable value. Polycom SoundPoint IP501 The power of a network. The simplicity of a phone. Polycom SoundPoint IP601 Six-line convenience with unsurpassed hands-free voice quality and a high-resolution display. 171 Netgear 8 Port Switch with 4 PoE Ports 78.95 78.95 Netgear Netgear PoE Switches http://www.voipon.co.uk/netgear-8-port-switch-with-4-poe-ports-p-171.html http://www.voipon.co.uk/images/netgear_4_poe.jpg new Availability: In Stock The ProSafe FS108P provides power and data from a single point, using Power over Ethernet (PoE) over a single Cat-5 cable. The eight Fast Ethernet ports can be used for any 10/100 Mbps link and four of these ports can supply industry-standard IEEE 802.3af power. Advanced auto-sensing algorithm gives power only to 802.3af end devices, so no need to worry about damaging proprietary PoE or non-PoE equipment. In addition, it discontinues the power when PoE devices are disconnected. Easy and reliable, the ProSafe FS108P automatically determines PoE requirements, speed, duplex, and cable type using AutoUplink®. The affordably priced ProSafe FS108P delivers PoE to any small business network that wants to simplify the installation of wireless access points and IP-based surveillance cameras. These devices are optimally installed on a ceiling or high on a wall, away from most electrical outlets. PoE eliminates the requirement for a dedicated electrical outlet to power these devices. This allows for flexibility in situating devices where AC power is difficult to access and lowers installation costs. Compact and flexible, the ProSafe FS108P is ideal for small business network that want to inexpensively use PoE to deploy wireless access points and IP-based network surveillance cameras. Flexible Choose to plug in up to eight Ethernet or Fast Ethernet devices and mix in up to four 802.3af IP-based devices like wireless access points or IP-based network surveillance cameras. Place these 802.3af-compliant devices where they belong - high up on walls and ceiling for maximum coverage - or anywhere else you need them. Power and data are carried over standard Cat-5 cabling. Plug and Play The standards-based ProSafe FS108P senses and adjusts for network speed and cabling type automatically, for easy integration into your existing 10/100 Ethernet network. For PoE, the switch automatically detects 802.3af-compliant devices, and supplies power as needed. Front panel LEDs keep you informed of switch and PoE status. Quiet and Compact Engineered for compact convenience, it features a 9-inch, durable metal case that is easily positioned on your desktop or a wall, using the included mounting hardware. The fan-less design quietly integrates in your small office environment. Great Value With data switching and Power over Ethernet integrated into one unit, the FS108P saves space, reduces cables and eliminates the requirement for dedicated electrical outlets - lowering installation costs, simplifying installation of PoE-capable devices, and eliminating the need for electricians or extension cords. All in all a great benefit for a modest price. 172 Netgear 16 Port Switch with 8 port PoE 124.0000 142.95 Netgear Netgear PoE Switches http://www.voipon.co.uk/netgear-16-port-switch-with-8-port-poe-p-172.html http://www.voipon.co.uk/images/netgear_8_poe.jpg new Availability: In Stock The ProSafe FS116P provides power and data from a single point, using Power over Ethernet (PoE) over a single Cat-5 cable. The sixteen Fast Ethernet ports can be used for any 10/100 Mbps link and eight of these ports can supply industry-standard IEEE 802.3af power. Advanced auto-sensing algorithm gives power only to 802.3af end devices, so no need to worry about damaging proprietary PoE or non-PoE equipment. In addition, it discontinues the power when PoE devices are disconnected. Easy and reliable, the ProSafe FS116P automatically determines PoE requirements, speed, duplex, and cable type using AutoUplink®. Affordable Power over Ethernet Switching The affordably priced ProSafe FS116P delivers PoE to any small business network that wants to simplify the installation of wireless access points and IP-based surveillance cameras. These devices are optimally installed on a ceiling or high on a wall, away from most electrical outlets. PoE eliminates the requirement for a dedicated electrical outlet to power these devices. This allows for flexibility in situating devices where AC power is difficult to access and lowers installation costs. Compact and flexible, the ProSafe FS116P is ideal for small business network that want to inexpensively use PoE to deploy wireless access points and IP-based network surveillance cameras. Flexible Choose to plug in up to eight Ethernet or Fast Ethernet devices and mix in up to four 802.3af IP-based devices like wireless access points or IP-based network surveillance cameras. Place these 802.3af-compliant devices where they belong high up on walls and ceiling for maximum coverage or anywhere else you need them. Power and data are carried over standard Cat-5 cabling. Plug and Play The standards-based ProSafe FS116P senses and adjusts for network speed and cabling type automatically, for easy integration into your existing 10/100 Ethernet network. For PoE, the switch automatically detects 802.3af-compliant devices, and supplies power as needed. Front panel LEDs keep you informed of switch and PoE status. Quiet and Compact Engineered for compact convenience, it is only 1-inch high and 4-inches deep, with a durable metal case that is easily positioned on your desktop or a wall, using the included mounting hardware. The fan-less design quietly integrates in your small office environment. Great Value With data switching and Power over Ethernet integrated into one unit, the FS116P saves space, reduces cables and eliminates the requirement for dedicated electrical outlets lowering installation costs, simplifying installation of PoE-capable devices, and eliminating the need for electricians or extension cords. All in all a great benefit for a modest price. 174 Breakout Box - TDM2400P 78 78.00 VoIP Accessories http://www.voipon.co.uk/breakout-box-tdm2400p-p-174.html http://www.voipon.co.uk/images/break_out_box.jpg new 25 RJ-11 Jacks + Connection to both Male and Female Connectors This Breakout Box interfaces perfectly with the Digium TDM2400P, which (unlike previous Digium boards which had RJ-11 or RJ-45 jacks on the back of the card), features a 25 pair (50 pin) \"Amphenol\" connector and requires a Breakout Box and 25-pair Connector Cable. The Breakout Box includes cable gripper, double stick pads, and screws. It is fully compatible with male and female connectors. 175 Digium TDM2400P FXO Module 105.5000 111.00 Digium TDM2400P FXS/FXO Module http://www.voipon.co.uk/digium-tdm2400p-fxo-module-p-175.html http://www.voipon.co.uk/images/tdm2400p_fxo.jpg new The FXO module allows the TDM400P card to terminate Analogue phone lines (POTS). Due to its modular design, a user can activate additional ports at any time with more FXS or FXO daughter cards. The FXO module passes all the call features any standard Analogue phone line will support. Worldwide certifications are pending. 176 Digium TDM2400P FXS Module 114.9500 121.00 Digium TDM2400P FXS/FXO Module http://www.voipon.co.uk/digium-tdm2400p-fxs-module-p-176.html http://www.voipon.co.uk/images/tdm2400p_fxs.jpg new The FXS module allows the TDM2400P card to terminate Analogue phones. Because of the modular design, a user can activate additional ports at any time with more FXS or FXO daughter cards. The FXS module passes all the call features any standard Analogue phone will support. Worldwide certifications are pending. 177 Digium TDM2401E 0 FXS / 4 FXO with Echo Cancellation 372.0000 504.50 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2401e-0-fxs-4-fxo-with-echo-cancellation-p-177.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2401E 0 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 1 (1 Module, 4FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 178 Digium TDM2402E 0 FXS / 8 FXO with Echo Cancellation 481.7000 655.80 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2402e-0-fxs-8-fxo-with-echo-cancellation-p-178.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2402E 0 FXS / 8 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 179 Digium TDM2403E 0 FXS / 12 FXO with Echo Cancellation 591.6000 807.20 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2403e-0-fxs-12-fxo-with-echo-cancellation-p-179.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2403E 0 FXS / 12 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 180 Digium TDM2404E 0 FXS / 16 FXO with Echo Cancellation 701.4000 958.50 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2404e-0-fxs-16-fxo-with-echo-cancellation-p-180.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2404E 0 FXS / 16 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 181 Digium TDM2405E 0 FXS / 20 FXO with Echo Cancellation 811.3000 1109.90 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2405e-0-fxs-20-fxo-with-echo-cancellation-p-181.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2405E 0 FXS / 20 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 5 (5 Modules, 20 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 182 Digium TDM2406E 0 FXS / 24 FXO with Echo Cancellation 921.0000 1261.20 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2406e-0-fxs-24-fxo-with-echo-cancellation-p-182.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2406E 0 FXS / 24 FXO with Echo Cancellation # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 6 (6 Modules, 24 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 183 Digium TDM2410E 4 FXS / 0 FXO with Echo Cancellation 356.6000 482.80 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2410e-4-fxs-0-fxo-with-echo-cancellation-p-183.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2410E 4 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 184 Digium TDM2411E 4 FXS / 4 FXO with Echo Cancellation 466.5000 634.20 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2411e-4-fxs-4-fxo-with-echo-cancellation-p-184.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2411E 4 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 185 Digium TDM2412E 4 FXS / 8 FXO with Echo Cancellation 668.2000 854.90 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2412e-4-fxs-8-fxo-with-echo-cancellation-p-185.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2412E 4 FXS / 8 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 186 Digium TDM2414E 4 FXS / 16 FXO with Echo Cancellation 795.9000 1088.20 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2414e-4-fxs-16-fxo-with-echo-cancellation-p-186.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2414E 4 FXS / 16 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 187 Digium TDM2413E 4 FXS / 12 FXO with Echo Cancellation 686.1000 936.90 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2413e-4-fxs-12-fxo-with-echo-cancellation-p-187.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2413E 4 FXS / 12 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 188 Digium TDM2415E 4 FXS / 20 FXO with Echo Cancellation 905.8000 1239.60 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2415e-4-fxs-20-fxo-with-echo-cancellation-p-188.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2415E 4 FXS / 20 FXO with Echo Cancellation # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 5 (5 Modules, 20 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 189 Digium TDM2420E 8 FXS / 0 FXO with Echo Cancellation 451.1000 612.60 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2420e-8-fxs-0-fxo-with-echo-cancellation-p-189.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2420E 8 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 190 Digium TDM2421E 8 FXS / 4 FXO with Echo Cancellation 561.0000 763.90 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2421e-8-fxs-4-fxo-with-echo-cancellation-p-190.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2421E 8 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 191 Digium TDM2422E 8 FXS / 8 FXO with Echo Cancellation 670.7000 915.30 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2422e-8-fxs-8-fxo-with-echo-cancellation-p-191.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2422E 8 FXS / 8 FXO with Echo Cancellation # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 192 Digium TDM2423E 8 FXS / 12 FXO with Echo Cancellation 780.6000 1066.60 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2423e-8-fxs-12-fxo-with-echo-cancellation-p-192.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2423E 8 FXS / 12 FXO with Echo Cancellation # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 193 Digium TDM2424E 8 FXS / 16 FXO with Echo Cancellation 890.4000 1218.00 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2424e-8-fxs-16-fxo-with-echo-cancellation-p-193.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2424E 8 FXS / 16 FXO with Echo Cancellation # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 194 Digium TDM2430E 12 FXS / 0 FXO with Echo Cancellation 545.6000 742.30 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2430e-12-fxs-0-fxo-with-echo-cancellation-p-194.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2430E 12 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 195 Digium TDM2431E 12 FXS / 4 FXO with Echo Cancellation 654.6000 893.60 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2431e-12-fxs-4-fxo-with-echo-cancellation-p-195.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2431E 12 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 196 Digium TDM2432E 12 FXS / 8 FXO with Echo Cancellation 765.2000 1045.00 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2432e-12-fxs-8-fxo-with-echo-cancellation-p-196.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2432E 12 FXS / 8 FXO with Echo Cancellation # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 197 Digium TDM2433E 12 FXS / 12 FXO with Echo Cancellation 875.1000 1197.30 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2433e-12-fxs-12-fxo-with-echo-cancellation-p-197.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2433E 12 FXS / 12 FXO with Echo Cancellation # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 198 Digium TDM2440E 16 FXS / 0 FXO with Echo Cancellation 640.1000 872.00 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2440e-16-fxs-0-fxo-with-echo-cancellation-p-198.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2440E 16 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 199 Digium TDM2441E 16 FXS / 4 FXO with Echo Cancellation 750.0000 1023.40 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2441e-16-fxs-4-fxo-with-echo-cancellation-p-199.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2441E 16 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 200 Digium TDM2442E 16 FXS / 8 FXO with Echo Cancellation 859.7000 1174.70 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2442e-16-fxs-8-fxo-with-echo-cancellation-p-200.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2442E 16 FXS / 8 FXO with Echo Cancellation # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 201 Digium TDM2450E 20 FXS / 0 FXO with Echo Cancellation 734.6000 1002.80 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2450e-20-fxs-0-fxo-with-echo-cancellation-p-201.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2450E 20 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 5 (5 Modules, 20 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 202 Digium TDM2451E 20 FXS / 4 FXO with Echo Cancellation 844.5000 1153.10 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2451e-20-fxs-4-fxo-with-echo-cancellation-p-202.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2451E 20 FXS / 4 FXO with Echo Cancellation # of Quad FXS Resources: 5 (5 Modules, 20 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 203 Digium TDM2460E 24 FXS / 0 FXO with Echo Cancellation 829.1000 1131.50 Digium TDM2400P w/Echo Cancel http://www.voipon.co.uk/digium-tdm2460e-24-fxs-0-fxo-with-echo-cancellation-p-203.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2460E 24 FXS / 0 FXO with Echo Cancellation # of Quad FXS Resources: 6 (6 Modules, 24 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) This Module HAS Echo Cancellation Onboard. The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 204 Digium TDM2401B 0 FXS / 4 FXO 259.5000 350.40 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2401b-0-fxs-4-fxo-p-204.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2401B 0 FXS / 4 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 205 Digium TDM2402B 0 FXS / 8 FXO 369.4000 501.80 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2402b-0-fxs-8-fxo-p-205.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2402B 0 FXS / 8 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 206 Digium TDM2403B 0 FXS / 12 FXO 479.2000 653.10 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2403b-0-fxs-12-fxo-p-206.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2403B 0 FXS / 12 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 207 Digium TDM2404B 0 FXS / 16 FXO 589.1000 807.50 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2404b-0-fxs-16-fxo-p-207.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2404B 0 FXS / 16 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 208 Digium TDM2405B 0 FXS / 20 FXO 698.8000 955.80 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2405b-0-fxs-20-fxo-p-208.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2405B 0 FXS / 20 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 5 (5 Modules, 20 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 209 Digium TDM2406B 0 FXS / 24 FXO 808.7000 1108.20 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2406b-0-fxs-24-fxo-p-209.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2406B 0 FXS / 24 FXO # of Quad FXS Resources: 0 (0 Modules, 0 FXS Ports Total) # of Quad FXO Resources: 6 (6 Modules, 24 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 210 Digium TDM2410B 4 FXS / 0 FXO 244.3000 328.80 Digium TDM2400P http://www.voipon.co.uk/digium-tdm2410b-4-fxs-0-fxo-p-210.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2410B 4 FXS / 0 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 211 Digium TDM2411B 4 FXS / 4 FXO 354.0000 480.10 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2411b-4-fxs-4-fxo-p-211.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2411B 4 FXS / 4 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 212 Digium TDM2412B 4 FXS / 8 FXO 463.9000 631.50 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2412b-4-fxs-8-fxo-p-212.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2412B 4 FXS / 8 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 213 Digium TDM2413B 4 FXS / 12 FXO 573.7000 782.80 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2413b-4-fxs-12-fxo-p-213.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2413B 4 FXS / 12 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 214 Digium TDM2414B 4 FXS / 16 FXO 683.6000 934.20 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2414b-4-fxs-16-fxo-p-214.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium\'s Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2414B 4 FXS / 16 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 215 Digium TDM2415B 4 FXS / 20 FXO 793.3000 1085.50 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2415b-4-fxs-20-fxo-p-215.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2415B 4 FXS / 20 FXO # of Quad FXS Resources: 1 (1 Modules, 4 FXS Ports Total) # of Quad FXO Resources: 5 (5 Modules, 20 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 216 Digium TDM2420B 8 FXS / 0 FXO 338.8000 458.50 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2420b-8-fxs-0-fxo-p-216.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2420B 8 FXS / 0 FXO # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 217 Digium TDM2421B 8 FXS / 4 FXO 448.5000 609.90 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2421b-8-fxs-4-fxo-p-217.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2421B 8 FXS / 4 FXO # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 218 Digium TDM2422B 8 FXS / 8 FXO 780.6000 1066.50 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2422b-8-fxs-8-fxo-p-218.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2422B 8 FXS / 8 FXO # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 219 Digium TDM2423B 8 FXS / 12 FXO 668.2000 912.60 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2423b-8-fxs-12-fxo-p-219.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2423B 8 FXS / 12 FXO # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 220 Digium TDM2424B 8 FXS / 16 FXO 778.1000 1063.90 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2424b-8-fxs-16-fxo-p-220.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2424B 8 FXS / 16 FXO # of Quad FXS Resources: 2 (2 Modules, 8 FXS Ports Total) # of Quad FXO Resources: 4 (4 Modules, 16 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 221 Digium TDM2430B 12 FXS / 0 FXO 433.5000 588.20 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2430b-12-fxs-0-fxo-p-221.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2430B 12 FXS / 0 FXO # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 222 Digium TDM2431B 12 FXS / 4 FXO 543.0000 739.60 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2431b-12-fxs-4-fxo-p-222.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2431B 12 FXS / 4 FXO # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 223 Digium TDM2432B 12 FXS / 8 FXO 652.9000 890.90 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2432b-12-fxs-8-fxo-p-223.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2432B 12 FXS / 8 FXO # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 224 Digium TDM2433B 12 FXS / 12 FXO 762.7000 1042.30 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2433b-12-fxs-12-fxo-p-224.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2433B 12 FXS / 12 FXO # of Quad FXS Resources: 3 (3 Modules, 12 FXS Ports Total) # of Quad FXO Resources: 3 (3 Modules, 12 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 225 Digium TDM2440B 16 FXS / 0 FXO 527.8000 718.00 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2440b-16-fxs-0-fxo-p-225.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2440B 16 FXS / 0 FXO # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 226 Digium TDM2441B 16 FXS / 4 FXO 637.5000 869.30 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2441b-16-fxs-4-fxo-p-226.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2441B 16 FXS / 4 FXO # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 227 Digium TDM2442B 16 FXS / 8 FXO 747.4000 1020.70 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2442b-16-fxs-8-fxo-p-227.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2442B 16 FXS / 8 FXO # of Quad FXS Resources: 4 (4 Modules, 16 FXS Ports Total) # of Quad FXO Resources: 2 (2 Modules, 8 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 228 Digium TDM2450B 20 FXS / 0 FXO 622.3000 847.70 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2450b-20-fxs-0-fxo-p-228.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2450B 20 FXS / 0 FXO # of Quad FXS Resources: 5 (5 Modules, 20 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 229 Digium TDM2451B 20 FXS / 4 FXO 732.0000 999.00 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2451b-20-fxs-4-fxo-p-229.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2451B 20 FXS / 4 FXO # of Quad FXS Resources: 5 (5 Modules, 20 FXS Ports Total) # of Quad FXO Resources: 1 (1 Modules, 4 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 230 Digium TDM2460B 24 FXS / 0 FXO 716.8000 977.40 Digium Digium TDM2400P http://www.voipon.co.uk/digium-tdm2460b-24-fxs-0-fxo-p-230.html http://www.voipon.co.uk/images/digium_tdm2400p_small.jpg new Availability: In Stock The Wildcard TDM2400P is a full-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. Digium TDM2460B 24 FXS / 0 FXO # of Quad FXS Resources: 6 (6 Modules, 24 FXS Ports Total) # of Quad FXO Resources: 0 (0 Modules, 0 FXO Ports Total) The TDM2400P takes the place of an expensive channel bank and brings the system price point to a low level. By using X400M and S400M modules with the TDM2400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM2400P cards populated with modules. 231 ZyXEL Prestige 2000W VoIP Wireless Phone 134.4200 141.55 Zyxel Zyxel IP Telephone http://www.voipon.co.uk/zyxel-prestige-2000w-voip-wireless-phone-p-231.html http://www.voipon.co.uk/images/zyxel_2000w_l.jpg new The mobile Prestige 2000W VoIP Wi-Fi phone allows users to make and receive low cost phone calls though the internet via a SIP VoIP service provider. Compatible with both 802.11b/g wireless networks and easy to install and use, the Prestige 2000W may be used as a cordless handset for business or residential users and allows users to place VoIP phone calls in public 802.11-based hot spots. The Prestige 2000W supports industry standard SIP (Session Initiation Protocol) (RFC 3261) and is compatible with all SIP-based call services, IP-PBXs and other SIP based VoIP devices. The Prestige 2000W can be configured with the easy to navigate screen or via a web browser. Make/receive low cost phone calls though the internet via a SIP VoIP gateway service. Easy to use and install, configure via the LCD screen or with a Web browser. Compatible with any Wireless 802.11b or g Internet connected router. Supports 802.11b and g wireless networks, 64/128bit WEP encryption. Industry standard SIP compatible with most major VoIP service providers. High quality voice with G.711 and G.729 voice compression technology. Intercom facility with direct IP-to-IP calls with phone book. Supports VLAN, QoS such as ToS and Diffserv, and has an auto-provisioning mechanism. DHCP client. Battery life 24 hours standby, 4 hours talk-time. 2 Year warranty. 232 1m 50 pin Centronics Male to Male 4.95 4.95 VoIP Accessories http://www.voipon.co.uk/1m-50-pin-centronics-male-to-male-p-232.html http://www.voipon.co.uk/images/1m_centronics.jpg new 1 metre 50 pin Centronics male - 50 pin Centronics male 233 Zyxel Prestige 2002 ATA 40.4600 44.95 Zyxel Zyxel Analog Adaptors http://www.voipon.co.uk/zyxel-prestige-2002-ata-p-233.html http://www.voipon.co.uk/images/zyxel_prestige_2002.jpg new Availability: Please note this product has now been discontinued. The ZyXEL Prestige 2002 2 Port VoIP ATA (Analogue Telephone Adaptor) is an easy to use and install, low cost way for users to make/receive low cost phone calls though the internet using normal BT phones via a SIP VoIP service provider with out using a PC. Designed for home and telecommuters who wish to make/receive low cost phone calls though the internet using normal BT phones via a SIP VoIP service provider. Main Features Make/receive low cost VoIP phone calls though the internet. Industry compatible SIP compatible. Compatible with all other Internet Connected Routers. Features Easy to use and easy to install using a web browser for the 2002 ATA. Industry compatible SIP (Session Initiation Protocol, RFC 3261) for maximum compatibility with service providers and other VoIP devices. 2 x VoIP Telephone ports and 2 x 10/100 Ethernet ports. Great sound quality. Compatible with all other Internet Connected Routers for the 2002 ATA. Can be centrally managed using Telnet management, and TFTP Auto-Provisioning. Configure more than one SIP phone number, each number number can assigned to one or both phone ports.If one phone is used, a user can still make/receive another call. 2 Year warranty for the 2002 ATA. 234 ZyXEL Prestige 2002L ATA Lifeline 47.9500 49.95 Zyxel Zyxel Analog Adaptors http://www.voipon.co.uk/zyxel-prestige-2002l-ata-lifeline-p-234.html http://www.voipon.co.uk/images/zyxel_prestige_2002L.jpg new Availability: Please note this product has now been discontinued. The ZyXEL Prestige 2002L 2 Port VoIP ATA (Analogue Telephone Adaptor) is an easy to use and install, low cost way for users to make/receive low cost phone calls though the internet using normal BT phones via a SIP VoIP service provider with out using a PC. Designed for home and telecommuters who wish to make/receive low cost phone calls though the internet using normal BT phones via a SIP VoIP service provider. Main Features Make/receive low cost VoIP phone calls though the internet. Industry compatible SIP compatible. Compatible with all other Internet Connected Routers. Telephone lifeline if broadband line fails. Features Easy to use and easy to install using a web browser. Telephone lifeline if broadband line fails for the 2002L. Industry compatible SIP (Session Initiation Protocol, RFC 3261) for maximum compatibility with service providers and other VoIP devices. 2 x VoIP Telephone ports and 2 x 10/100 Ethernet ports. Great sound quality for the 2002L. Compatible with all other Internet Connected Routers. Can be centrally managed using Telnet management, and TFTP Auto-Provisioning. Configure more than one SIP phone number, each number number can assigned to one or both phone ports.If one phone is used, a user can still make/receive another call. 2 Year warranty for the 2002L. 235 Boscom Cellulink Smartcell 111L 160.0000 175.75 Boscom Boscom Cellular Gateways http://www.voipon.co.uk/boscom-cellulink-smartcell-111l-p-235.html http://www.voipon.co.uk/images/smartcell_111l.gif new Availability: In Stock The CelluLink product line creates a direct connection between the office PBX and the cellular network. CelluLink enables cost savings of up to 60%, as it eliminates the interconnection charges between cellular and landline (PTT) networks. Features Integrated dual band GSM Module (900/1800MHz, or 850/1900MHz) SMS, Data & Fax capabilities (Optional) Network lock (optional) Well designed and compact Toll restriction Echo cancellation Reverse Polarity Additional advantages: Superb audio quality, cutting edge technology Quick and easy installation # Quick Return On Investment (ROI) Interface to Cellular Least Cost Routing (LCR) 236 Boscom CelluLink Smartcell 112 342 342.00 Boscom Boscom Cellular Gateways http://www.voipon.co.uk/boscom-cellulink-smartcell-112-p-236.html http://www.voipon.co.uk/images/smartcell_112.gif new Availability: In Stock The CelluLink product line creates a direct connection between the office PBX and the cellular network. CelluLink enables cost savings of up to 60%, as it eliminates the interconnection charges between cellular and landline (PTT) networks. Features One voice channel ISDN BRI interface Integrated dual band GSM Module (900/1800MHz) 850/1900 MHz is optional Remote management via GSM network Remote supervision of RF levels Alarm SMS message Automatic Routing, Fixed and Self-Learning Transmit and Receive Gain Adjustment Network lock BRI can be programmed as NT or TE Compact & Well designed Heavy duty operation Toll restriction Additional advantages: Superb audio quality, cutting edge technology Quick and easy installation Quick Return On Investment (ROI) Interface to Cellular Least Cost Routing(LCR) 237 Boscom Cellulink Smartcell 212 469.0000 522.50 Boscom Boscom Cellular Gateways http://www.voipon.co.uk/boscom-cellulink-smartcell-212-p-237.html http://www.voipon.co.uk/images/smartcell_212.gif new Availability: In Stock The CelluLink product line creates a direct connection between the office PBX and the cellular network. CelluLink enables cost savings of up to 60%, as it eliminates the interconnection charges between cellular and landline (PTT) networks. Features Two voice channel ISDN BRI interface Two integrated dual band GSM Module (900/1800MHz) 850/1900 MHz is optional Remote management via GSM network Remote supervision of RF levels Alarm SMS message Automatic Routing, Fixed and Self Learning Tranmit and Receive Gain Adjustment Network lock BRI can be programmed as NT or TE Compact & Well designed Heavy duty operation Toll restriction Additional advantages: Superb audio quality, cutting edge technology Quick and easy installation Quick Return On Investment (ROI) Interface to Cellular Least Cost Routing (LCR) 238 Boscom Claro 2010 627 627.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-2010-p-238.html http://www.voipon.co.uk/images/claro_2010.gif new Availability: In Stock The Claro 2010 IP telephony gateways deliver the quickest, simplest installation available on the market. Proprietary quality-boosting algorithms ensure the highest voice quality, while continuous operations are guaranteed because there is no single point of failure. In short, these gateways deliver superb audio quality, are easy to install and maintain, and are highly affordable. Call Handling Features H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by -call basis Automatic call type detection (voice/fax) Rule-based numbering management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary automatic hop-off support NAT (Network Address Translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 239 Boscom Claro 2020 627 627.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-2020-p-239.html http://www.voipon.co.uk/images/claro_2020.gif new Availability: In Stock The Claro 2020 IP telephony gateways deliver the quickest, simplest installation available on the market. Proprietary quality-boosting algorithms ensure the highest voice quality, while continuous operations are guaranteed because there is no single point of failure. In short, these gateways deliver superb audio quality, are easy to install and maintain, and are highly affordable. Call Handling Features H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by -call basis Automatic call type detection (voice/fax) Rule-based numbering management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary automatic hop-off support NAT (Network Address Translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 240 Boscom Claro 2060 2517.5 2517.50 Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-2060-p-240.html http://www.voipon.co.uk/images/claro_2060.gif new Availability: In Stock The Claro 2060 IP telephony gateways deliver the quickest, simplest installation available on the market. Proprietary quality-boosting algorithms ensure the highest voice quality, while continuous operations are guaranteed because there is no single point of failure. In short, these gateways deliver superb audio quality, are easy to install and maintain, and are highly affordable. Call Handling Features H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by -call basis Automatic call type detection (voice/fax) Rule-based numbering management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary automatic hop-off support NAT (Network Address Translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 241 Boscom Claro 2050 2185 2185.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-2050-p-241.html http://www.voipon.co.uk/images/claro_2050.gif new Availability: In Stock The Claro 2050 IP telephony gateways deliver the quickest, simplest installation available on the market. Proprietary quality-boosting algorithms ensure the highest voice quality, while continuous operations are guaranteed because there is no single point of failure. In short, these gateways deliver superb audio quality, are easy to install and maintain, and are highly affordable. Call Handling Features H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by -call basis Automatic call type detection (voice/fax) Rule-based numbering management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary automatic hop-off support NAT (Network Address Translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 242 Boscom Claro 3031 380 380.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-3031-p-242.html http://www.voipon.co.uk/images/claro_3031.jpg new Availability: In Stock Designed for small offices, Claro 3031 is a compact combination call and data routing solution for remote and branch offices. It transforms any two-line analog office phone system into a feature-rich converged voice/fax/data communication network. Claro 3031 delivers superb audio quality, is easy to install and maintain, and is highly affordable. Claro 3031 supports LAN switching, WAN routing and VoIP in one small-footprint package. It guarantees carrier-grade voice quality connections while maximizing cost-savings, with automatic routing to PSTN when the IP network cannot deliver the pre-specified quality level. Features Fully compatible with all vendor legacy systems and infrastructure No modifications to existing equipment Simple, centralized, secure browser-based remote administration No need for gatekeeper, no single point of failure Fully transparent to user H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by-call basis Discretionary, automatic least cost routing options (carrier selection; hop-off) Hop-on with authentication and auto-dialing support Smart QoS-driven automatic multi-path call switching Pass-through to user-selected PSTN destinations Automatic call type detection voice/fax Public, private, and abbreviated dial plan support; Rule Based Numbering Management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary, automatic hop-off support NAT (network address translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 243 Boscom Claro 3030 712.5 712.50 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-3030-p-243.html http://www.voipon.co.uk/images/claro_3031.jpg new Availability: In Stock Designed for small offices, Claro 3030 is a compact combination call and data routing solution for remote and branch offices. It transforms any two-line analog office phone system into a feature-rich converged voice/fax/data communication network. Claro 3030 delivers superb audio quality, is easy to install and maintain, and is highly affordable. Claro 3030 supports LAN switching, WAN routing and VoIP in one small-footprint package. It guarantees carrier-grade voice quality connections while maximizing cost-savings, with automatic routing to PSTN when the IP network cannot deliver the pre-specified quality level. Features Fully compatible with all vendor legacy systems and infrastructure No modifications to existing equipment Simple, centralized, secure browser-based remote administration No need for gatekeeper, no single point of failure Fully transparent to user H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by-call basis Discretionary, automatic least cost routing options (carrier selection; hop-off) Hop-on with authentication and auto-dialing support Smart QoS-driven automatic multi-path call switching Pass-through to user-selected PSTN destinations Automatic call type detection voice/fax Public, private, and abbreviated dial plan support; Rule Based Numbering Management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary, automatic hop-off support NAT (network address translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 244 Boscom Claro 3040 1149.5 1149.50 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-3040-p-244.html http://www.voipon.co.uk/images/claro_3031.jpg new Availability: In Stock Designed for small offices, Claro 3040 is a compact combination call and data routing solution for remote and branch offices. It transforms any two-line analog office phone system into a feature-rich converged voice/fax/data communication network. Claro 3040 delivers superb audio quality, is easy to install and maintain, and is highly affordable. Claro 3040 supports LAN switching, WAN routing and VoIP in one small-footprint package. It guarantees carrier-grade voice quality connections while maximizing cost-savings, with automatic routing to PSTN when the IP network cannot deliver the pre-specified quality level. Features Fully compatible with all vendor legacy systems and infrastructure No modifications to existing equipment Simple, centralized, secure browser-based remote administration No need for gatekeeper, no single point of failure Fully transparent to user H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by-call basis Discretionary, automatic least cost routing options (carrier selection; hop-off) Hop-on with authentication and auto-dialing support Smart QoS-driven automatic multi-path call switching Pass-through to user-selected PSTN destinations Automatic call type detection voice/fax Public, private, and abbreviated dial plan support; Rule Based Numbering Management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary, automatic hop-off support NAT (network address translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 4 Channels = 2 x ISDN2 Lines 8 Channels = 4 x ISDN2 Lines 245 Boscom Claro 3050 2090 2090.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-3050-p-245.html http://www.voipon.co.uk/images/claro_3031.jpg new Availability: In Stock Designed for small offices, Claro 3050 is a compact combination call and data routing solution for remote and branch offices. It transforms any two-line analog office phone system into a feature-rich converged voice/fax/data communication network. Claro 3050 delivers superb audio quality, is easy to install and maintain, and is highly affordable. Claro 3050 supports LAN switching, WAN routing and VoIP in one small-footprint package. It guarantees carrier-grade voice quality connections while maximizing cost-savings, with automatic routing to PSTN when the IP network cannot deliver the pre-specified quality level. Features Fully compatible with all vendor legacy systems and infrastructure No modifications to existing equipment Simple, centralized, secure browser-based remote administration No need for gatekeeper, no single point of failure Fully transparent to user H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by-call basis Discretionary, automatic least cost routing options (carrier selection; hop-off) Hop-on with authentication and auto-dialing support Smart QoS-driven automatic multi-path call switching Pass-through to user-selected PSTN destinations Automatic call type detection voice/fax Public, private, and abbreviated dial plan support; Rule Based Numbering Management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary, automatic hop-off support NAT (network address translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 246 Boscom Claro To Go 52.25 52.25 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-to-go-p-246.html http://www.voipon.co.uk/images/boscom_claro_to_go.gif new Availability: In Stock Claro to Go is the ideal USB device for IP telephony with analog phones. Now the user away from the office can enjoy all the savings of IP in a completely transparent manner. Now you can benefit from superb audio quality, easy installation and maintenance, all at a very affordable price. Claro to Go plugs into the USB port of a PC or laptop on one side, and into any analog telephone on the other. Digital voice quality is crystal clear. Claro to Go works seamlessly with other Claro to Go units and Claro IP telephony gateways, and can take advantage of all their features. Works seamlessly with gatekeeper and SIP proxies for service providers. Claro to Go supports Flash to enable supplementary services such as Call Transfer, Conference Call and Voice Mail. Features Numbering compatibility with all Claro gateways Built-in self-test Robust call handling Voice priority modulation Communications 247 Boscom Claro 3060 2090 2090.00 Boscom Boscom VoIP Gateways http://www.voipon.co.uk/boscom-claro-3060-p-247.html http://www.voipon.co.uk/images/claro_3060.jpg new Availability: In Stock Designed for small offices, Claro 3060 is a compact combination call and data routing solution for remote and branch offices. It transforms any two-line analog office phone system into a feature-rich converged voice/fax/data communication network. Claro 3060 delivers superb audio quality, is easy to install and maintain, and is highly affordable. Claro 3060 supports LAN switching, WAN routing and VoIP in one small-footprint package. It guarantees carrier-grade voice quality connections while maximizing cost-savings, with automatic routing to PSTN when the IP network cannot deliver the pre-specified quality level. Features Fully compatible with all vendor legacy systems and infrastructure No modifications to existing equipment Simple, centralized, secure browser-based remote administration No need for gatekeeper, no single point of failure Fully transparent to user H.323 V4 support SIP V2 support Dynamic protocol selection (SIP, H.323) on a call-by-call basis Discretionary, automatic least cost routing options (carrier selection; hop-off) Hop-on with authentication and auto-dialing support Smart QoS-driven automatic multi-path call switching Pass-through to user-selected PSTN destinations Automatic call type detection voice/fax Public, private, and abbreviated dial plan support; Rule Based Numbering Management (RBNM) Automatic stripping and appending of digits to dialed numbers Discretionary, automatic hop-off support NAT (network address translation) traversal Voice Quality IP TOS Diff Serv Bad frame interpolation Adaptive, dynamic jitter buffer Line Echo Cancellation ITU G.165, ITU G.168 Comfort noise generation (CNG) Voice activity detection (VAD)/Silence Suppression 248 Draytek Vigor2800V 130 130.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2800v-p-248.html http://www.voipon.co.uk/images/draytek_2800.jpg new Availability: Discontinued The Draytek 2800 series has been discontinued, and replaced by the 2820 series. The replacement for the 2800V is the Draytek Vigor 2820Vn . Draytek Vigor 2800V VPN ADSL 2/2+ Router w/Voice Over IP The DrayTek Vigor2800V has all of the features of the Vigor2800 ADSL router with the addition of two phone sockets, to provide Voice-Over-IP (VoIP) facilities. VoIP enables you to use your existing broadband capacity to carry regular voice calls to suitably equipped remote sites, for example another Vigor2800V router, to any other compatible VoIP device or, via a PSTN gateway to or from any other regular phone line or mobile phone worldwide. Using the Voice-over-IP facility is simple. You connect any standard telephone (corded or cordless) to one of the phone ports on the Vigor2800V. Although you can dial IP addresses, you would normally dial someone's 'SIP Address' which they obtain from their SIP Registrar (such as DrayTEL). You hear ringing and bust signals just like on a normal phone line. The voice calls (speech) are digitised and compressed in real time to make maximum use of the bandwidth and DrayTek's high performance 'codecs' provide 'toll quality' voice calls (quality similar to a regular fixed telephone line). A preset QoS system automatically gives voice traffic priority over other traffic without your available bandwidth. The calls between the two sites above are free of charge, making use of your existing always-on ADSL connection, but cost isn't the only advantage; using VOIP means that you have additional call capacity in your home or office, without tying up your regular phone line. You can also make calls to regular phone lines, via a PSTN gateway. Vigor2800V Series Voice-over-IP Features : ADSL2+ Router & Firewall Compatible with ADSL, ADSL2, ADSL2+ (up to 24Mb/s) Easy to use user interface for setup and control Four-Port Ethernet autosensing 10/100BaseT Switch VPN Server for up to 16 simultaneous tunnels Stateful Packet Inspection for NAT and Fully routed connections Internet URL Content Filtering (Blacklist or Whitelist) Parental Control with Surfcontrol™ Block IM & P2P Applications QoS Assurance for mission-critical applications/services Time Scheduling for Internet Access Virtual LAN (VLAN) - Segment Ethernet ports into distinct common/isolated groups Bandwidth Throttling - Restrict speed on each Ethernet port USB Printer Port - share regular printer centrally SNMPv2, Syslog, uPNP & Radius Support Voice-over-IP Features: Twin VoIP Telephone ports Connect any standard telephone (cordless or corded) SIP Compliant Voice codecs supported: 8Kb/s-64Kb/s Multiple Simultaneous SIP server registrations     249 Epygi Quadro 2 Expansion Key 162 162.00 Epygi Quadro IP PBX Systems http://www.voipon.co.uk/epygi-quadro-2-expansion-key-p-249.html http://www.voipon.co.uk/images/key.jpg new Availability: In Stock Expansion key for 4 additional local IP extensions 250 Parlay VoXIP 104 ISDN BRI Gateway 548 548.00 Parlay Parlay Gateways http://www.voipon.co.uk/parlay-voxip-104-isdn-bri-gateway-p-250.html http://www.voipon.co.uk/images/voxip_104.jpg new Availability: In Stock Ethernet interface and 2 ISDN BRI interfaces and up to 4 voice channels. Parlay VoXip Standard is the cost-efficient choice for ITSP's. It reliably connects any kind of ISDN-equipment to the SIP service, allowing customers to keep single ISDN lines. The Parlay VoXip ensures DDI to PBX extensions by individual SIP accounts as well as in accountless mode. Its well-proven ISDN implementation guarantees satisfied customers. It enables migration from ISDN to low cost VoIP without replacing or upgrading the existing ISDN voice communication platform. Key Features Makes any ISDN PBX VoIP capable - no modification of the PBX Supports call routing to / from SIP based VoIP Service Providers VPN: Transparent tunnelling of ISDN and QSIG via IP between 8 Endpoints Advanced call routing schemes via ISDN, SIP, VPN Seamless re-routing of calls via ISDN if no VoIP connectivity Well-proven ISDN stack, handles many national ISDN dialects Full Feature List ISDN protocols Euro-ISDN protocol according to ETS 300 102. Point-Point and Point-Multipoint, including line hunting. En-bloc and Overlap sending / receiving. QSIG tunneling. Supports national dialects: VN2 - VN6 (France), 1TR6 (Germany), BAKOM (Switzerland), INS64 (Japan). Voice Codecs and features G.711 (64 kbit/s PCM), A-/µ-law G.726 (32 kbit/s ADPCM), A-/µ-law Silence Suppression Voice Activity Detection Comfort Noise Generation. Echo cancellation Number of simultaneous IP calls: 4 Calls (VoXip 104) 8 Calls (VoXip 108) 16 Calls (VoXip 116) Inbound and outbound DTMF support SIP Features Supports 5 Proxies simultaneously 100 accounts (DDI) at the Proxy SIP 2.0 (RFC 3261, RFC 2543) Outbound Proxy and STUN support Re-routing via ISDN in case of poor IP-line QoS, IP Type of Service (DSCP marking) ISDN tunneling over IP Proprietary ISDN and QSIG tunneling over IP between 8 Parlay VoXip endpoints. RTP Multiplexing to reduce IP overhead Voice Switching Features Local switching, allowing internal calls between ISDN interfaces of the Parlay VoXip. Local call progress tone generation. Local Advice of Charge (AOC) generation, also at SIP calls. Call Transfer, Call Forwarding. Calling Number modification (e.g. present corporate numbers). Called Number modification (e.g. add Carrier Selection Code). Voice Routing Table specifying handling of calls: Least Cost Routing (e.g. corporate and international calls via IP) Toll bypass (e.g. via IP to branch and drop out to ISDN there) Programming Via built-in web server, providing a Graphical User Interface. Via Telnet session, menu-driven user interface. Via RS232 using terminal program, menu-driven user interface. 251 Parlay VoXIP 108 ISDN BRI Gateway 948 948.00 Parlay Parlay Gateways http://www.voipon.co.uk/parlay-voxip-108-isdn-bri-gateway-p-251.html http://www.voipon.co.uk/images/voxip_108.jpg new Availability: In Stock Ethernet interface and 4 ISDN BRI interfaces and up to 8 voice channels Parlay VoXip Standard is the cost-efficient choice for ITSP's. It reliably connects any kind of ISDN-equipment to the SIP service, allowing customers to keep single ISDN lines. The Parlay VoXip ensures DDI to PBX extensions by individual SIP accounts as well as in accountless mode. Its well-proven ISDN implementation guarantees satisfied customers. It enables migration from ISDN to low cost VoIP without replacing or upgrading the existing ISDN voice communication platform. Key Features Makes any ISDN PBX VoIP capable - no modification of the PBX Supports call routing to / from SIP based VoIP Service Providers VPN: Transparent tunnelling of ISDN and QSIG via IP between 8 Endpoints Advanced call routing schemes via ISDN, SIP, VPN Seamless re-routing of calls via ISDN if no VoIP connectivity Well-proven ISDN stack, handles many national ISDN dialects Full Feature List ISDN protocols Euro-ISDN protocol according to ETS 300 102. Point-Point and Point-Multipoint, including line hunting. En-bloc and Overlap sending / receiving. QSIG tunneling. Supports national dialects: VN2 - VN6 (France), 1TR6 (Germany), BAKOM (Switzerland), INS64 (Japan). Voice Codecs and features G.711 (64 kbit/s PCM), A-/µ-law G.726 (32 kbit/s ADPCM), A-/µ-law Silence Suppression Voice Activity Detection Comfort Noise Generation. Echo cancellation Number of simultaneous IP calls: 4 Calls (VoXip 104) 8 Calls (VoXip 108) 16 Calls (VoXip 116) Inbound and outbound DTMF support SIP Features Supports 5 Proxies simultaneously 100 accounts (DDI) at the Proxy SIP 2.0 (RFC 3261, RFC 2543) Outbound Proxy and STUN support Re-routing via ISDN in case of poor IP-link QoS, IP Type of Service (DSCP marking) ISDN tunneling over IP Proprietary ISDN and QSIG tunneling over IP between 8 Parlay VoXip endpoints. RTP Multiplexing to reduce IP overhead Voice Switching Features Local switching, allowing internal calls between ISDN interfaces of the Parlay VoXip. Local call progress tone generation. Local Advice of Charge (AOC) generation, also at SIP calls. Call Transfer, Call Forwarding. Calling Number modification (e.g. present corporate numbers). Called Number modification (e.g. add Carrier Selection Code). Voice Routing Table specifying handling of calls: Least Cost Routing (e.g. corporate and international calls via IP) Toll bypass (e.g. via IP to branch and drop out to ISDN there) Programming Via built-in web server, providing a Graphical User Interface. Via Telnet session, menu-driven user interface. Via RS232 using terminal program, menu-driven user interface. 252 UTstarcom F1000 Earbud Headset 8.99 8.99 UTstarcom VoIP Accessories http://www.voipon.co.uk/utstarcom-f1000-earbud-headset-p-252.html http://www.voipon.co.uk/images/f1000_headset.jpg new Availability: In stock Earbud headset for hands-free use. Works with any phone with a 2.5 mm jack. Has clip for attaching to a shirt or seatbelt to keep it in place better. Microphone is on the wire that connects the phone to the earbud. Slips into ear for comfortable wearing. Compatible with the F1000 WiFi Phone. 253 Polycom SoundStation IP4000 458.0000 495.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundstation-ip4000-p-253.html http://www.voipon.co.uk/images/polycom_ip-4000_sml.jpg new Availability: In stock Polycom Soundstation IP4000 - The Clear SIP Conferencing Solution Remarkable Voice Quality For Your SIP Environment With improved audio quality over the original Polycom SoundStation IP 3000 conference phone, the SoundStation IP 4000 is the answer for organisations that are ready for the benefits and versatility of a SIP voice conferencing unit. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP unit provides remarkable room coverage. Users can speak naturally from up to 10-feet away from a microphone and still be heard clearly at the other end of the call. Need better coverage for a larger room? Optional extension microphones offer an increased pickup for larger rooms. Furthermore, with gated microphone technology, echo and background noise is almost entirely eliminated. The SoundStation IP 4000 SIP also delivers familiar features that are easy to use. The menu driven user interface, viewable on a high-res backlit LCD, offers convenient access to the most-used business telephony functions including transfer, hold, redial, and conference. For your next conference call, choose to enjoy great voice quality and the convenience of connecting directly into your SIP-based PBX with the Polycom SoundStation IP 4000 SIP. Coupled with Polycom's desktop SoundPoint IP SIP telephones, the SoundStation IP 4000 SIP is the natural addition to your IP conference room or office. With the greatest breadth and depth of integrated video, voice, and Web solutions, Polycom delivers the ultimate communications experience. Polycom\'s market-leading conferencing and collaboration technologies, supported by world-class service, enable people and organisations to maximize their effectiveness and productivity. With the most experience and proven best-practices in the industry, it's clear why Polycom is the smart choice for organisations seeking a strategic advantage in a real-time world.   Polycom Soundstation IP 4000 Features & Benefits More productive calls - Polycom's patented Acoustic Clarity Technology allows simultaneous, natural, freeflowing conversation. Hear and be heard clearly - Speak up to 10-feet from a microphone and be heard clearly on the far end of the call. Ability to increase coverag e - Optional extended microphones expand coverage to 20 x 30 feet for larger conference rooms. Reduce echo and background noise - Gated microphones nearly eliminate distracting echo and room noise. Backlit display provides ease of use - 248 X 68 pixel resolution LCD makes menu navigation easy; Support for foreign languages including Asian characters. Ideal for offices and small- to medium-sized conference rooms Polycom Acoustic Clarity Technology for outstanding sound quality and full-duplex speakerphone performance Business-class SIP features Easy-to-read backlit LCD with intuitive user interface Enjoy increased productivity and faster decision making through more natural Business-class SIP features: directories, 3-way local conferencing, call forwarding, transfer, hold, voice mail, presence, HTTPS secure provisioning Three-microphone design guarantees 360-degree room coverage Can be centrally provisioned and upgraded in the field together with Polycom SIP desktop phones 254 Snom HS-MM2 Headset 31.0000 35.00 Snom VoIP Accessories http://www.voipon.co.uk/snom-hsmm2-headset-p-254.html http://www.voipon.co.uk/images/snom_hs_mm2_headset.jpg new Availability: In stock Snom headsets have been designed to incorporate improved ergonomics and usability. With the snom headsets, snom provides a complete product line for use in a variety of voIP contexts: SMEs (small and medium-sized enterprises), home offices as well as call centres. The HS-MM2 headset is designed for use with the snom 320, 360 and snom 370 phones. For a Snom 300 headset, please take a look at the HS-MM3 . The snom HS-MM2 and HS-MM3 monaural headsets provide users with more headset stability, ease of use, comfort, and hands-free convenience throughout daily use. The snom monaural headset has one ear pad which can be worn on either ear. With the noise-canceling microphone, the snom headset keeps your voice crystal clear. The snom Headsets are an excellent investment for your business, allowing users working in telephone intensive applications more flexibility, freedom and comfort. The use of headsets optimizes the operating process whilst increasing work productivity. 255 Asterisk - The Future of Telephony 12.95 12.95 Asterisk Books http://www.voipon.co.uk/asterisk-the-future-of-telephony-p-255.html http://www.voipon.co.uk/images/asterisk_future_telephony_s.gif new Book Description: Discover the open source application that has traditional telephone providers running scared! Asterisk allows you to implement flexible dialplans that support just about any telephony application youwant; you can configure it to use traditional analogue phones and trunks, in addition to VoIP phones from any vendor and VoIP services from any standards-compliant VoIP service provider. Because it is so powerful, configuring Asterisk can seem tricky and difficult. This book will walk you through the proces of configuring your frst Asterisk system. Along the way, you'll learn how to: Prepare a system for Asterisk, and install it Configure Asterisk to use analog phones and trunks Configure Asterisk to use the SIP and IAX VoIP protocols Write dialplans, from the simple to the more complex Set up applications, from find-me-follow-me to Music on Hold Set up features such as speech synthesis and voice recognition Script Asterisk, using the Asterisk Gateway Interface (AGI) Manage Asterisk Installations FREE PDF Download available HERE 256 Rachel Female British Asterisk Voice Prompt 69.9800 69.92 Asterisk Voice Prompts http://www.voipon.co.uk/rachel-female-british-asterisk-voice-prompt-p-256.html http://www.voipon.co.uk/images/rachel.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Rachel to handle your calls? Rachel is a British English, female, full Asterisk Pack. All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Technical Specifications The Voice prompts are recorded and processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. We can provide the packs in other audio formats if you need them. This pack includes prompts for: Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetics Listen to some samples: (in 8 bit mono. but 44.1khz) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav (in 8 bit mono. Same as the phones sound) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav 257 Paul Male British Asterisk Voice Prompt 69.9800 69.92 Asterisk Voice Prompts http://www.voipon.co.uk/paul-male-british-asterisk-voice-prompt-p-257.html http://www.voipon.co.uk/images/paul.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Paul to handle your calls? Paul is a British English, male, full Asterisk Pack. All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Technical Specifications The Voice prompts are recorded and processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. We can provide the packs in other audio formats if you need them. This pack includes prompts for: Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetics Listen to some samples: (in 8 bit mono. but 44.1khz) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav (in 8 bit mono. Same as the phones sound) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav 258 John Male British Asterisk Voice Prompt 69.9800 69.92 Asterisk Voice Prompts http://www.voipon.co.uk/john-male-british-asterisk-voice-prompt-p-258.html http://www.voipon.co.uk/images/john.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ John to handle your calls? John is a British English, male, full Asterisk Pack. All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Technical Specifications The Voice prompts are recorded and processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. We can provide the packs in other audio formats if you need them. This pack includes prompts for: Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetics Listen to some samples: (in 8 bit mono. but 44.1khz) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav (in 8 bit mono. Same as the phones sound) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav 259 Carine Female French Asterisk Voice Prompt 69.9800 68.32 Asterisk Voice Prompts http://www.voipon.co.uk/carine-female-french-asterisk-voice-prompt-p-259.html http://www.voipon.co.uk/images/carine.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly French, why give them the impression you are American? Why not employ Carine to handle your calls? Carine is a French, female, full Asterisk Pack. All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Technical Specifications The Voice prompts are recorded and processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. We can provide the packs in other audio formats if you need them. Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetic Listen to some samples: (in 8 bit mono. but 44.1khz) demo-abouttotry.wav voicemail-msginstruct.wav voicemail-options.wav (in 8 bit mono. Same as the phones sound) demo-abouttotry.wav voicemail-msginstruct.wav voicemail-options.wav 261 Grandstream GXV-3000 SIP Videophone 143.0000 155.00 Grandstream Grandstream IP Video Phone http://www.voipon.co.uk/grandstream-gxv3000-sip-videophone-p-261.html http://www.voipon.co.uk/images/grandstream_gxv_3000.jpg new Availability: In stock GXV-3000 brings unrivaled sharp picture quality, compelling functionality richness, stylish exterior design, intuitive user friendliness and unprecedented affordability to personal video communications over the Internet. It supports the latest H.264 international real-time video codec standard (base profile) with QVGA or CIF resolution, up to 30 fps frame rate, and at bit rates between 64Kbps and 1Mbps. This makes commercial quality, personal video conferencing practical for all broadband users, even those who have access to only modest bandwidth. Grandstream GXV-3000 SIP VideoPhone H.264 Protocol Features: SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP/HTTPS, ICMP, ARP/RARP, DNS. DHCP (client and server), NTP, PPPoE, TFTP, Telnet and TLS (pending) Dual 10/100 auto-sensing ethernet ports configurable for either switch or router mode Powerful video DSP with advanced adaptive jitter control and packet loss concealment technology to ensure superb audio and video quality Advanced H.264 base real time video code (at CIF or QVGA resolution and up to 30 frames/second) to ensure the highest quality video at 32kbps - 1Mbps bandwidths. Configurable bit rate and frame rate Various Audio vocoder support including G.711 A-law/U-law, G.723.1, G.729A/B, G.726 (pending), GSM (pending) and wideband G.722 (pending) with dynamic negotiation of of codec type and packet time Support for popular voice features including 3 line indicators (each of which can be configured using independent SIP accounts) Caller ID/Name display or block, Hold, Call Waiting, Call Transfer, Call Forward on no answer/busy or unconditional), Do Not Disturb, various DTMF options (in audio, RFC2833, SIP INFO) Full Duplex hands-free speaker phone with advanced acoustic echo cancellation (G.167) 3-way conferencing Downloadable music ring-tones Advanced Video features including 5.6 inch TFT colour LCD (allowing 2 dimensional view angle adjustment including 180 degrees vertical rotation and nearly 300 degrees horizontal rotation Advanced VGA resolution camera sensor (adjustable view angle) Anti Flickering Auto Focus and Auto Exposure Zoom Picture-in-Picture (PIP) Audio Mute and camera block (for privacy) Call Log Video Phone book Configurable Screen Saver pictures Still picture capture/store/send (VGA resolution) Visual Voice Message Waiting Indicator Intuitive GUI enabled by 5 navigation buttons 2 USB 2.0 host ports 1 Audio and 1 Video output jack (capable of outputting to an external TV) Headset Jack Encryption and Authentication using Digest (MD5 and MD5-sess) and AES Secure Signalling (SIP over TLS, pending) and secure voice/video communication (SRTP) (pending) Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 (DiffServ, ToS) QoS Automated NAT traversal without manual manipulation of firewall/NAT Remote automated and secure provisioning and software upgrade through firewall/NAT to enable "zero-configuration" and "plug-and-dial" for end users Remote device monitoring and event reporting using Syslog Device configuration via LCD/Keypad, Web Browser or central secure configuration file (TFTP) Grandstream GXV-3000 Technical Specifications: Ethernet Port Dual 10/100Mb auto-sensing, switch or router mode LCD 5.6 inch TFT colour LCD Camera Advanced CMOS sensor (VGA resolution) Auxiliary Ports RCA style stereo audio and composite video output jack Headset jack 2 USB 2.0 ports Exterior Silver ABS plastic, 30 buttons (6 with LEDs) Universal PSU 100-240v input, _12VDC/1.2A output. Dimension 6.5cmx18cmx16cm Weight 1.2Kg Operating Temp 0-40C Humidity 10-95% non-condensing Compliance FCC/CE/C-Tick (pending) 262 Digium Wildcard B410P BRI ISDN Card 355.5000 395.00 Digium Digium Digital (BRI) Card http://www.voipon.co.uk/digium-wildcard-b410p-bri-isdn-card-p-262.html http://www.voipon.co.uk/images/digium_b410P.jpg new Availability - In stock Digium's Quad S/T BRI interface with Echo Cancellation The B410P is a half-length universal PCI 2.2-compliant Basic Rate ISDN card that supports TE and NT mode over 4-wire S/T interface. Using Digium's digital hardware, Open Source Asterisk PBX software, and a Linux PC, users can create an intelligent telephony environment which includes all the sophisticated features of a high-end PBX/ Voicemail platform. The B410P is a half-length, full-height universal 3.3V and 5.OV 32-bit PCI 2.2 card supporting four BRI S/T interfaces. Each of the four ports of the B410P can be independently configured for TE or NT mode, with optional PWR400M module for supplying power to ISDN telephones. The B410P features on-board hardware echo cancellation performing 64ms or 512 taps per channel for each of the eight voice channels. Target Applications PC based PBX VoIP gateways ISDN monitoring ISDN recording Physical Specifications Size: 3.75" x 5.5", not including the bracket Weight: 1.1 ounces.; 31.2 grams Environment Conditions Operation Range: 0° to 50°C, 32° to 122° F Storage Range: -20° to 65°C, 4° to 149° F Humidity: 10-90% non-condensing Configurable Modes EuroISDN TE EuroISDN NT PCI Interface 32-bit 33MHz Universal PCI 2.2 compliant Hardware Echo Cancellation 64ms (512 taps) of echo cancellation per B-channel Hardware Requirements 500-MHz Pentium III or better with 64MB of RAM Available 3.3V or 5.OV PCI Slot NT Power requirements PWR400M module Available 4-pin hard drive connector 263 Hitachi Wireless IP 3000 133.0000 139.00 Hitachi Hitachi IP Telephone http://www.voipon.co.uk/hitachi-wireless-ip-3000-p-263.html http://www.voipon.co.uk/images/hitachi_ip_3000.jpg new Availability: Due to a component shortage Hitachi have advised the IP 3000 will be available Late September Hitachi's stylish new Wireless IP3000 is a SIP based 802.11b Wireless IP phone. Aimed at the consumer or SMB business user the Wireless IP 3000 boasts many of the features of the market leading Wireless IP5000, but at a lower price point. Telephone Feature Phone book Capacity: 200 entries; Call history - Display a log of up to 20 incoming/outgoing calls Call time Display Talk time display Dialed number display - Display dialed phone numbers Calling extension number display - Display extension number when on hold Incoming call extension number display - Opposite extension number (including office number) display at incoming call Call Waiting Call waiting using a holding tone Dial-tone forwarding - Forward dial tone (DT) ID linking* - Select different ringtones depending on whether a call is coming from an outside line or an extension Mute - Voice not sent to opposite party Speed dial* (one-touch dial) - Registration and use of speed dialing Redial* - Press and hold the button to redial the last number called Call-back transfer* - While on the phone with A (inside or outside line), put A on hold, call B, then use the transfer button to connect A and B Call pickup* - Answering by special number when there was an incoming call in a pickup group Call waiting* - When you receive a call while already on the line, a feature similar to call waiting on fixed phones is available (notification via a tone, change connections by pressing button) Unanswered call display - A pop-up displays unanswered calls 264 Linksys SPA-9000 IP PBX 172.9000 182.00 Linksys Linksys IP PBX http://www.voipon.co.uk/linksys-spa9000-ip-pbx-p-264.html http://www.voipon.co.uk/images/linksys_spa-9000.jpg new Availability: In stock The SPA-9000 marries the rich feature set of high-end PBX telephone systems with the convenience and cost advantages of Voice over IP. It has common voice system features such as an auto-attendant, shared line appearances, three way call conferencing, intercom, music on hold, call-forwarding and much more. The SPA-9000 opens up access to the benefits of VoIP, including low cost long distance service, telephone number portability, and one network for both voice and data. The SPA-9000 is so easy to configure that a fully working system can be set up in minutes. New telephones are automatically detected and registered when they are connected to the SPA-9000. The SPA-9000 has an integrated web server that allow features to be configured using a web browser. The web server has multiple levels of password protected access to user and service level features. Service level settings may be locked by the Internet Telephone Service Provider to ensure they are not inadvertently corrupted. The Internet Telephone Service Provider also can remotely update the software and settings through a secure encrypted connection. With its integrated router, the SPA-9000 can be either connected directly to the internet connection or to another router on your network. The SPA-9000 has separate WAN and LAN Ethernet ports. The WAN connection can be connect through DHCP or a fixed IP address. The LAN port can assign IP addresses to IP telephones and computers using NAT and DHCP. While the SPA-9000 will work with any SIP compatible IP telephone, it is the ideal host for Linksys business telephones, such as the SPA901, SPA921, SPA922, SPA941, and SPA942. Powerful configuration capabilities enable the SPA-9000 to support a greater set of advanced features with these telephones, such as shared line appearances, hunt groups, call transfer, call parking lot, and group paging. With its two FXS ports, the SPA-9000 can support traditional analog devices such as telephones, answering machines, FAX machines, and media adapters. Features SIP Application Server, Proxy, Registrar and Location Server (RFC3261) Multiple Service Provider Lines / SIP Account Support (4) Shared Line Appearance (SLA) Automated Attendant (AA) Configurable AA Answer Delay Interactive Voice Response (IVR) Recordable IVR Prompts Automatic Call Distribution (ACD) Configurable Call Routing - Least Cost Routing - Multiple DID Numbers Per VoIP Line - Call Routing to Multiple Extensions or Targeted User - Call Hunting - Sequential, Round Robin, Random Phone Configuration and Management Server - Discovery and Configuration of IP Phones - Assignment of Extension - Assignment of Dial plan Hot Line and Warm Line Automatic Calling Call Log (60 entries each): Made, Answered, Missed Calls Personal Directory with Auto-dial (100 entries) Do Not Disturb for the SPA-9000 URI (IP) Dialing Support (Vanity Numbers) On Hook Default Audio Configuration (Hands Free/Headset) Multiple Ring Tones with Selectable Default Ring Tone per Line Called Number with Directory Name Matching Calling Number with Name - Directory Matching or via Caller ID Subsequent Incoming Calls with Calling Name and Number Date and Time with Intelligent Daylight Savings Support Call Duration with Call Time Stamp Stored in Call Logs Name/Identity (Text) Display at Start Up Distinctive Ringing Based on Calling and Called Number User Downloadable Ring Tones and Ring Tone Generator (Free from www.linksys.com) Download on Demand Ring Tones - 10 Speed Dial Support Configurable Dial/Numbering Plan Support - per Line DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy Syslog, Debug, Report Generation and Event Logging Secure Call Encrypted Voice Communication Support Built-in Web Server for Admin and Config with Multiple Security Levels Automated Provisioning, Multiple Schemes-Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP) Require Admin Password to Reset Unit to factory Defaults Option Compliance Security ** Service feature availability is call feature server platform dependent. FCC (Part 15 Class B), CE, A-Tick, ICES-003 Password Protected System Reset to Factory Default Password Protected Admin and User Access Authority HTTPS with Factory Installed Client Certificate HTTP Digest - Encrypted Authentication via MD5 (RFC 1321) Up to 256-bit AES Encryption for the SPA-9000 Power, Ethernet (WAN), Phone 1, Phone 2 Quick Installation and Configuration Guide, User Guide, Administration Guide - Service Providers Only, Provisioning Guide - Service Providers Only 1 - SPA-9000 System 1 - 5 Volt Power Adapter 1 - RJ45 Ethernet Cable 1 - Quick Installation Specification Note: Many specifications are programmable within a defined range or list of options. Please see the SPA Administration Guide for details. The target configuration pro-file is uploaded to the SPA-9000 at the time of provisioning. Data Networking MAC Address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) ARP - Address Resolution Protocol DNS - A Record (RFC 1706), SRV Record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) DHCP Server - Dynamic Host Configuration Protocol (RFC 2131) PPoE Client - Point to Point Protocol over Ethernet (RFC 2516) ICMP - Internet Control Message Protocol (RFC792) TCP - Transmission Control Protocol (RFC793) UDP - User Datagram Protocol (RFC768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349) VLAN Tagging - 802.1p/q SNTP - Simple Network Time Protocol (RFC 2030) Upload Data Rate Limiting - Static and Automatic QoS - Voice Packet Prioritization over Other Packet Types Router or Bridge Mode of Operation MAC Address Cloning Port Forwarding Voice Gateway SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264) SIP Proxy Redundancy - Dynamic via DNS SRV, A Records Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks - NAT (incl. STUN) Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP Codec Name Assignment Voice Algorithms: - G.711 (A-law and µ-law) - G.726 (16/24/32/40 kbps) - G.729 A - G.723.1 (6.3 kbps, 5.3 kbps) Dynamic Payload Support Adjustable Audio Frames Per Packet DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO) Flexible Dial Plan Support with Inter-Digit Timers IP Address / URI Dialing Support for the SPA-9000 Call Progress Tone Generation Jitter Buffer - Adaptive Frame Loss Concealment VAD - Voice Activity Detection w/ Silence Suppression Attenuation / Gain Adjustments MWI - Message Waiting Indicator Tones VMWI - Via NOTIFY, SUBSCRIBE Caller ID Support (Name & Number) Provisioning, Administration & Maintenance Web Browser Administration & Configuration via Integral Web Server Telephone Key Pad Configuration of Select Networking Parameters via IVR Maintenance: - Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP - Asynchronous Notification of Upgrade Availability via NOTIFY - Non-intrusive, In-Service Upgrades - Report Generation & Event Logging - Stats in BYE Message - Syslog & Debug Server Records - Per Line Configurable Physical Interfaces: 2 10/100BaseT RJ-45 Ethernet Port (IEEE 802.3) -- 1 WAN, 1 LAN 2 RJ-11 FXS Phone Ports - For Analog Circuit Telephone Device (Tip/Ring) Subscriber Line Interface Circuit (SLIC) Ring Voltage: 40-55 VRMS Configurable Ring Frequency: 10 Hz - 40 Hz (SLIC): Ring Waveform: Trapezoidal and Sinusoidal Maximum Ringer Load: 3 REN On-hook/off-hook Characteristics: On-hook voltage (tip/ring): -50 V NOMINAL, Off-hookcurrent: 25 mA min, Terminating Impedance: 8 Configurable Settings including North America 600 ohms, European CTR21Switching Type (100-240v) Automatic - Proxy Logging of SIP Messages - Phone Firmware Upgrade Management Corporate Directory with Automatic Update Configuration and Maintenance via Web Interface (Local or Remote - Status Display of All Connections Remote Configuration via - HTTPS with XML Formatted Files - HTTP or TFTP with 256-Bit Encrypted Binary Files Call Park - User Definable Parking Space Number Call Unpark Call Transfer Call Forward Group Paging Intercom Directed Call Pick Up Group Call Pick Up Music / Information via Streaming Audio Server (SAS) for Calls: - On Hold - Parked in the Parking Lot - Being Transferred Simultaneous Ringing (Find Me Service) Do Not Disturb Voice Mail Integration - Service Provider Based - Voice Mail Notification via SUBSCRIBE / NOTIFY - Forward Call Directly to Voice mail Integrated Media Proxy or Direct RTP Routing to ITSP Differentiated Services (DiffServ) / Type of Service (TOS) Support Two FXS Ports for Phones, Fax machines, Media Adapters Voice encoding according to G.711 (64kbit/s) Fax Support using G.711 Pass-Through or T.38 Echo Cancellation (G.165) Line Status - Active Line Indication, Name/Number Digits Dialed with Number Auto-Completion Call Hold Call Waiting for the SPA-9000 Call Transfer - Attended and Blind Call Conferencing Automatic Redial Call Pick Up - Selective and Group ** Call Swap Call Forwarding - Unconditional, No Answer, On Busy 266 Snom 300 PoE IP Telephone 67.0000 72.00 Snom Snom IP Telephones http://www.voipon.co.uk/snom-300-poe-ip-telephone-p-266.html http://www.voipon.co.uk/images/snom_300_small.jpg new Availability: In stock The snom PoE 300 is the base model of the snom business telephone range and fulfils the most important requirements of VoIP telephony whilst offering numerous functions that are indispensable in the business world. For efficient and effective day-to-day work, the snom PoE 300 provides all important office functions such as choice of trunk line, status display, group lines, the engaged option or picking up calls. When it comes to user friendliness, the snom PoE 300 sets new standards: A two-line graphical LCD display enables the display of call information, and the menu-driven user interface provides the simplest of feature management. Via the navigation key, the user is guided intuitively through the telephone menu. More complex telephone functions, call details and configuration possibilities are accessible via the browser over the connected PC. Six free user or administrator-configurable (or carrier-preconfigurable) function keys can be easily allocated to security-related menu functions, or assigned to multiple lines. The snom 300 comes factory-equipped to enable two of its six programmable keys to be configured as line appearances, and snom provides upgrades that let you configure (up to) all six function keys in this way - flexible enough to suit the needs of every user. This option enables an individual adaptation of the device to specific areas of application and the personal user behavior - a functionality that is becoming increasingly popular, particularly in call centers and for sales agents. The snom 300 is designed for different environments: for small offices, call centers, lobbies, recreation rooms, or in the home. It fits into its environment without any troubles. Through numerous telephone functions, the demands of everyday office communication can be easily managed. The snom 300 has a headset connection and can be used as a freestanding or wall-mounted model. As the snom 300 supports all of the common compression codecs such as G.729a and G.723.1, it is compatible with numerous components of other manufacturers and can be used in low-bandwidth environments. An integrated 2 Ethernet port switch enables connection to the network over an RJ-45 interface simultaneously with the PC connection! Main Features / Specification Two-line display (2 x 16 characters) 27 keys, 7 LEDs 6 programmable function keys 2 Ethernet ports 2 multi-line registration, or option of 6 Headset connection SIP RFC3261 Security: SIPS/SRTP, TLS STUN, ENUM, NAT, ICE Codecs: G.711, G.729A, G.723.1, G.722, G.726, GSM Snom IP Phone Comparison Table Phone Power over ethernet Display SIP Identities Programmable Keys with LEDs Large LED for incoming calls Extension Keyboard Available Wireless headset adaptor Snom 300 Y 2 Line 16 Characters 4 6 N N N Snom 320 Y Hinged 2 line character display with graphical field 12 12 N Y Y Snom 360 Y Hinged, backlit graphical display (128x64 pixels) 12 12 N Y Y Snom 370 Y Hinged, backlit, high-definition graphical display (240x128 pixels) 12 12 Y Y Y   267 Snom 320 360 370 Expansion Module 72.0000 74.85 Snom VoIP Accessories http://www.voipon.co.uk/snom-320-360-370-expansion-module-p-267.html http://www.voipon.co.uk/images/snom_exp_small.jpg new Availability: In Stock With the snom 320, 360 and 370 Expansion Module, your snom 320, 360 and 370 has the best features to meet the requirements of executive and administrative staff who require high call coverage and flexibility from communication systems. Your snom 320, 360 and 370 VoIP phone is equipped with 12 programmable keys and LEDs. Using your snom 320, 360 and 370 with the snom Expansion Module, call coverage and management is even more easy and efficient to handle. With the expansion module, you add 42 additional programmable keys and LEDs, increasing the total number of keys and LEDs to 54. More comfort is provided by the built-in web interface of your snom 360 VoIP phone. With the built-in web interface, you can easily configure each function key according to your preferences. Each key of your expansion module can be programmed for Line, Destination, Intercom, Park Orbit, or Voice Recorder mapping. Line or call state, e.g. ringing, on-hold, connected call etc., will be shown by the LEDs associated with each key. Snom IP Phone Comparison Table Phone Power over ethernet Display SIP Identities Programmable Keys with LEDs Large LED for incoming calls Extension Keyboard Available Wireless headset adaptor Snom 300 N 2 Line 16 Characters 4 6 N N N Snom 320 Y Hinged 2 line character display with graphical field 12 12 N Y Y Snom 360 Y Hinged, backlit graphical display (128x64 pixels) 12 12 N Y Y Snom 370 Y Hinged, backlit, high-definition graphical display (240x128 pixels) 12 12 Y Y Y 268 UK VoIP Geographic Numbers 19.0000 21.99 Incoming Numbers http://www.voipon.co.uk/uk-voip-geographic-numbers-p-268.html http://www.voipon.co.uk/images/0102.gif new Availability: Instant Activation For £21.99 per year you can choose to have a UK Geographic number. This allows people to call you from UK landlines and mobiles at the standard rates for their telephone service.   With VoIPon geographic numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. Geographic numbers are the normal way of numbering landlines and denote your location. So that in the old telecommunication world, a 0207 number is fixed to London and a 01273 number is fixed to Brighton. However, with Internet Telephony you can choose ANY geographical number, no matter where you are actually located. You could even take a Birmingham number and be actually in Calcutta. (And, it's worth repeating, calls to this number from any landline in the UK will only be charged at normal UK rates - even if the call does finally finish up in Calcutta!) There are many uses for these numbers; the obvious one being that a business can have a prestigious telephone address whilst being somewhere else entirely. You can also now move your office to anywhere in the world and keep this telephone number. For the personal user, it means that if you have moved abroad but still have family in the UK you could take a number for the town your relative lives in, they can then make ordinary calls to you, no matter where in the world you are, for the cost of a local call. DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 269 0870 UK VoIP National Number 2.5 2.50 Incoming Numbers http://www.voipon.co.uk/0870-uk-voip-national-number-p-269.html http://www.voipon.co.uk/images/0870.gif new Availability: Instant Activation For a one off payment of only £2.50 you can choose to have a Non-Geographic 0870 Telephone Number. Your caller will pay the standard rates which vary between 1.5p and 8p/min depending on time of day and service provider. People calling your 0870 number from outside the UK will normally be charged at their standard international calling rate.   With VoIPon 0870 numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. (note: calls to 0870 numbers from mobiles are charged at differing rates by Cellular Service Providers) DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 271 Xorcom TS-1 IP PBX 470.4000 483.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-ts1-ip-pbx-p-271.html http://www.voipon.co.uk/images/xorcom_ts-1_ip_pbx.gif new Availability: In Stock Xorcom TS-1 Asterisk-based iPBX TS-1 is an out-of-the box Asterisk based iPBX. Its 100% solid state architecture provides maximum reliability for the long run. Configuration and set-up are possible from distant locations via a Web management portal and do not require connection of keyboard and monitor. For more advanced features, the TS-1 offers SSH or direct (keyboard and monitor) access to the system menus. System set-up is extremely straightforward. The TS-1 includes automatic detection and configuration of Zaptel hardware. A default system can be set up in less than 5 minutes! TS-1 has one PCI slot for internal telephony cards and works seamlessly with all Astribank models. TS-1 has been tested with different codecs and at varying stress levels. Features: Powered by Asterisk Includes AMP &mdash; configuration and maintenance via Web interface Runs Flash Operator Panel Runs Zaptel auto&mdash;configuration system Includes one PCI slot for telephony cards Compatible with Xorcom Astribank&trade; telephony interface Hardware Specifications: * VIA C3 1GHz processor * 256 MB DDR memory * 512 MB flash memory * VIA PCI 10/100 Ethernet * 4 x USB type 2 * VIA S3G Unicrome video with DVI and SVGA * Very low power consumption &mdash; 36 Watts * 100% solid state embedded unit &mdash; no moving parts Supported Hardware for Xorcom TS-1 Xorcom Astribank family Digium analog cards and compatible: TDM400P X100P Digium PRI cards and compatible Junghanns quadBRI and octoBRI ISDN cards HFC-S-based PCI ISDN (BRI) cards 8 Acer ISDN-Surf Billion Bipac ISDN Billion/Asuscom (Asuscom/Askey) Conceptronic ISDN PCI card C128i(r) Creatix ISDN-S0/PCI D-LINK DMI-128+ HFC cards from "Conrad Elektronik" Longshine LCS-8051 Neolec FREEWAY ISDNPCI Trust PCI ISDN Modem I-tec ISDN-128 274 0871 UK VoIP National Number 0 0.00 Incoming Numbers http://www.voipon.co.uk/0871-uk-voip-national-number-p-274.html http://www.voipon.co.uk/images/0871.gif new Availability: Instant Activation For FREE you can choose to have a national rate 0871 telephone number. This number is charged to the caller at the same rate as a standard rate UK telephone call.   With VoIPon 0871 numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. (note: calls to 0870 numbers from mobiles are charged at differing rates by Cellular Service Providers) DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 275 0845 UK VoIP Local Number 5.0000 7.50 Incoming Numbers http://www.voipon.co.uk/0845-uk-voip-local-number-p-275.html http://www.voipon.co.uk/images/0845.gif new Availability: Instant Activation For a one off payment of only £7.50 you can choose to have a local rate 0845 number. This allows people to call you from landlines at local call rates no matter where they are in the UK. With VoIPon 0845 numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. 0845 numbers are normally used to provide a friendly, low cost access number for your customers and can be used regularly for your company's main number or one-off for sales promotions etc. (note: calls to 0845 numbers from mobiles are charged at differing rates by Cellular Service Providers DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 276 0709 UK VoIP National Number 7.5 7.50 Incoming Numbers http://www.voipon.co.uk/0709-uk-voip-national-number-p-276.html http://www.voipon.co.uk/images/0709.gif new Availability: Instant Activation For a one off payment of only £7.50 you can choose to have a local UK national rate 0709 telephone number . This number is charged to the caller at the same rate as a standard rate UK telephone call (5p / min). With VoIPon 0709 numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 277 Utstarcom F3000 Wireless IP Telephone 132.0000 134.95 UTstarcom UTstarcom IP Telephone http://www.voipon.co.uk/utstarcom-f3000-wireless-ip-telephone-p-277.html http://www.voipon.co.uk/images/f3000_ip_telephone.jpg new Availability: In stock The UTstarcom F3000 promises increased security with the addition of WPA encryption, which utilizes the temporal key integrity protocol (TKIP). TKIP utilizes a hashing algorithm to scramble the keys and, by adding an integrity-checking feature, ensures that the keys have not been tampered with. User authentication, which is generally missing in WEP, through the extensible authentication protocol (EAP). WEP regulates access to a wireless network based on a computers hardware-specific MAC address, which is relatively simple to be sniffed out and stolen. EAP is built on a more secure public-key encryption system to ensure that only authorized network users can access the network. Product Features Clamshell Type Design 65K CSTN 1.8", 128x160 pixels 32 Chord Polyphonic Ring Tone 802.11b/g for the F3000 VoIP support SIP (Session Initiation Protocol) DHCP (Dynamic Host Configuration Protocol) Support POP3/SMTP Email SIP-Message 3-way Calling / Call Waiting Call Transfer / Call Forwarding Detailed Feature List 802.11b/g WEP (64 and 128 bit )/WPA MD5 Authentication (Phase II) Call Features Three-way calling / Call waiting Call rejecting/redial/mute for the F3000 Call transfer / Call forwarding Call hold/resume Voice Process Codec G.711, G.729a/b, G.726 Comfort noise generation (CNG) Voice activity detection (VAD) Adaptive jitter buffer Echo suppression Advanced Features Handover/Roaming between APs Net Search Auto Scan TFTP / HTTP provisioning Remote / local firmware upgrade Web Configuration Interface for the F3000 STUN support 3 sets of SIP Profiles Protocols SIP (Session Initiation Protocol) SDP (Session Description Protocol) RTP (Real-Time Transfer Protocol)/RTCP DHCP (Dynamic Host Configuration Protocol) TFTP (Trivial File Transfer Protocol) Web Browser (Phase II) Phonebook SyncML (Phase II) for the F3000 Powerful Personal Information Management Colourful Schemes/Screensaver 278 RJ11 Adaptor with Ring Capacitor 2.49 2.49 VoIP Accessories http://www.voipon.co.uk/rj11-adaptor-with-ring-capacitor-p-278.html http://www.voipon.co.uk/images/rjp_bts_small.gif new An adapter to enable a standard UK phone to be connected to an American standard socket. This adaptor includes a ring circuit which enables the phone to ring. 279 Linksys SPA-942 IP Telephone 74.0000 77.90 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa942-ip-telephone-p-279.html http://www.voipon.co.uk/images/linksys_spa_942_small.jpg new Availability: In Stock The new Linksys SPA-942 is a functionally powerful, easy to use business phone, featuring comprehensive interoperability and a secure mass deployment capability. Standard features on the SPA-942 include two active lines, dual switched Ethernet ports, Power-over-Ethernet, a high resolution graphical display, a full duplex speaker-phone and a 2.5mm head-set port. The SPA-942 is an affordable, business class telephone which can be easily configured as a two line phone. With a simple software update the SPA-942 is upgradeable to four lines. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones. The SPA-942 can be used in SOHO, enterprise, small to medium business service offerings including IP PBX, hosted IP telephony and IP Centrex along with residential use. The SPA-942 is based on industry leading VoIP technology from Linksys to deliver a high quality IP phone that is unparalleled in features, value and support. The SPA-942 is simple to use, with well laid out function keys & utilises the call processing functionality found in existing Linksys products. Key Features Up to 4 lines with independent configuration Dual switched Ethernet ports Hands free option Call hold, call waiting, call transfer, call conferencing, Do not disturb. Call logs- made, answered, missed calls, with time- 60 entries each Caller ID Multiple ringtones with selectable default ring tone per line Personal address book with auto dial- 100 entries Speed dialling Built in web server for administration 128x64 Monochrome LCD graphical display with back light Automated provisioning- up to 256 byte encryption - TFTP - DHCP option 66 support - HTTP - HTTPS Data Networking MAC Address (IEEE 802.3) IPv4- Internet Protocol Version 4 (RFC 791) ARP- Address Resolution Protocol DNS- A Record (RFC 1706) SRV Record (RFC 2782) DHCP Client- Dynamic Host Configuration Protocol (RFC 2131) ICMP- Internet Control Message Protocol (RFC 792) TCP- Transmission Control Protocol (RFC 793) UDP- User Datagram Protocol (RFC 768) RTP- Real Time Protocol (RFC 1889) (RFC 1890) RTCP- Real Time Control Protocol (RFC 1889) DiffServ (RFC 2475) Type of Service- TOS (RFC 791/1349) VLAN Tagging 802.1p/q - Layer 2 Qos SNTP- Simple Network Time Protocol (RFC 2030) Voice Over IP SIPv2- Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264) SIP Proxy Redundancy- Dynamic via DNS SRV, A records Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks- NAT (incl. STUN) SIPFrag (RFC 3420) Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP Codec Name Assignment Voice Algorithms G.711 (A-law and mu-law) G.726 (16/24/32/40 kbps) G.729 A G.723.1 (6.3 kbps, 5.3 kbps) Dynamic Payload Support Adjustable Audio Frames per Packet DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO) Flexible Dial Plan Support with Configurable Inter-Digit Timers IP Address/URI Dialing Support Call Progress Tone Generation Jitter Buffer- Adaptive Frame Loss Concealment VAD- Voice Activity Detection w/ Silence Suppression Attenuation/Gain Adjustments MWI- Message Waiting Indicator Tones VMWI- Voice Mail Waiting Indicator Caller ID Support (Name & Number) Third Party Call control (RFC 3725) Security Password Protected System Reset to Factory Default and Configuration Password Protected Admin and User Access Authority Provisioning/Configuration/Authentication HTTPS with Factory Installed Client Certificate HTTP Digest- Encrypted Authentication via MD5 (RFC 1321) -Up to 256-bit AES Encryption Provisioning, Administration and Maintenance Web Browser Administration and Configuration via Integral Web Server Automated Provisioning & Upgrade Availability via HTTPS Non-Intrusive, In-Service Upgrades Report Generation and Event Logging Syslog and Debug Server Records Per Line and Purpose Configurable Syslog and Debug Options Physical Data Interfaces 2 x 100baseT RJ-45 Ethernet Ports (IEEE 802.3) Voice I/O Interfaces Handset: RJ-7 Connector Speaker-phone & Microphone- Built-in Headset 2.5mm Port Regulatory Compliance FCC Part 15 Class A, B CE Mark Power Supply Switching Type with Modular Wall Plug Clip- County/Region Specific DC Input Voltage: +5 VDC at 2.0 A Max Power Consumption: 5 WATTs Power Adaptor: 100-240v- 50-60Hz (26-34VA) AC Input, 1.8m cord Indicator LEDs- Lights Line (4) Speaker-phone Headset Message Waiting Mute + Status (Provision, Alert, Upgrade) Voicemail Message Retrieval button Environmental Operating Temperature 32 to 113 F (0 to 45 C) Storage Temperature -13 to 185 F (-25 to 85 C) Relative Humidity 10 to 90% non-condensing, operating and non- operating Box Contents 1- Linksys IP Phone- Colour- Dark Grey 1- Handset- Colour- Dark Grey 1- Handset Cord- Colour- Dark Grey 1- Telephone Desk Stand 1- 5v Power Adapter- 1.8m (3ft) Cord ( purchased separately ) 1- RJ-45 Ethernet Cable- 1.8m (3ft) Cord- Colour Black 1- Linksys Quick Start Guide Documentation Quick-Start Guide User Guide Administration Guide- Available to Service Providers Only Provisioning Guide- Available to Service Providers Only The Linksys SPA 942 requires a Power adapter if your network does not support PoE (power over ethernet). Please select the appropriate option from the drop-down menu below. Linksys SPA Series VoIP Telephone (SIP) Comparison Table SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernet Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 280 Custom Voice Prompts 135 135.00 Asterisk Voice Prompts http://www.voipon.co.uk/custom-voice-prompts-p-280.html http://www.voipon.co.uk/images/custom.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order Customise the prompts with your own message for £174.99 per prompt of 300 words or less. Each of the recorded packs can be customised as needed and are available for download shortly after the order has been processed. Usually within a couple of days depending on demands on the voice artists. Note. You will need to have purchased the appropriate Asterisk pack before doing any customisations. If, by mistake, it seems that you request a customisation from the wrong voice artists we will get in touch before proceeding, just to make sure. 281 Grandstream GXP-2000EXT Expansion Module 39.0000 48.00 Grandstream VoIP Accessories http://www.voipon.co.uk/grandstream-gxp2000ext-expansion-module-p-281.html http://www.voipon.co.uk/images/grandstream_2000-em_sml.jpg new Availability: In Stock with the GXP-2000EXT Expansion Module, it is now possible to transform your Grandstream GXP-2000 into a high-performance communications centre! Your IP telephone will be able to handle ranges of phone extensions, perform switchboard operations and other business functions where these type of abilities are required. Up to 112 freely programmable keys The new GXP-2000EXT expansion module offers 56 keys that have a 2-coloured LED. Each of these keys is freely programmable and can be set with speed dial. The GXP-2000 EXT attaches with a short cable (included) and integrates almost seamlessly with the GXP-2000 for looks and function. By attaching two of these devices you can have up to 112 programmed buttons. New firmware functionality The new firmware upgrade for GXP2000 supports speed dialing, BLF (Busy Lamp Field), call transfer/forward/pickup on each of the programmable buttons on GXP-2000EXT module.   Features & Benefits 56 programmable buttons Call transfer/forward/pickup Busy Lamp Field (BLF) Busy Line Answer (BLA) Speed dial for each line Daisy-chain 2 modules for 112 programmable buttons   282 Grandstream Budgetone 200 IP Telephone 37.9300 41.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-budgetone-200-ip-telephone-p-282.html http://www.voipon.co.uk/images/budgetone_200.jpg new Availability: In stock The Grandstream BudgeTone 200 is the most powerful phone in the BudgeTone series. It offers improved features including 3-way conferencing, full-duplex hands-free speakerphone, voicemail with indicator and custom ring tones, as well as switched or routed dual 100Mbps Ethernet ports and advanced SRTP/TLS security features. The BudgeTone 200 is an affordable, easy to use VoIP solution for the home or office. Grandstream IP phones are based on industry open standards. Built upon innovative technology, Grandstream IP phones feature market leading superb sound quality and rich functionalities at ultra-affordable prices. Features Product Description: Grandstream BudgeTone 200 - IP phone Product Type: IP phone Main Features: Integrated Ethernet switch VoIP Protocols: SIP Voice Codecs: G.711, G.722, G.723, G.729 Intercom: Yes Speakerphone: Yes ( digital duplex ) Caller ID: Yes Display: LCD display - monochrome Network Ports: Qty 2 x Ethernet 10/100Base-TX Body Colour: Matte black   Specification LAN Interface 2x RJ45 10/100Mbps Headset Jack 2.5mm Headset Port LED 1 LED (red color) Phone Case 25-Button Keypad and 12-Digit Call ID LCD Universal Switching Power Adapter Input: 100-240VAC 50-60 Hz, Output: +5VDC, 1200mA, UL certified Dimensions 18cm (W) x 22cm (D) x 6.5cm (H) Compliance FCC / CE / C-Tick 284 Polycom Soundpoint IP430 92.0000 97.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip430-p-284.html http://www.voipon.co.uk/images/polycom_ip430_sml.jpg new Availability: In stock Polycom SoundPoint IP 430 - Full-Duplex Speakerphone & Robust Feature Set in a 2-Line Phone The Polycom SoundPoint IP 430 two-line desktop IP phone is designed to meet the telephony needs of general business users such as cubicle workers with low-to-medium telephone call volumes. With simple menu navigation, easy-to-read graphical LCD display and convenient, one-button access to core telephony features, the SoundPoint IP 430 voIP phone provides an easy transition from legacy PSTN systems to the world of IP telephony. The SoundPoint IP 430 desktop IP phone's full-duplex speakerphone, utilising Polycom Acoustic Clarity Technology, provides excellent voice quality and enables two-way interactive conversations that are as natural as talking in person. As an enterprise grade IP phone, the SoundPoint IP 430 delivers a robust feature set that encompasses traditional telephony features such as call hold, pick-up, park and transfer, as well as more advanced capabilities such as shared call / bridged line appearance, multiple call appearances and integration with Microsoft LCS 2005 for telephony and presence. The phone's Quality of Service capabilities incorporating Layer 3 TOS, 802.1 p/Q VLAN tagging, and DSCP2 help to ensure that a superb communications experience is delivered in a network environment. With integrated IEEE 802.3af Power over Ethernet circuitry and dual-port 10/100 Mbps Ethernet switch for PC and LAN connection, the Polycom SoundPoint IP 430 desktop phone offers a choice of powering and cabling options that can help reduce cord clutter and cabling cost. A highly secure IP telephony terminal, the phone supports HTTPS3 and TLS2 security, "signed" software executables as well as encrypted configuration files. Make Great Things Happen with Polycom SoundPoint IP 430 voIP Phone In today's Internet driven world, being able to conduct real time communication and collaboration has become critical to an organisation's survival. As the market leader in voice, video, data and Web solutions, Polycom makes it easy for people to interact and maximise productivity over any network, in just about any environment, anywhere around the globe. That's why more organisations worldwide use and prefer Polycom call and conferencing solutions. When people work together, great things happen. See how you too, can achieve great things with Polycom SoundPoint IP 430 IP phone. Polycom Soundpoint IP 430 Features & Benefits Display 132 x 46 pixel graphical LCD 2 line keys with bi-color LEDs Message waiting indicator Keys 9 dedicated keys - 2 volume control keys - 3 feature keys - Mute key- Headset key - Hands-free speakerphone key - Hold key 4 context-sensitive soft keys 5 display control keys Power Integrated IEEE 802.3af Power over Ethernet support Power consumption: 3.0W nominal (3.8W max) Note: Power consumption indicated may be exceeded under start-up conditions. Customers should always use the original power source supplied with the phone External universal AC adapter (included, 24V DC) Headset Capability Amplified headsets with an RJ-9 jack are recommended Hearing Aid Compatible (HAC) in accordance with Section 508 Standards for Electronic and Information Technology, Telecommunication Products (1194.23) Audio Full-duplex speakerphone with Polycom Acoustic Clarity Technology Headset, handset, and hands-free modes Local 3-way conferencing Individual volume settings with visual feedback for each audio path Adaptive jitter buffers and packet loss concealment algorithms Acoustic echo cancellation Customizable audio sound effects Support for G.711 µ/A and G.729A (Annex B) codecs Voice activity detection and comfort noise fill Telephony Tone Signaling (RFC2833) Record and playback for diagnostic purposes Network and Provisioning Dual 10/100 Mbps switched Ethernet ports Manual or dynamic host configuration protocol (DHCP) setup Central provisioning for mass deployments from an FTP, TFTP,HTTP, or HTTPS3server. Provisioning server redundancy supported Time and date synchronization using SNTP QoS support - IEEE 802.1 p/Q tagging (VLAN), Layer 3 TOS, and DSCP2 Network Address Translation (NAT) support RTCP support (RFC1889) User-selectable hardware diagnostics(Network/CPU/Memory monitoring) Event logging Protocol Support IETF Session Initiation Protocol (SIP) - RFC3261 and companion RFCs Security Transport Layer Security (TLS) Encrypted configuration files Digest authentication Support for URL syntax with password for boot server HTTPS secure provisioning Support for signed software executables Password login Summary of Features Up to 2 dedicated lines Local feature-rich GUI Shared call / bridged line appearance Multiple call appearances Flexible line appearances Call transfer, hold, divert (forward) One-touch speed dial, redial Local 3-way conferencing Called, calling, connected party identification / information User configurable contact directory and call history (Missed,placed and received) Integration with Microsoft LCS 2005 for telephony and presence2-Compatibility with Microsoft Office Communicator andWindows?Messenger 5.1 Client Distinctive incoming call treatment / call waiting Do Not Disturb function Automatic on-hook call placement ("hot-dial") Call timer Multilingual user interface supporting Danish, Dutch, English(Canadian / UK / US), French, German, Italian, Norwegian,Portuguese, Russian, Spanish, and Swedish languages Country-specific call progress tones Wave file support for call progress tones Hardware diagnostics Status and statistics query SoundPoint IP 430.. In the Box: SoundPoint IP 430 console Handset and handset cord Quick Start Guide Reversible desk-mount / wall-mount stand Network cable Universal power adapter (including country-specific cord kit) Approvals FCC Part 15 (CFR 47) Class B ICES-003 Class B CISPR22 Class B EN55022 Class B AS/NZS 3548 Class B VCCI Class B EN55024 EN61000-3-2; EN61000-3-3, EN61000-6-1 ROHS compliant Safety UL 60950 CE Mark CAN/CSA C22.2 No. 60950 EN 60950 EN61000-4-2 EN61000-4-4 EN61000-4-11 AS/NZS 3260 IEC 60950 Operating Conditions Temperature: 10 to 40 degrees C (50 to 104 degrees F) Relative Humidity: 20% to 85%, non-condensing Storage Temperature -40 to +70 degrees C (-40 to +160 degrees F) Size 8.25 in x 6.5 in x 7.0 in x 1.25 in (21 cm x 16.5 cm x 18 cm x 3 cm) (W x H x D x T) Weight Shipping: 4.0 lb (1.9 kg) Part Numbers 2200-12430-015 (APAC, UK) 2200-12430-001 (North America, Taiwan) 2200-12430-002 (Japan) 2200-12430-012 (Australia/New Zealand) 2200-12430-122 (Rest of Europe) 285 Linksys SPA-901 IP Telephone 33.5000 34.90 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa901-ip-telephone-p-285.html http://www.voipon.co.uk/images/linksys_spa901.jpg new Availability: Please call for latest availability Comprehensive Interoperability and SIP Based Feature Set Based on the SIP Standard, the SPA901 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enablind service providers to quickly rollout competitive, feature-rich services to their customers. Linksys SPA-901 Entry Level SIP VOIP Phone With hudreds of features and configurable service parameters, the SPA-901 addresses the requirements of traditional business users while leveraging the advantages of IP Telephony. Features such as easy station moves and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA-901. Carrier-Grade Security, Provisioning and Management The SPA-901 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, pre-loading and re-configuring customer premise equipment (CPE). Telephony Features One service provider line Two call appearances accessed via Flash Key or Hook Flash Shared line appearance** Line status indicator Call Hold Music on Hold** Call Waiting Outbound CallerID Blocking Call transfer - Atended and Blind Three Way conferencing with local mixing Multi-Party Call Conferencing via external Conference Bridge** Call Pick Up - Selective and Group** Call park and UnPark** Call back on Busy Call Blocking - Anonymous and Selective Call Forwarding - Unconditional, No Answer, On Busy Call Return - Redial Last Caller Hot Line and Warm Line Automatic Calling Call Logs (60 Entries Each) Made, Answered and Missed Calls. Accessed via HTTP Server Redial Last Called Number Do Not Disturb (Caller Hears Busy Line tone) Block Anonymous Incoming Calls URI (IP) Dialing support (Vanity NUmbers) Built-in Web Server for Administration and Configuration, with User and Admin Access Levels Built-In Interactive Voice Response (IVR) System to check status and change configuration Date and Time w/Intelligent Daylight Savings Support Call Start Time stored in Call Logs Distinctive Ringing 10 User-Downloadable Ring Tones - Ring Tone Generator free from www.Linksys.com Speed Dial (8 entries) Group Paging (Outbound Only)** Intercom (Outbound Only)** Set preferred CODEC, Per Call, All Calls Configurable Dial/Numbering Plan Support Ringer and Handset Voluem Controls DNS SRV and Multiple A Records for Proxy Lookup and Proxy redundancy Syslog, Debug, Report Generation, an Event Logging Secure Call Encrypted Voice Communication Support NAT Traversal Automated Provisioning, Multiple Methods. Up to 256Bit encryption: (HTTP, HTTPS, TFTP) Support Linksys Voice System Automatic Configuration Optionally require Admin Password to Reset unit to factory defaults **Feature requires support by call server. Hardware Voice Mail Message Waiting Indicator Light Redial Button Dedicated Flash Button Volume Control button cycles through Voluem Levels. Controls Ringer and Handset Volume. Standard 12-Button dialing pad High Quality Handset and Cradle Ethernet LAN - 10Base-T RJ-45 5v DC Universal (100-240v) Switching Power Adapter Specifications: Data Networking MAC Address (IEEE 802.3) IPv4 ARP DNS DHCP Client ICMP TCP UDP RTP RTCP DiffServ VLAN Tagging SNTP Voice Gateway SIPv2 SIP Proxy redundancy Re-Registration with Primary SIP Proxy Server SIP Support in NAT Networks (including STUN) SIPFrag Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP CODEC Name Assignment G.711 G.726 G.729 G.723.1 Dynamic Payload Support Adjustable Audio Frames Per Packet DTMF: In-Band and Out-of-Band Flexible Dial Plan Support with Inter-Digit Timers I P Address / URI Dialing Support Call Progress Tone Generation Adaptive Jitter Buffer Frame Los Concealment VAD Attenuation / Gain Adjustments MWI and VMWI Third Party Call Control Security Password Protected System, preset to factory default Password Protected access to Administrator and User Level Features HTTPS with Factory Installed Client Certificate HTTP Digest - encrypted authentication via MD5 (RFC 1321) Up to 256-Bit AS Encryption Linksys SPA Series VoIP Telephone (SIP) Comparison Table SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernet Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 286 Linksys SPA-921 IP Telephone 50.8000 53.40 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa921-ip-telephone-p-286.html http://www.voipon.co.uk/images/linksys_spa921.jpg new Availability: In stock Stylish and functional in design, the SPA-921 IP Phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA-921 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support. Standard features on the SPA921 include a high resolution graphical display, speakerphone, and a 2.5 mm head-set port. The SPA921 supports one line with two call appearances and provides support for three way conferencing, attended call transfer, and placing a call on hold to answer an incoming call. The line can be configured as a unique phone number (or extension), or can be configured to share a number that is assigned to multiple phones. Features: Full featured one-line business class IP phone Connect directly to an Internet Telephone Service Provider or connect to an IP PBX Speakerphone. Caller ID. Call Hold, Transfer, Conferencing,and more Easy installation with secure remote provisioning. Menu based and web based configuration. Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the SPA-921 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure leaders enabling service providers to quickly roll-out competitive, feature rich services to their customers. With undreds of features and configurable sevice parameters, the SPA-921 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence,and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA921. Carrier-Grade Security, Provisioning, and Management The SPA921 uses standard encryption protocols to provide secure remote in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, preloading, and re-configuring customer premise equipment (CPE). Package Contents SPA-921 IP Phone, Handset, and Stand Handset Cord - 56 cm (26 in) 5v Power Adapter - 1.8 m (6 ft) Cord RJ45 Ethernet Cable - 1.8 m (6 ft) Cord Quick Installation Guide Telephony Features One Voice Line with Two Call Appearances Backlit Pixel Based Display: 128x64 Monochrome Graphical Liquid Crystal Display (LCD) Line Status - Active Line Indication, Name and Number Menu Driven User Interface Shared Line Appearance ** Speakerphone Call Hold Music on Hold ** Call Waiting Caller ID Name and Number and Outbound Caller ID Blocking Outbound Caller ID Blocking Call Transfer - Attended and Blind Three Way Call Conferencing with Local Mixing Connects to External Conference Bridge for Multi-party Conferencing Automatic Redial of Last Calling and Last Called Numbers On-Hook Dialing Call Pick Up - Selective and Group ** Call Park and UnPark ** Call Swap Call Back on Busy Call Blocking - Anonymous and Selective Call Forwarding - Unconditional, No Answer, On Busy Hot Line and Warm Line Automatic Calling Call Logs (60 entries each): Made, Answered, and Missed Calls Redial from Call Logs Personal Directory with Auto-dial (100 entries) Do Not Disturb (callers hear line busy tone) Digits Dialed with Number Auto-Completion Anonymous Caller Blocking URI (IP) Dialing Support (Vanity Numbers) On Hook Default Audio Configuration (Speakerphone and Headset) Multiple Ring Tones with Selectable Ring Tone per Line Called Number with Directory Name Matching Call Number using Name - Directory Matching or via Caller ID Subsequent Incoming Calls with Calling Name and Number Date and Time with Intelligent Daylight Savings Support Call Duration and Start Time Stored in Call Logs Call Timer Name and Identity (Text) Displayed at Start Up Distinctive Ringing Based on Calling and Called Number Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com Speed Dialing, Eight Entries Configurable Dial/Numbering Plan Support Intercom ** Group Paging ** NAT Traversal, including STUN Support DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy Syslog, Debug, Report Generation, and Event Logging Secure Call Encrypted Voice Communication Support Built-in Web Server for Administration and Configuration with Multiple Security Levels Automated Remote Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP) Optionally Require Admin Password to Reset Unit to Factory Defaults ** Feature requires support by call server Hardware Features Pixel Based Display: 128x64 Monochrome LCD Graphical Display Dedicated Illuminated Buttons for: Audio Mute On/Off Headset On/Off Speakerphone On/Off Four Soft Key Buttons Four Way Rocking Directional Knob for Menu Navigation Voice Mail Message Waiting Indicator Light Voice Mail Message Retrieval Button Dedicated Hold Button Settings Button for Access to Feature, Set-up, and Configuration Menus Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer Standard 12-Button Dialing Pad High Quality Handset and Cradle Built-In High Quality Microphone and Speaker Headset Jack - 2.5 millimeter Ethernet LAN - 10BaseT RJ-45 5 volt DC Universal (100-240 Volt) Switching Power Adaptor LED Test Function Data Networking MAC Address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) ARP - Address Resolution Protocol DNS - A Record (RFC 1706), SRV Record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) ICMP - Internet Control Message Protocol (RFC792) TCP - Transmission Control Protocol (RFC793) UDP - User Datagram Protocol (RFC768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349) VLAN Tagging 802.1p/q - Layer 2 QoS SNTP - Simple Network Time Protocol (RFC 2030) Voice Gateway SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264) SIP Proxy Redundancy - Dynamic via DNS SRV, A Records Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks - NAT (including STUN) SIPFrag (RFC 3420) Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP Codec Name Assignment Voice Algorithms: - G.711 (A-law and &#956;-law) - G.726 (16/24/32/40 kbps) vo - G.729 A - G.723.1 (6.3 kbps, 5.3 kbps) Dynamic Payload Support Adjustable Audio Frames Per Packet DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO) Flexible Dial Plan Support with Inter-Digit Timers IP Address / URI Dialing Support Call Progress Tone Generation Jitter Buffer - Adaptive Frame Loss Concealment VAD - Voice Activity Detection with Silence Suppression Attenuation / Gain Adjustments MWI - Message Waiting Indicator Tones VMWI - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE Caller ID Support (Name and Number) Third Party Call Control (RFC 3725) Provisioning, Administration & Maintenance Integrated Web Server Provides Web Based Administration and Configuration Telephone Key Pad Configuration via Display Menu / Navigation Automated Provisioning and Upgrade via HTTPS, HTTP, TFTP Asynchronous Notification of Upgrade Availability via NOTIFY Non-intrusive, In-Service Upgrades Report Generation and Event Logging Statistics Transmitted in BYE Message Syslog and Debug Server Records - Configurable Per Line Physical Interfaces 1 10baseT RJ-45 Ethernet Port (IEEE 802.3) Handset: RJ-7 Connector Built-in Speakerphone and Microphone Headset 2.5 mm Port Power Supply Switching Type (100-240v) Automatic DC Input Voltage: +5 Volts DC at 2.0 Amps Maximum Power Consumption: 5 Watts Power Adapter: 100-240v - 50-60Hz (26-34VA) AC Input, 1.8m (6 ft) cord Indicator Lights/LED Four (4) Call Appearance/Line Buttons with Associated Tricolor LED Line LED State Indication: Active, Idle, On Hold, Unregistered Speakerphone On/Off Button with LED Headset On/Off Button with LED Mute Button with LED Message Waiting Indicator LED Voicemail Message Retrieval Button Hold Button LED Test Function Linksys SPA Series VoIP Telephone (SIP) Comparison Table SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernet Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 287 AVM Fritz!Box Fon 75.0000 78.95 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-p-287.html http://www.voipon.co.uk/images/avm_fritzbox_fon.jpg new Availability: Discontinued Features at a glance VoIP PBX for telephony and web surfing over ADSL Use both conventional phone lines and Internet telephony simultaneously, with analog telephones Powerful ADSL modem Built-in ADSL router to connect one or more computers to the the Internet Differential quality of service (QoS) for optimum voice and data communication Added Internet security with built-in firewall Contains the powerful ADSL software package FRITZ!DSL Connect more computers using a LAN switch or hub PBX for Internet telephony and conventional lines Two fully configurable extensions Use PBX features for Internet phone calls too Call routing for customized use of Internet and conventional telephony One phone book for both Internet telephony addresses and conventional phone numbers Speed dialing, blacklist, and more Versatile connectivity options Two extensions for analog phones, faxes and answering machines ADSL, ISDN and analog phone lines PC or notebook with a USB port, using the USB cable supplied (Windows, Linux, or Mac) Computer with an Ethernet port (10BASE-T/100BASE-TX) using the LAN cable supplied Network hub or switch (to connect additional computers) using the LAN cable supplied Game console (Xbox, PS2) using the LAN cable supplied SIP standard Internet telephony Conforms to SIP (RFC 3261) Manages multiple SIP accounts (Internet phone addresses) Optimum performance Bandwidth management for optimum quality of service (QoS) IP traffic shaping permits simultaneous uploads and downloads at full ADSL speed--for power websurfing during eDonkey file sharing, for example for the Fritz!Box Fon Ensures intelligent distribution of the ADSL bandwidth among all applications Easy to use for all common ADSL bandwidth configurations Internet security with FRITZ!DSL and in ADSL router operation IP masquerading/Network Address Translation Packet firewall with stateful inspection Secure port forwarding for local servers System requirements ADSL line: Standard: ITU G.992.1 Annex A and Annex B LAN port (standard Ethernet, 10BASE-T/100BASE-TX):PC, notebook, Mac, game console, or other Ethernet device Switch or hub to connect more PCs or networks USB port:PC or notebook with a USB interface (USB version 1.1 or 2.0) Operating system: Windows XP, 2000, Me or 98; Linux (SuSE 9.0 or later); Mac OS X (10.3.3 or later) Package contents VoIP PBX system with integrated ADSL router for multiple computers 4 m ADSL cable to connect to splitter; RJ45 plug4 m combined ISDN/analog telephone cable 1.5 m LAN cable 1.5 m USB cable RJ45-RJ11 adapter to connect to analog phone line RJ45-RJ11 adapter to connect to DSL line External power supply CD-ROM with installation and application software, and user manual Printed installation manual Service Warranty: 5 years Free AVM support by phone and e-mail Free software updates over the Internet 288 AVM FritzBox Fon WLAN 95.0000 95.00 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-wlan-p-288.html http://www.voipon.co.uk/images/avm_fritzbox_fon_wlan.jpg new Availability: Discontinued The Fritz!Box Fon WLAN makes IP Telephony as simple as it should be. You won't need to turn your computer on to make and receive calls, and you have the freedom that a wireless network brings. You can access the internet through the Fritz!Box via Ethernet, USB or wireless signals. The Fritz!Box Fon WLAN includes an integrated ADSL modem, a full-featured router, 2 phone ports for VoIP and a fully functional PSTN port (analogue/ISDN). Integrated within the Fritz!Box WLAN is a powerful firewall and an elegant VoIP system with a simple interface. Simply place the WLAN away from large metal or signal-emitting devices and with minimal configuration you are on your way. FRITZ!Box Fon WLAN at a Glance: FRITZ!Box Fon WLAN combines an ADSL router and a PBX Economical phone calls over the Internet with VoIP, or over the ISDN or analog phone line Internet telephony with existing phones, without turning on the PC Connect analog terminal equipment, ISDN telephones and PBXs Intelligent bandwidth management for perfect voice quality Router with integrated ADSL modem; connects to multiple PCs WLAN encrypted with factory-set unique key Integrated firewall protects connected PCs FRITZ!Box Fon WLAN connections: 2 connectors for analogue telephone, fax and answering machine Connection alternatives for DSL + ISDN or DSL + analogue pstn network USB-connector for PC or Notebook (Windows/Linux/Mac) Ethernet connector (10/100 Base-T) for the computer, Games console (i.e. X-Box, Playstation) or a network hub or switch (to connect more PCs) WLAN connects with the IEEE 802.11g Standard (54 MBit/s) FRITZ!Box Fon WLAN configuration: Simple, intuitive browser menu Analogue ports individually configurable Specialised configurations possible with Internet and fixed-line telephony Speed Dial, Connection journal, Call statistics, Fixed-line fallback WLAN- Encoded with individual passwords activated already by the factory WLAN-Useable with attached phones or with wireless Comfortable online accounting for controlling the DSL call budget Expert Mode available for specialised application functions FRITZ!Box Fon WLAN Details: Telephone system for the Internet and fixed-line telephones (ISDN and anlogue connection) WLAN-Router with DHCP-Server, IP-Masquerading/Network Address Translation Integrates DSL-Modem with up to 8 MBit/s Data transmission rate (already prepared for ADSL 2+) Internet telephony SIP conforming to RFC 3261 Intelligent Codec Management (Codecs: G.711, G.726-32, G.726-40, G.726-24) WLAN-Security encryption with WPA-II, WPA, WEP-64 or WEP-128 Stateful Packet Inspection Firewall, secure port release Bandwidth management & "Traffic Shaping" Device measures 85x140x35 mm, Tabletop or wall mounting possible (3.3 x 5.5 x 1.4 inches) FRITZ!Box Fon WLAN Contents: VoIP PBX system for telephony and wireless ADSL web surfing with WLAN Integrated ADSL modem with router capability to connect multiple PCs 4 m ADSL cable to connect to splitter; RJ45 plug 4 m cable for ISDN or analog phone line 1.5 m LAN cable 1.5 m USB cable RJ45-RJ11 adapter to connect to analog phone line RJ45-RJ11 adapter to connect to DSL line External power supply CD-ROM with installation and application software, and user manual Printed installation manual 289 AVM FritzBox Fon WLAN 7050 105.0000 108.00 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7050-p-289.html http://www.voipon.co.uk/images/avm_fritzbox_fon_7050.jpg new Availability: The 7050 has been replaced by the AVM 7140 . The FRITZ!Box Fon 7050 brings your PCs to the Internet with lightning-fast DSL-speeds - over Ethernet, USB or wireless via WLAN. Simple to operate and highly secure from the start. The wireless portion 125MBit/s-WLAN-interface from the web with individual password protection, thus wireless communications are always protected through a Firewall, starting as soon as it is activated. The Fritz!Box Fon WLAN 7050 includes an integrated ADSL modem, a full-featured router, 2 phone ports for VoIP and a fully functional PSTN port (analogue/ISDN). With the integrated ADSL/ADSL2+-Modem the FRITZ!Box Fon 7050 can connect to both DSL connections and to cable - or the modem can be also switched off to operate as a VoIP router. With the FRITZ!Box Fon 7050 calling over the Internet is finally very easy, the way it should be: attach your existing phone, lift the handset, call! You can call with Voice over IP through the Internet without switching on your PC. Also you can use your existing phone with the Fritz!box: calls are passed to and from the integrated telephones ports. Additional functions and features are easy to learn and use due to simple and easy instructions and automated installation software. Additionally, this model FRITZ!Box allows configurations between ISDN, pstn and VoIP lines and uses the high-security WPA2 encryption (under the 802.11i standard) and can allow you to create a network that is "unbreakable by any known technique" - AVM. Main Features and Functions include: DSL-Router and Telephone device Calling over VoIP, ISDN or analogue landlines Port for 3 analogue-devices Integrated Modem, supports DSL and ADSL2+ Intelligent Broadband management for perfect call quality WLAN Access Point with IEEE 802.11b and g WLAN encryption using WEP-64, WEP-128 or WPA (WPA-PSK with TKIP) Bring more PCs from your network online USB connection Technical Features VoIP-Telephone device for calling and surfing over DSL Connection to DSL 2x LAN-ports (Standard-Ethernet, 10/100 Base-T) USB-port WLAN Access Point S0-NT-port Conformance with SIP RFC 3261 Register up to 10 Internet-phone numbers (SIP-Accounts) Up to 3 ports for analogue telephones, Fax and answering machines The FRITZ!Box Fon 7050 supports ISDN in two ways: Both ISDN telephone connections and ISDN terminals (telephones, telephone systems) can be attached. So a existing telephone system can be extended with little problem. This function is currently exclusive to AVM products. FRITZ!Box Fon WLAN 7050 Contents: VoIP PBX system for telephony and wireless ADSL web surfing with WLAN Integrated ADSL modem with router capability to connect multiple PCs 4 m ADSL cable to connect to splitter; RJ45 plug 4 m cable for ISDN or analog phone line 1.5 m LAN cable 1.5 m USB cable RJ45-RJ11 adapter to connect to analog phone line RJ45-RJ11 adapter to connect to DSL line External power supply CD-ROM with installation, application software, and user manual Printed installation manual 290 AVM FritzBox Fon ATA 64.0000 71.95 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-ata-p-290.html http://www.voipon.co.uk/images/avm_fritzbox_fon_ata.jpg new Availability: Discontinued FRITZ!Box Fon ata makes IP Telephony as easy as you'd like it to be: connect your existing telephones; pick up the receiver -- that's all! Now you can talk over the Internet with Voice over IP (and you don't need to turn on your PC), or over the conventional phone line. Simply connect FRITZ!Box Fon ata to your existing router or ADSL modem. FRITZ!Box Fon ata connects your PCs to the Internet through an Ethernet or USB port - with integrated firewall protection. The Fritz!Box Fon ata includes a full-featured router, 2 phone ports for VoIP and a fully functional PSTN port (analogue/ISDN). FRITZ!Box Fon ata at a Glance: FRITZ!Box Fon ata combines a Broadband Router and a PBX Economical phone calls over the Internet with VoIP, or over the ISDN or analog phone line Internet telephony with existing phones, without turning on the PC Connect analog phones, answering machines, and fax machines Intelligent bandwidth management for perfect voice quality PC-lines over Ethernet or USB Integrated, TÜV-tested firewall protects connected PCs FRITZ!Box Fon ata Connections: Two extensions for analog phones, faxes and answering machines Conventional phone line, either ISDN or POTS USB port for PC or notebook (Windows, Linux, or Mac) Ethernet port (10/100 Mbit/s) for computers, game consoles (such as Xbox or PlayStation), or a LAN hub or switch to connect additional PCs FRITZ!Box Fon ata Operation: Easy router configuration through standard browsers Extensions configurable individually Call routing for customized use of Internet and landline telephony Extensions of an existing PBX can also use Internet telephony Speed dialing, dial-out blacklist, call-through function, caller list Convenient on-line meter to monitor the ADSL budget Switch to Expert Mode to configure custom application scenarios FRITZ!Box Fon ata Details: PBX for Internet and landline telephony (ISDN and POTS lines) Internet telephony in conformance with SIP (RFC 3261) Intelligent codec management ADSL router with DHCP server and IP masquerading/Network Address Translation Firewall with stateful packet inspection and secure port forwarding for local servers Bandwidth management: ADSL traffic shaping for optimum use of the ADSL line Dimensions: 160 × 122 × 31 mm; tabletop or wall-mounted installation FRITZ!Box Fon ata Contents: VoIP PBX for telephony and web surfing over ADSL 4 m long cable for ISDN and analog lines 1.5 m LAN cable 1.5 m USB cable RJ45-RJ11 adapter to connect to analog phone line External power supply CD-ROM with installation and application software, and user manual Printed installation manual 291 AVM Fritz WLAN USB Stick 26.0000 28.95 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritz-wlan-usb-stick-p-291.html http://www.voipon.co.uk/images/avm_fritz_wlan_usb_stick.jpg new Availability: In Stock Now immediately available throughout Europe is the new FRITZ!WLAN USB Stick from AVM. The outstanding features of FRITZ! WLAN USB Stick are its maximum security, very fast transmission speeds and ease of installation - with at the same time an extremely compact design. Unbreakable encryption and turbo mode The FRITZ!WLAN USB Stick connects your computer to the FRITZ!Box Fon WLAN or any other WLAN router that supports the current IEEE 802.11b/g standards. With FRITZ!WLAN USB Stick the WLAN is particularly secure since it offers the WPA2 encryption standard (802.11i) which is considered unbreakable. However, for remote terminals that are unable to support this standard, it is possible to select WPA or WEP encryption. FRITZ!WLAN USB Stick supports the new 802.11g++ WLAN mode (125 Mbit/s) and thus permits up to 35% net faster data transmission in comparison to 802.11g. Convenient self-installation With the Windows XP operating system (Service Pack 2) the installation is even simpler and more convenient: if the FRITZ!WLAN USB Stick is connected to the PC, it self-installs entirely without the driver CD. Simple operation with the WLAN software The software of the FRITZ!WLAN USB Stick allows fast and uncomplicated setting up of connections, including to WLAN access points of other manufacturers. The control software also allows individual settings to be made. For example, one can flexibly adjust the transmitting power of the USB stick, for in many cases the maximum transmitting power is not necessary. This significantly reduces the power consumption, which is an advantage particularly with mobile operation. Minimal size - maximum service Weighing just 10 grams and with a length of only 53 millimetres, FRITZ!WLAN USB Stick is very small and compact. The new AVM product is on the market right now. FRITZ!WLAN USB Stick comes with software and drivers for Microsoft Windows XP Professional x64 Edition, XP, 2000, Me and 98 SE. AVM provides its usual five-year warranty. Included in AVM's service offering are comprehensive support via email as well as free updates 293 Digium Wildcard TE407P PCI ISDN PRI Card with Echo Cancellation 1167.1000 1423.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te407p-pci-isdn-pri-card-with-echo-cancellation-p-293.html http://www.voipon.co.uk/images/digium_wildcard_te407p.jpg new Availibility: In stock Digium TE407P VoIP Card with Echo Cancellation The Digium TE407P card incorporates an on-board DSP-based echo cancelation module, supports E1, T1, and J1 environments and is selectable on a per-card/per-port basis. The TE407P merges the leading TE405P card and VPM450M Octasic DSP-based echo cancellation module. The VPM450M provides a certified carrier-grade algorithm that has been labeled an industry benchmark for echo cancelation. This new module improves upon the older VPM400M and TE411P/TE406P products. Previously, with the VPM400M, 16ms or 128taps of echo cancellation were possible across 128 channels. With the new VPM450M, users can eliminate echo tails up to 128ms or 1024 taps across all 128 channels in E1 mode or 96 channels in T1/J1 modes. In addition, this module utilises the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. The Digium TE407P has been designed for full compatibility with existing software applications and is fully integrated with the open source Asterisk PBX/IVR platform. Furthermore, the open source driver supports an API interface for custom application development. The TE407P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE407P is for use only with a 5.0 volt PCI slot. Certifications for the TE407P are pending. 294 Digium Wildcard TE212P PCI ISDN PRI Card with Echo Cancellation 645.0000 679.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te212p-pci-isdn-pri-card-with-echo-cancellation-p-294.html http://www.voipon.co.uk/images/digium_wildcard_te212p.jpg new Availibility: In stock The Digium TE212P incorporates an onboard DSP-based echo cancelation module, supports E1, T1, and J1 environments and is selectable on a per-port or per-card basis. The TE212P is a bundling of our the TE210P and the VPM450M Octasic DSP-based echo cancelation module. The TE212P is Digium's and the industry's first two-port digital card with hardware-based echo cancelation. The module enables users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes. In addition, this module makes use of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. The Digium TE212P is designed to be fully compatible with existing software applications and is fully integrated with Asterisk. The open source driver supports an API interface for custom application development. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications. The TE212P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE212P supports a 3.3v PCI slot only - typically available on newer motherboards and in 64-bit PCI bus architectures. Certifications for the TE212P are pending. 295 Digium Wildcard TE207P PCI ISDN PRI Card with Echo Cancellation 645.0000 679.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te207p-pci-isdn-pri-card-with-echo-cancellation-p-295.html http://www.voipon.co.uk/images/digium_wildcard_te207p.jpg new Availibility: In stock The Digium TE207P - with Octasic Echo Cancellation The Digium TE207P offers an on-board DSP-based echo cancelation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis. The TE207P is a bundling of Digium's leading TE205P and VPM450M Octasic DSP-based echo cancelation module. The TE207P is an industry first - a two-port digital card featuring hardware-based echo cancelation. The new module allows users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes. In addition, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. The TE207P has been designed to be fully compatible with existing software applications and is fully integrated with the open-source Asterisk PBX/IVR platform. Furthermore, the open source driver supports an API interface for custom application development. The TE207P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. The TE207P is for use only with a 5.0 volt PCI slot. Certifications for the TE207P are pending. 296 Digium Wildcard TE412P PCI ISDN PRI Card with Echo Cancellation 1034.0000 1149.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te412p-pci-isdn-pri-card-with-echo-cancellation-p-296.html http://www.voipon.co.uk/images/digium_wildcard_te412p.jpg new Availibility: In Stock Quad T1/E1 Digium TE412P SIP/IAX/Asterisk Card with Echo Cancellation The TE412P incorporates an on-board DSP-based echo cancelation module. Supports E1, T1, and J1 environments and is selectable on a per-card/per-port basis.   The TE412P combines the TE410P card and the VPM450M Octasic DSP-based echo cancellation module. The VPM450M provides a certified carrier-grade algorithm noted as a benchmark for echo cancelation. The VPM450M improves upon the older VPM400M and TE411P/TE406P products. Previously, with the VPM400M, 16ms or 128taps of echo cancellation were possible across 128 channels. The VPM450M eliminates echo tails up to 128ms or 1024 taps across all 128 channels in E1 mode or 96 channels in T1/J1 modes. This module also takes advantage of the Octasic Voice Quality Enhancement to provide superior quality of sound on all calls. The TE 412P is fully integrated with the Asterisk PBX/IVR and Digium has designed the TE412P to be fully compatible with existing software applications. The open source driver also supports an API interface for custom-made applications. The TE412P adheres to industry standard telephony and data protocols, including Primary Rate ISDN (N. American and Standard Euro) protocol families for voice, Cisco, PPP, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, as well as advanced call features. The TE412P is for use only with a 3.3 volt PCI slot. Certifications for the TE412P are pending. 298 Boscom Quasar Cellunet Gateway 315.0000 389.00 Boscom Boscom Cellular Gateways http://www.voipon.co.uk/boscom-quasar-cellunet-gateway-p-298.html http://www.voipon.co.uk/images/boscom_quasar_cellunet.gif new Availability: In Stock Quasar's proven cellular technology creates gateways between the corporate PBX and cellular network to enable cost savings of up to 60% in the corporate and SOHO markets. Over 100,000 installations worldwide are proving the point: Quasar gateways get the message through at the lowest cost and with the highest reliability. Compatible with any PBX, Quasar gateways connect between corporate PBXs and all types of cellular phones, eliminating Network charges on fixed to cellular calls! VoIP GSM GatewayCellular Gateway for a SIP VoIP connection The Quasar CelluNet VoIP gateway provides a bridge between the SIP VoIP and the GSM Cellular world. It allows you to provide GSM access with IP PBX systems that support SIP or for direct connection to a SIP service provider, this is the next generation of gateway that helps the move away from legacy interfaces such as analogue or ISDN and helps to leverage the IP network. The gateway enables users of IP networks to make use of reduced mobile to mobile costs. Calls from Cellular to VoIP Calls from VoIP to Cellular SIP Support Auto Dial Follow Me Easy Configuration Dual Band GSM Compatible with Asterisk Additional advantages: Programmable via Browser Voice Response for Setting and Status (Dial in from mobile) Quick and easy installation Quick Return On Investment (ROI) Standard SIP Protocol (RFC2543, RFC3261) 299 VXI Passport 10V Headset 42.5000 46.00 VXI Corporation VoIP Accessories http://www.voipon.co.uk/vxi-passport-10v-headset-p-299.html http://www.voipon.co.uk/images/vxi_passport_10v_headser.jpg new Availability: In Stock Professional grade headset solutions with easy interchangeability The Passport 10V-DC is a high quality, professional grade monaural headset solution for call-centre use. Passport V-DC headsets work with all VXI standard amplifiers but are specifically designed for direct connect cords. Passport headsets use the same impact-resistant plastic used in sports equipment to withstand the intensity of the 24 hour call-centre. Durable construction means fewer repairs and less hassle. Features Noise cancelling microphone greatly reduces background noise for a more professional sounding call Durable components for long product life Adjustable ratchet headband for secure, comfortable fit Left or right side microphone placement Flexible microphone boom easily adjusts and stays in position for consistent voice input The Passport 10v-DC is compatible with the GXP 2000 , the Aastra 9112i , 9133i , 480i and SNOM 320 , 360 . Please select the correct option from the drop-down menu below. 300 Rhino R4T1 Quad T1/E1/PRI PCI Card (no EC) 653.0000 817.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r4t1-quad-t1e1pri-pci-card-no-ec-p-300.html http://www.voipon.co.uk/images/rhino_r4t1.gif new Availability: In Stock Linux Open Source Telephony Quad T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asteriskand Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, forlarge scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phonesand wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IPwhen you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can beconfident that Rhino will always work hard in your Open Source Telephony application. Quad T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested andready Zaptel-compliant open sourceLinux module source code Quad T1/E1 embedded CSU Line buildout software selec-table Custom Rhino PCI interfacechip means no excess CPUoverhead Fractional voice and datacapable Field software upgradable T1 crossover cables included Alarm and Link status LEDsvisible from the rear bracket foreach individual port All major signaling modes sup-ported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such asCaller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer,that is software controlled andsoftware programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiverfunctionality Long-haul and short-haul lineinterface for clock/datarecovery and waveshaping Crystal-less jitter attenuator Fully independent transmit andreceive functionality Single chip line interface unit(LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 3.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 301 Linksys SPA-3102 Analog Phone Adaptor 43.0000 47.95 Linksys Linksys Analog Adaptors http://www.voipon.co.uk/linksys-spa3102-analog-phone-adaptor-p-301.html http://www.voipon.co.uk/images/linksys_spa3102.jpg new Availability: In Stock The SPA3102 features the ability to connect standard telephones and fax machines to IP-based data networks with the additional benefit of an integrated connection for legacy telephone network "hop-on, hop-off" applications. SPA3102 users will be able to leverage their broadband phone service more than ever by automatically routing local calls from mobile phones and land lines over to VoIP service providers and vice versa. If power is lost to the unit or Internet service is down, calls can be redirected to a traditional carrier via the FXO interface. A user calling from a mobile phone or land line will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA3102 via a local telephone number. The advanced authentication and call routing intelligence programmed into the SPA3102 will route the call via the Internet to the far end destination. In addition, when using the SPA3102 at the far end, VoIP calls placed to that location can be either answered or further processed and routed on as a local call to any legacy land line or mobile phone. The SPA3102 supports one RJ-11 POTS (Plain Old Telephone Service) FXS port to connect an existing analog phone or fax machine. The SPA3102 also supports one PSTN FXO port to connect to a Telco or PBX circuit. The SPA3102 includes 2 100BaseT RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to a broadband modem or router. The SPA3102 FXS and FXO lines can be independently configured via software controlled by the service provider or the end user. Installed by the end user and remotely provisioned, configured and maintained by the service provider, each SPA3102 converts voice traffic into data packets for transmission over an IP network. Compact in design, the SPA3102 can be used in consumer and business VoIP service offerings including a full-featured IP Centrex environment. The SPA3102 uses international standards for voice and data networking for reliable voice and fax operation. Telephony Service Authentication via PIN, Digest, Caller ID (Bellcore Type 1) Per Call Authentication and Associated Routing Least Cost Routing Support Impedance Agnostics - 8 Settings Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Detection (Bellcore Type 1) Caller ID with Name/Number (Multi-national Variants) Caller ID Blocking Call Forwarding to PSTN or VoIP Service: No answer, Busy, All Do Not Disturb Call Transfer Three-way Conference Calling with Local Mixing Message Waiting Indication - Visual and Tone Based Call Return Call Back on Busy Call Blocking with Toll Restriction Delayed Disconnect Distinctive Ringing - Calling and Called Number Off-hook Warning Tone Selective/Anonymous Call Rejection Hot line and Warm Line Calling Speed Dialing of 8 Numbers/Addresses Music on Hold Fax: G.711 Pass Through or Real Time Fax over IP via T.38 Toll Quality Voice and Carrier-Grade Feature Support The SPA3102 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via our advanced implementation of standard voice coding algorithms. The SPA3102 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems. Large-Scale Deployment and Management The SPA3102 offers all the key features and capabilities which service providers can provide customized VoIP services to their subscribers. The SPA3102 can be remotely provisioned and supports dynamic, in-service software upgrades. A secure profile upload saves providers the time, expense, and hassle of managing and pre-configuring or re-configuring customer premise equipment (CPE) for deployment. Ironclad Security Linksys understands that security for end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The SPA3102 supports secure, standard encryption-based methods for communication, provisioning and servicing. Linksys SPA Series Analogue Telephone Adapter (ATA) Comparison Chart Model Service Lines Active Calls 3-Way Call Conferences Network Ports PSTN (FXO) Ports Phone (FXS) Ports SPA1001 2 2 1 1 0 1 PAP2T 2 4 2 1 0 2 SPA2102 2 4 2 2 0 2 SPA3102 2 3 1 2 1 1 SPA400 N/A N/A N/A 1 4 0 302 Mediatrix 1404 1498 1498.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1404-p-302.html http://www.voipon.co.uk/images/mediatrix_1404.jpg new Availability: In Stock Allowing enterprises to lower communications costs over any IP link, the Mediatrix® 1404 BRI VoIP gateway provides eightVoIP channels. The Mediatrix 1404 units are integrated VoIP gateways with two ISDN basic rate interface ports. It constitutes an ideal solution for LAN-based voice applications or for connecting to a service provider&rsquo;s broadband access over DSL, WLL or cable. Mediatrix 1404 ISDN BRI VoIP Gateway The Mediatrix 1404 BRI VoIP gateway provides private voice PBX networking and remote PBX extensions by connecting ISDN phones to the IP, and PBXs with each other through VoIP. Supporting eight simultaneous calls from the IP network or the PSTN, the Mediatrix 1404 allows any office to use an existing IP network for lower-cost voice communications. The Mediatrix 1404 gateway tunnels ISDN/QSIG call control messages. T.38 FoIP, fax bypass, and modem bypass capabilities ensure that the Mediatrix 1404 gateway seamlessly connects to all voice and data, and PBX networking services. With SIP, H.323 and multiple codec support, the Mediatrix 1404 offers greater flexibility and scalability for low bandwidth voice. With call routing and selectable NT/TE interfaces, the Mediatrix 140 4 gateway allows user programmable call handling based on hunt groups, caller/called ID, and time of day and local PSTN call-through without packet conversion. Example Scenarios: In enterprise networks, the Mediatrix 1404 gateway integrates ISDN telephony and IP communications for best use of bandwidth, improved office-tooffice communication, and reduced network costs. Instead of installing a separate PBX in a remote office, the Mediatrix 1404 gateway is able to provide transparent extension of PBX phones. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. The Mediatrix 1404 gateway also supports transparent ISDN & QSIG tunneling. PBXs can be networked using IP while still retaining complete functionality to private voice networking. Designed specifically for enterprise applications, the Mediatrix 1404 gateway makes use of existing broadband access equipment to connect to any standards-based VoIP network. 303 Mediatrix 1402 578 578.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1402-p-303.html http://www.voipon.co.uk/images/mediatrix_1402.jpg new Availability: In Stock Allowing enterprises to lower communications costs over any IP link, the Mediatrix® 1402 BRI VoIP gateway provides four VoIP channels. The Mediatrix 1402 units are integrated VoIP gateways with two ISDN basic rate interface ports. It constitutes an ideal solution for LAN-based voice applications or for connecting to a service provider&rsquo;s broadband access over DSL, WLL or cable. Mediatrix 1402 ISDN BRI VoIP Gateway The Mediatrix 1402 BRI VoIP gateway provides private voice PBX networking and remote PBX extensions by connecting ISDN phones to the IP, and PBXs with each other through VoIP. Supporting four simultaneous calls from the IP network or the PSTN, the Mediatrix 1402 allows any office to use an existing IP network for lower-cost voice communications. The Mediatrix 1402 gateway tunnels ISDN/QSIG call control messages. T.38 FoIP, fax bypass, and modem bypass capabilities ensure that the Mediatrix 1402 series gateway seamlessly connects to all voice and data, and PBX networking services. With SIP, H.323 and multiple codec support, the Mediatrix 1402 offers greater flexibility and scalability for low bandwidth voice. With call routing and selectable NT/TE interfaces, the Mediatrix 1402 gateway allows user programmable call handling based on hunt groups, caller/called ID, and time of day and local PSTN call-through without packet conversion. Example Scenarios: In enterprise networks, the Mediatrix 1402 gateway integrates ISDN telephony and IP communications for best use of bandwidth, improved office-tooffice communication, and reduced network costs. Instead of installing a separate PBX in a remote office, the Mediatrix 1402 gateway is able to provide transparent extension of PBX phones. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. The Mediatrix 1402 gateway also supports transparent ISDN & QSIG tunneling. PBXs can be networked using IP while still retaining complete functionality to private voice networking. Designed specifically for enterprise applications, the Mediatrix 1402 gateway makes use of existing broadband access equipment to connect to any standards-based VoIP network. 304 Tamara Female Russian Asterisk Voice Prompt 28.43 28.43 Asterisk Voice Prompts http://www.voipon.co.uk/tamara-female-russian-asterisk-voice-prompt-p-304.html http://www.voipon.co.uk/images/tamara.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly Russian, why give them the impression you are American? Why not employ Tamara to handle your calls? Tamara is a Russian, female, full Asterisk voice prompts Pack. All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. Technical Specifications The Voice prompts are recorded and processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. We can provide the packs in other audio formats if you need them. This pack includes prompts for: Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetics Listen to some samples: (in 8 bit mono. but 44.1khz) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav (in 8 bit mono. Same as the phones sound) Demo About to Try.wav Voicemail Instructions.wav Voicemail Options.wav 306 Rhino Modular Channel Bank Chassis 445 445.00 Rhino Rhino Channel Banks http://www.voipon.co.uk/rhino-modular-channel-bank-chassis-p-306.html http://www.voipon.co.uk/images/rhino_modular_chassis.jpg new Availability: In Stock Modular Chassis taking up to 6 4-port FXO or FXS modules in any combination. Providing reliable, flexible, and leading-edge solutions for a demanding industry Managing your telecommunication needs has never been easier than with Rhino products. Rhino satisfies the needs of any T1channel bank application, no matter how stringent the requirement. Unique Rhino features like real-time T1 status on our four line by 40 character (4x40) LCD display, or automatic, hands-off configuration utilizing artificial intelligence software, and the crystal clear audio quality proves that Rhino products are in a top class of their own. Knowing that the Rhino is ready to out perform means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost T1 channel banks including FXS, FXO, or mixed mode FXS/FXO analogue interfaces. As a bonus, every system comes with our standard fractional V.35 data port. Add the Rhino modular, internal power supply system to the list and Rhino crushes the competition. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your T1 application. 308 Sangoma A108D PCI PRI ISDN Card 2019.6900 2546.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a108d-pci-pri-isdn-card-p-308.html http://www.voipon.co.uk/images/sangoma_a108_pcix_small.gif new Availability: In Stock The A108D PCIx has hardware based echo cancellation across 240 channels An Advanced Flexible Telecommunications (AFT) card with eight T1/E1 or fractional T1/E1 ports, supporting multiple DS0 channels of HDLC or non HDLC data. Used to support WANPIPE® in multichannel hub configurations, and as T1/E1 voice gateways for PBX systems. Guaranteed to work with Asterisk. All Sangoma Products are protected by a 30 Day Money Back Guarantee The sandwich DSP card on the "d" model features Octasic's certified carrier-grade algorithms providing carrier grade echo cancellation and Voice Quality Enhancement (VQE) functions. Supporting 32-672 channels G.168-2002 echo cancellation, a minimum of 1024 taps for a 128ms tail/channel on all channel densities, the system also supports Octasic music protection, acoustic echo control and adaptive noise reduction. The "d" model also features on-board DTMF decoding and tone recognition. The A108 is the, eight port version of Sangoma's range of Advanced, Flexible a104, DSP, TDM voiceTelecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A108 provides full speed 132 Mbps PCI bus transfer with FPGA and DSP based processing to unload the host CPU in demanding environments such as soft PBX/IVR voice applications. Compatible with both the 3.3v and 5v PCI bus, A108 cards operate in all commercially available motherboards sharing IRQs properly with themselves and all other PCI compatible devices, so you never have to worry about hardware compatibility issues. Like all the Sangoma AFT Series, the A108 is field upgradeable to take advantage of the hardware and software improvements as they become available. Operational Modes Voice modes: The A104 and drivers fully support TDM voice gateways for the Asterisk® , YateT , OPALT PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDACT (patent pending) technology is integrated to drastically reduce the load due to software echo cancellation, which is the largest component of CPU load in a typical soft PBX system. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Data only: T1/E1 and fractional T1/E1, single channel HDLC per line. Full channelized mode to act as major network hub for sub-DS1 remotes. The A108 can support any configuration of up to 124 remote 64kbps connections carrying Frame Relay, PPP or HDLC data. Timeslots can be concatenated to support remote fractional T1/E1/J1 sites in any combination. Mixed Voice/Data mode: Robust combination of router/PBX functions in one server. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack , completely independently of TDM voice application for total system reliability. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Technical Specifications Eight port T1/E1/J1 card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features related to voice and/or data to be added when they become available. DSP card on the A108d: - G.168-2002 echo cancellation in hardware - 1024 taps/128ms tail per channel on all channel densities - DMF decoding and tone recognition - Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up). 309 Sangoma A108x PCI Express PRI ISDN Card 1219.8900 1538.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a108x-pci-express-pri-isdn-card-p-309.html http://www.voipon.co.uk/images/sangoma_a108_pci_express_small.gif new Availability: In Stock The A108 is the octal port version of Sangoma's family of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A108 provides up to 16.4Mbps of full duplex data throughput or 240 voice calls over eight T1 and or E1 lines to support high performance PCI-based routing and telephony systems. Advanced clocking features allow E1 and T1 lines to be mixed with full synchronization. As part of Sangoma's AFT range of products, the A108 makes use of the same high performance PCI interface that has been deployed in large quantities all over the world. The available A108d card includes Octasic's DSP hardware and certified algorithms providing carrier-grade echo cancellation and Voice Quality Enhancement (VQE) functions. Like all the Sangoma AFT family, the A108 is field upgradeable to take advantage of hardware and software improvements as they become available. Technical Specifications Eight T1/E1 ports with a single PCI interface for high performance voice and data applications. TDM clocking mode: Network timing can be passed from a network -connected DS0 to any or all of the other ports. Both T1 and E1 are supported simultaneously, making it possible to mix T1 channel banks and E1 networks with full channel synchronization. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Support for the Asterisk®, OPAL® Yate, FreeSwitch® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. All Sangoma's AFT products, including the A108 card use the same base PCI interface card, and the same professionally engineered firmware on the same family of Field Programmable Gate Arrays. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing without manual tuning. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers. High quality, tested RJ45 cables included. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Autosense compatibility with 5v and 3.3v PCI busses. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDAC® (patent pending) technology will be integrated to drastically reduce the cost of echo cancellation. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols such as non-byte aligned monosynch or bisynch. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack, and is completely independent of TDM voice application for total system reliability. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Optional DSP daughterboard on the A108d - G.168-2002 echo cancellation in hardware - 1024 taps/128ms tail per channel on all 256 channels - DTMF decoding and tone recognition - Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. T1/E1 Status alarms RED: Telco Red Alarm condition YEL: Receive Telco Yellow Alarm ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows, FreeBSD, Open BSD, NetBSD, Solaris Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Higher level protocols Asterisk, Yate, OPAL Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Certification FCC Part 15 Class A, FCC Part 68, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 310 Camrivox Flexor 151 Analog Telephone Adaptor 37.0000 38.95 Camrivox Camrivox Analog Adaptor http://www.voipon.co.uk/camrivox-flexor-151-analog-telephone-adaptor-p-310.html http://www.voipon.co.uk/images/flexor_151.gif new Availability: In Stock Flexor sets new standards for VoIP access devices for the home and small business markets. This state-of-the-art VoIP access device allows traditional phones to be connected to the latest Internet VoIP services, without the need for a PC. Flexor includes the features found on current VoIP devices, and adds unique innovative features combined with a desirable and elegant product design. Flexor is part of a range of VoIP products designed, developed, and marketed by Camrivox. Highlights A truly user friendly and reliable product. Reduced support and deployment burden for operators. Unified VoIP and PSTN. Quality of Service that really works. Emergency calls always possible. Remote configuration and upgrade. Operator branding available. PC integration. Highly competitive pricing. Simplicity Camrivox has invested considerable effort in ensuring that their products require the minimum of configuration, giving the end-user the easiest introduction to VoIP. The Flexor 151 is the easiest-to-configure VoIP device on the market today, demonstrated by the single page install guide. The Flexor 151 supports multiple deployment scenarios, including pre-configuration, pre-provisioned and end-user install using Camrivox CD The product supports an optional connection to the user\'s analogue telephone line. This simplifies migration to VoIP from PSTN, and also provides enhanced features for all incoming analogue telephone calls. The Flexor 151 combines the ease of a PSTN/land line phone, with the features and advantages of VoIP Quality and Reliability Other devices claim Quality of Service (QoS) features, but fail to deliver in basic ways. Camrivox recognises that reliability of service and voice quality are vital for any telephony device. Camrivox offers QoS that really works. Unique Camrivox QoS features ensure superb voice call quality is maintained, even when data traffic is contending for network bandwidth.* The Flexor 151 VoIP implementation is robust and allows interoperability with major VoIP access devices and Service Providers. PSTN failsafe features enable calls to be made if ever the VoIP service is unavailable (e.g. power failure). Emergency calls are always possible on Camrivox VoIP devices. 311 Liona Female Italian Asterisk Voice Prompt 68.32 68.32 Asterisk Voice Prompts http://www.voipon.co.uk/liona-female-italian-asterisk-voice-prompt-p-311.html http://www.voipon.co.uk/images/liona.jpg new Availability: Voice Prompts dispatched via email within 3 hours of order When a customer calls, what voice would you like them to hear? If your customers are mainly Italian, why give them the impression you are American? Why not employ Liona to handle your calls? Liona is an Italian, female, full Asterisk voice prompts Pack . All the major voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Call transfers, Call Parking, Voice Mail, Error messages, Numbers (digits), letters and Phonetics. All the recordings are conducted in our professional top of the line Protools equipped studios which are acoustically designed by professional acoustic architects for accurate audio monitoring ensuring every voice prompt sounds the same on the phone as it does in the studio. Technical Specifications The Voice prompts are recorded in 44.1 khz but get processed into the gsm audio format in8.000 kHz, Mono 1 kb/sec. If you need higher quality voice prompts we can provide them in any audio format you require. Voice Menus Call Queues Call transfers Call Parking Conferencing Voice Mail Error messages Numbers (digits) Letters Phonetic Listen to some samples: (in 8 bit mono. but 44.1khz) demo-abouttotry.wav voicemail-msginstruct.wav voicemail-options.wav (in 8 bit mono. Same as the phones sound) demo-abouttotry.wav voicemail-msginstruct.wav voicemail-options.wav 312 Digium VPM450M Hardware Echo Cancellation Module 478.4000 559.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-vpm450m-hardware-echo-cancellation-module-p-312.html http://www.voipon.co.uk/images/no_image_available.jpg new Availability: In Stock The Digium TE407P and TE412P are the next generation of Digium hardware that offer on-board Octasic echo cancellation, each compatible with a 5.0-volt and 3.3-volt slot respectively. Echo cancellation is supported for four full T1s or J1s (96 channels), or E1s (124 channels) to eliminate echo and improve voice quality in situations where software echo cancellation is not sufficient, where echo cancellation is not done at the CO, or where CPU utilization must be minimized. By supporting 128ms of echo cancellation on each and every channel, the card will perform in the most difficult environments while providing capacity/length tradeoff. The Digium TE407P/TE412P provides robust echo cancellation on the network side for situations involving a 4-to-2 wire hybrid, giving users its highest performance. Designed for Asterisk Digium has designed the TE407P/TE412P to be fully compatible with existing software applications and it is fully integrated with the open source Asterisk as well as with Asterisk Business Edition. The open source driver supports an API interface for custom application development, so developers can more accurately meet their needs. Superior Voice Quality The new Digium TE407P/412P performs high quality echo cancellation and Voice Quality Enhancements (VQE). The Digium TE407P/412P adds a series of Octasic voice processors which apply a Least Squares adaptive filter algorithm with frequency awareness providing superior performance for today's solutions. Therefore, the Digium TE407P/412P can significantly increase density in voice gateways while offloading DSPs to maximize their use for other features. Carrier-Grade Echo Cancellation When tested by leading carrier labs, the Octasic algorithm received very high marks and was labeled a benchmark-algorithm for echo cancellation. 313 AVM Fritz!Box Fon WLAN 7140 98 98.00 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7140-p-313.html http://www.voipon.co.uk/images/avm_fritzbox_fon_wlan_7140.jpg new Availability: In stock The brand-new Box by AVM is a WLAN router, a VoIP and fixed-line PBX, and an ADSL modem in one. FRITZ!Box Fon WLAN 7140 connects two analog extensions to the Internet and the fixed phone line. It permits phone calls, whether over the Internet (with Voice over IP) or over the fixed line, even while the computer is off. Encryption of WLAN connections using secure WPA standards is already activated in the factory settings. The Stick & Surf function, developed by AVM, allows connecting computers to the FRITZ!Box securely, with no need for user configuration. FRITZ!Box Fon WLAN 7140's USB host port paves the way for new applications. Memory sticks, printers or disk drives, for example, can be shared throughout the LAN. And like all of AVM's FRITZ!Box Fon products, FRITZ!Box Fon WLAN 7140 provides a hardware and software firewall as well as convenience features, such as a separate Night Service (standby), a "Do Not Disturb" function and a Push Mail message service. Furthermore, the WLAN switch on the back of the device allows the user to enable and disable its WLAN access point at the push of a button. Highlights at a Glance: WLAN router with integrated ADSL modem Ready for ADSL 2+ 4 LAN ports USB port for network printers and storage media TÜV-certified firewall security (engineering safety lab) PBX for Internet (VoIP) and fixed-line telephony (ISDN or POTS) Two analog extensions for telephones and fax machines ISDN convenience features, such as parallel calls and call forwarding, even for VoIP WLAN access point with factory-activated WPA encryption AVM Stick & Surf technology automatically configures secure wireless links WLAN button to switch WLAN function on and off Technical Features Router Functions Full-featured ADSL router with firewall and NAT, DHCP server, DynDNS client, UPnP Built-in ADSL modem Telephony Function PBX for Internet and fixed-line telephony Analog ports to connect conventional phones, answering machines and faxes Internet telephony without turning on the PC Bandwidth management: optimum quality for simultaneous telephony and web surfing over ADSL Security Built-in firewall with stateful packet inspection and port forwarding for secure Internet use TÜV-tested firewall protects connected PCs AVM Stick & Surf: automatically secure wireless web surfing Connectivity Options 4 Ethernet ports (10/100 MBit/s) for computers, game consoles (such as Xbox or PlayStation), or a LAN hub or switch to connect additional PCs Port for USB accessories, such as printers and storage media 2 extensions for analog phones, faxes and answering machines Connects to ADSL+ISDN or to ADSL+PSTN phone line WLAN Functions Wireless LAN in conformance with to 802.11b (11 MBit/s), 802.11g (54 MBit/s), and 802.11g++ (125 MBit/s) Pre-configured WLAN encryption using WPA (IEEE 802.11i) Support for WEP, WPA, and WPA2 encryption standards Additional internal WLAN antenna WLAN switch, to enable and disable WLAN access manually VoIP with the Convenience of Advanced Features In addition to excellent voice fidelity, FRITZ!Box Fon also supports features, such as parallel calls, call waiting, call forwarding, caller ID and three-party conference calls. Up to 10 individual VoIP accounts, even with different providers, can be stored. High-end Wireless LAN FeaturesFRITZ!Box is the only WLAN router to provide security out of the box with factory-activated WPA encryption. Stick & Surf technology, developed by AVM, radically simplifies getting started in wireless surfing. Stick & Surf automatically transfers configuration and security settings, such as the network key and the type of encryption, from the FRITZ!Box to the FRITZ!WLAN USB Stick, as soon as you connect the stick to the FRITZ!Box's USB host port. Furthermore, the wireless LAN can be disabled by a separate overnight configu-ration, and by a hardware switch on the back of the FRITZ!Box. 316 Linksys SPA9000 Licence Upgrade - 16 Extensions 159.0000 158.95 Linksys Linksys IP PBX http://www.voipon.co.uk/linksys-spa9000-licence-upgrade-16-extensions-p-316.html http://www.voipon.co.uk/images/linksys_small.jpg new Availability: Lead time of 2-3 days Linksys SPA9000 Licence Upgrade - 16 Extensions We will need the MAC address and serial number of the unit to be upgraded. Please supply this in the comments of your order when checking out. 323 Linksys WIP330 IP Telephone 125.0000 129.00 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-wip330-ip-telephone-p-323.html http://www.voipon.co.uk/images/linkys_wip_330.jpg new Availability: Discontinued The Linksys WIP330 Wireless-G IP Phone enables high-quality voice over IP (VoIP) service through a Wireless-G network and high-speed internet connection. Connect at home, your office, or at a public hotspot, and make low-cost calls through your Internet Telephony Service Provider. Linksys WIP330 NA WIFI SIP Phone The handset features peer-to-peer dialing, speed dial, 3-way conferencing, call waiting, call transfer, and call forwarding, mute hold and selectable ringtones. The large, full color high resolution LCD display features an intuitive user interfaceenabling users to easily and quickly configure te handset when traveling within range of other Wireless-G networks so you can make VoIP calls wherever you go. You can also surf the Internet with the built-in web browser and can even receive live video from anywhere in the world and view it right on your WIP330 Wireless-G IP Phone when you access any web camera, like the WVC54GC Wireless-G Internet Video camera, also from Linksys. Handset stores the last 20 call history records and can save 250 phone book entries. Personalize your phone with a selection of ringtones that reflect your style. Get the value of low-cost VoIP service while unleashing the full potential of Wireless-G conenctivity with the Wireless-G IP Phone from Linksys. Features: Supports SIP V2.0 Standards Compliance with IEEE 802.11b/g Wireless Standards Powered by Microsoft Windows CD, with IE Web Browser High resolution color LCD screen Supports QoS (Quality of Service) to ensure best quality voice Enhanced power saving design for extended standby and talk time 50 hours standby time, 3 hours talk time on average 3-Way conferencing, Call Hold and Resume, and CallerID Fast Hotspot Authenticaton Supports auto-provisioning using HTTP or HTTPS for configuration and upgrades Package Contents: Handset USB Power Adapter Li-Ion battery Quick Install Guide User Guide on CD-Rom Minimum Requirements: Internet Connection (Cable/DSL/Other) Wireless Router / Access Point with DHCP Server Activated VOIP Service Specifications: Model: WIP330-NA Standards: 802.11b, 802.11g Channels: 11 Channels (US, Canada), 13 Channels (Europe) Access Control: CSMA/CA with ACK Band: 2.412~2.484 GHz Transmit Power: 14 dB for 802.11b/g @ Normal Temp Range Radio Range: Outdoor up to 300m via Embedded Antenna External Interface: One mini-USB socket (for charging only), one stereo earphone jack Display: QVGA TIF 2.2 inch LCD (340x320 Pixels) with 65K colors Memory: 32MB Flash, 64MB SDRAM WIFI Account Support: Multiple Access Point Registration Support Network Protocols: TCP/UDP/IP, IPV4, DNS, SDP, ARP, ICMP, DHCP Client, Static IP Security: WEP (64/128), WPA-PSK Encryption QoS: ToS Voice Protocols: SIP V2 Session Initiation Protocol (RFC3261)m SDP (RFC2327), SIP Session Keep Alive Voice Codecs: G.711 (A-Law an U-Law), G.729A DTMF Transmission: In-band, Out-band (RFC2833) Voice Quality - G.168 echo cancellaton - Jitter Buffer Control - (default 180ms, max 900ms) - Comfortable Noise Generation - Packet Loss Concealment - Speaker and Microphone Volume Control - VAD (Voice Activity Detection) - Telephony and Supplementary Features - 3-Way Conferencing - Peer-to-Peer Dialing - Call Hold and resume - Caller ID Presentation - Caller ID Presentation Restriction - Dial by Phone Number - Call Forward - DTMF Tone Detection - Consultation Hold and Transfer - Call Waiting and Retreive - Mute - Speed Dial - Last Number Redial - Volume Control - Ringtones: True Tones - Phone Book (250 records) - Call History (20 records) - Language (English / Spanish) - Vibrator (Silent Mode) - Password Security - 10 Profiles - Date & Time (NTP Time Synchronization) - Internet Web Browser (Microsoft Windows CE with Microsoft IE) - Management Features - AES or SSL encryption - Firmware Upgrades using HTTP or HTTPS - Configuration change using HTTP or HTTPS - Embedded web configuration interface (with password protection) - Power-Up Diagnostics) Linksys SPA Series VoIP Telephone (SIP) Comparison Chart SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernt Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 324 Mediatrix 1204 rack mount kit 45.0000 49.00 Mediatrix Mediatrix Gateways http://www.voipon.co.uk/mediatrix-1204-rack-mount-kit-p-324.html http://www.voipon.co.uk/images/no_image_available.jpg new Mediatrix 1204 rack mount kit 325 Rhino Ceros Chassis, 80GB Hard Drive (CEROS-80GB-ST) 569.0000 599.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-80gb-hard-drive-ceros80gbst-p-325.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 80GB Hard Drive for Rhino PCI Cards (CEROS-80GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   328 Sangoma A104DX PCI Express PRI ISDN Card 1138.3700 1435.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a104dx-pci-express-pri-isdn-card-p-328.html http://www.voipon.co.uk/images/sangoma_a104_pci_express_small.gif new Availability: In Stock The A104DX PCI Express has hardware based echo cancellation across 120 channels The sandwich DSP card on the "d" model features Octasic's certified carrier-grade algorithms providing carrier grade echo cancellation and Voice Quality Enhancement (VQE) functions. Supporting 32-672 channels G.168-2002 echo cancellation, a minimum of 1024 taps for a 128ms tail/channel on all channel densities, the system also supports Octasic music protection, acoustic echo control and adaptive noise reduction. The "d" model also features on-board DTMF decoding and tone recognition. The A104 is the, quad port version of Sangoma's range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A104 provides full speed 132 Mbps PCI bus transfer with FPGA and DSP based processing to unload the host CPU in demanding environments such as soft PBX/IVR voice applications. Compatible with both the 3.3v and 5v PCI bus, A104 cards operate in all commercially available motherboards sharing IRQs properly with themselves and all other PCI compatible devices, so you never have to worry about hardware compatibility issues. Like all the Sangoma AFT Series, the A104 is field upgradeable to take advantage of the hardware and software improvements as they become available. Optional Modes The A104 and drivers fully support TDM voice gateways for the Asterisk® , Yate® , OPAL® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. The A104 and drivers fully support TDM voice gateways for the Asterisk® , Yate® , OPAL® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDACT (patent pending) technology is integrated to drastically reduce the load due to software echo cancellation, which is the largest component of CPU load in a typical soft PBX system. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Full channelized mode to act as major network hub for sub-DS1 remotes. The A104 can support any configuration of up to 124 remote 64kbps connections carrying Frame Relay, PPP or HDLC data. Timeslots can be concatenated to support remote fractional T1/E1/J1 sites in any combination. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Mixed Voice/Data mode: Robust combination of router/PBX functions in one server. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack , completely independently of TDM voice application for total system reliability. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Technical Specifications Quad port T1/E1/J1 card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features related to voice and/or data to be added when they become available. DSP card on the A104d: o G.168-2002 echo cancellation in hardware o 1024 taps/128ms tail per channel on all channel densities o DTMF decoding and tone recognition Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up). Higher level protocols Asterisk Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty FIVE years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 329 Sangoma A108DX PCI Express PRI ISDN Card 2067.4100 2606.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a108dx-pci-express-pri-isdn-card-p-329.html http://www.voipon.co.uk/images/sangoma_a108_pci_express_small.gif new Availability: In Stock The A108DX PCI Express has hardware based echo cancellation across 240 channels The A108 is the octal port version of Sangoma's family of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A108 provides up to 16.4Mbps of full duplex data throughput or 240 voice calls over eight T1 and or E1 lines to support high performance PCI-based routing and telephony systems. Advanced clocking features allow E1 and T1 lines to be mixed with full synchronization. As part of Sangoma's AFT range of products, the A108 makes use of the same high performance PCI interface that has been deployed in large quantities all over the world. The available A108d card includes Octasic\'s DSP hardware and certified algorithms providing carrier-grade echo cancellation and Voice Quality Enhancement (VQE) functions. Like all the Sangoma AFT family, the A108 is field upgradeable to take advantage of hardware and software improvements as they become available. Technical Specifications Eight T1/E1 ports with a single PCI interface for high performance voice and data applications. TDM clocking mode: Network timing can be passed from a network -connected DS0 to any or all of the other ports. Both T1 and E1 are supported simultaneously, making it possible to mix T1 channel banks and E1 networks with full channel synchronization. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Support for the Asterisk™, OPAL™ Yate, FreeSwitch™ PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. All Sangoma's AFT products, including the A108 card use the same base PCI interface card, and the same professionally engineered firmware on the same family of Field Programmable Gate Arrays. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing without manual tuning. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers. High quality, tested RJ45 cables included. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 ™ 50C. Autosense compatibility with 5v and 3.3v PCI busses. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDAC™ (patent pending) technology will be integrated to drastically reduce the cost of echo cancellation. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols such as non-byte aligned monosynch or bisynch. WAN data connection is supported by Sangoma's standard WANPIPE™ routing stack, and is completely independent of TDM voice application for total system reliability. WANPIPE™ supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Optional DSP daughterboard on the A108d - G.168-2002 echo cancellation in hardware - 1024 taps/128ms tail per channel on all 256 channels - DTMF decoding and tone recognition - Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. T1/E1 Status alarms RED: Telco Red Alarm condition YEL: Receive Telco Yellow Alarm ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows, FreeBSD, Open BSD, NetBSD, Solaris Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Higher level protocols Asterisk, Yate, OPAL Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Certification FCC Part 15 Class A, FCC Part 68, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 330 Rhino R2T1-e Dual T1/E1/PRI PCI Card, PCI Express (no EC) 367.0000 459.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r2t1e-dual-t1e1pri-pci-card-pci-express-no-ec-p-330.html http://www.voipon.co.uk/images/rhino_r2t1.gif new Availability: In Stock Linux Open Source Telephony Dual T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Dual T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code Quad T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interfacechip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket for each individual port All major signaling modes supported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such as Caller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable usingZaptel tools (i.e. zttool) Mechanical Data Size: 3.0? tall, 5.20? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 332 Polycom SoundPoint IP650 178.0000 195.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip650-p-332.html http://www.voipon.co.uk/images/soundpoint_ip650.jpg new Availability: In Stock Soundpoint IP 650 - High-Performance IP Phone with Polycom HD Voice Designed for both executive users who require advanced features and applications as well as telephone attendants who need multiple line support, the Polycom SoundPoint IP 650 sets a new standard for high-performance IP phones. Revolutionary Voice Quality The SoundPoint IP 650 is the first IP phone to use Polycom’s revolutionary HD Voice technology to bring life-like clarity and richness to voice communications. Polycom HD Voice incorporates wideband audio for more than twice the voice clarity, Polycom's patented Acoustic Clarity Technology 2, as well as best-in-class system design to deliver unrivalled voice quality. Advanced Features & Applications The Soundpoint Ip 650's SIP 2.0 software fully supports Microsoft Live Communications Server 2005 for telephony and presence, and interoperates with Microsoft Office Communicator. The SoundPoint IP 650 also includes a USB port for future applications. Enhanced Call Handling Capabilities The Polycom SoundPoint IP 650 accommodates 6 lines in standalone mode which increases to 12 lines as an attendant console when equipped with SoundPoint IP Expansion Modules. The IP650 supports shared call/bridged line appearance, an essential feature for effective phone communication between executives and their assistants. The phone's busy lamp field (BLF) functionality enables phone attendants to monitor the on-hook / off-hook status of key contacts, and despatch incoming calls for those contacts more efficiently. Expandability When equipped with up to expansion modules (up to 3), the SoundPoint IP 650 delivers the advanced call handling capabilities and enhanced user interface of a high-performance attendant console. Engineered to improve productivity of telephone attendants, the SoundPoint IP attendant console allows effective and efficient management and monitoring of up to 24 simultaneous calls on up to 12 lines. Intuitive User Interface The IP 650 provides all of its capabilities through an intuitive user interface, featuring a backlit grayscale 320x160 graphical LCD display, easy-to-navigate menu and a combination of 26 dedicated hard keys and 4 context-sensitive soft keys for easy access to essential telephony features. Efficient Installation and Provisioning The SoundPoint IP 650 is designed to make installation, configuration and upgrading as simple and efficient as possible, and boasts a two-port Ethernet switch and integrated Power over Ethernet circuitry. The SoundPoint IP 650 can be centrally configured and upgraded in the field using a FTP, TFTP, HTTP4, or HTTPS4 server and supports provisioning server redundancy. Proven Polycom is the leading independent supplier of standards-basedIP phones that are fully interoperable withkey IP PBX and Softswitch platform Polycom IP 650 Features   Advanced Features & Applications Integration with Microsoft LCS 2005 and Microsoft Office Communicator USB port for future applications XHTML microbrowser Backlit 320x160 graphical grayscale LCD Integrated PoE support Advanced Call Handling Capabilities Six lines (standalone) / 12 lines with Expansion Module(s) Shared call / bridged line appearance 2 Busy lamp field (BLF) 2 334 Xorcom Astribank-32 - 32 FXO Channel Bank 1089.0000 1227.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank32-32-fxo-channel-bank-p-334.html http://www.voipon.co.uk/images/xorcom_astribank_32.jpg new Availability: In Stock Astribank is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 32 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Read more... Features: Flexible port assignment: - 32 FXS ports, or - 16 FXS and 16 FXO ports, or - 24 FXS and 8 FXO ports, or - 8 FXS and 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current Caller ID support 336 Grandstream GXW-4104 Analog FXO Gateway 158.0000 194.00 Grandstream Grandstream Gateways http://www.voipon.co.uk/grandstream-gxw4104-analog-fxo-gateway-p-336.html http://www.voipon.co.uk/images/grandstream_gxw_4104_analog_fxo.jpg new Availability: In stock Overview The GXW-4104 offers an easy to manage, easy to configure IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls. There are two models - the GXW4104 and GXW4108, which have either 4 and 8 FXO ports respectively. The installation is the same for either model. A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-4104 series. In this environment, the SIP server handles SIP registration and call control and the GXW-4104 processes media conversion between IP and PSTN calls. By design, the system supports the North American call progress tones and signaling standards on PSTN sides. GXW-4104 Features 4 FXO analog port gateways Video surveillance Two RJ-45 ports (switched or routed) TFTP and HTTP firmware upgrade support Supports Audio Codecs: G711, G723, G729 and GSM T.38 compliant Web management for easy configuration and installation TFTP and HTTP firmware upgrade support Multiple SIP accounts, associated with physical line ports, each account corresponding to one of the multiple SIP profile Multiple SIP profiles, max of 3 profiles per system. Each profile hosts 0 to multiple number of SIP accounts, depending on user need One stage and two stage dialing Two stage dialing means when after dialing the number to the GXW, be it from VoIP to GXW or from PSTN to GXW, a second dial-tone prompts users to input the final destination number to finish final dialing. One stage dialing means user only hear dial-tone once and input a final destination number along with a pre-fix. One stage dialing need SIP server to support SIP call forward via a dial-plan. VoIP to PSTN call setup and teardown Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. PSTN to VoIP call setup and teardown Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. One stage dialing requires user to configure Off-Hook Auto Dial to a SIP Number. Support: G711, G723, G729, and GSM Line echo canceller g.168 support Flexible DTMF transmission method User Interface of In-audio, RFC2833, and SIP Info Round-robin port scheduling to ensure available lines to access PSTN networks Configurable channel dialing to improve dial-out reliability - digit length: default 100ms - digit volume: gain [-31,0]dB, default -11dB - dial pause between digits: default 100ms - wait for dial-tone: yes/no, default yes (1 for Yes, 2 for No) - one-stage ( use 1 ) or 2 stage (use 2) dialing: default of 2 stage dialing - Syntax: ch (or chan or channel) x-y: val; ch Configurable PSTN Termination - Enable current disconnect: default of disabled. Some special PBXs and CO lines use line power drop to indicate PSTN hang-up. When this is the configuration, please consult your PSTN line service provider for the correct PSTN disconnect method. - AC termination impedance: default North America. This impedance works with parameters of Busy/Re-order tone in Call Progress Table. Users have to set BUSY/REORDER tone values to enable this parameter. - Busy or re-order tones: following busy or reorder tone of call progress tones is used to teardown regular PSTN call if detected Configurable call progress/termination tones via pattern matching - Dial-tone: f1/f2(350/440), v1/v2( -11/ -11), on1/off1(0/0), on2/off2(0/0) - Ring back tone: f1/f2(default 440/480), on/off(default 2s/4s) - Busy tone: f1/f2(480/620), on/off(0.5/0.5s), duration (8s) - Re-order tone: f1/f2( 480/620 ), on/off(25/25), duration (default 8s) - Confirmation tone: f1/f2(350/440), on/off(0.1/0.1s), duration (default 8s) - Usage Syntax: - ch x-y: f1(or freq1 or frequency1)=val1@vol1, f2 (or freq2 or frequency2) = val2@vol2, c (or cad or cadence) = on1/off1-on2/off2-on3/off3; ch3: - x,y - 0-9 digit. - Configure Channel voice settings, - Voice volume: gain control, [-31, 31], default 1 dB - Audio input gain: [-31, 31], default 0 dB - Silence Suppression: 1 - enabled, 2 - disabled, default is 1 - Line echo cancellation: 1 - enabled, 2 - disabled; default is 1 Configure other channel settings, PSTN Silence Timeout, default 60 sec. This serves as a last measure to address PSTN run-away calls. It is not supposed to replace above regular PSTN disconnect methods. DTMF Method via : default value is in-audio 1 - in-audio 2 - RFC2833 3 - in-audio and RFC2833 4 - SIP Info 5 - in-audio and RFC2833 337 Grandstream GXW-4108 Analog FXO Gateway 250.0000 292.00 Grandstream Grandstream Gateways http://www.voipon.co.uk/grandstream-gxw4108-analog-fxo-gateway-p-337.html http://www.voipon.co.uk/images/grandstream_gxw_4108_analog_fxo.jpg new Availability: In stock Overview The GXW-4108 offers an easy to manage, easy to configure IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls. There are two models - the GXW4104 and GXW4108, which have either 4 and 8 FXO ports respectively. The installation is the same for either model. A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-4108 series. In this environment, the SIP server handles SIP registration and call control and the GXW4108 processes media conversion between IP and PSTN calls. By design, the system supports the North American call progress tones and signaling standards on PSTN sides. GXW-4108 Features 8 FXO analog port gateways Video surveillance Two RJ-45 ports (switched or routed) TFTP and HTTP firmware upgrade support Supports Audio Codecs: G711, G723, G729 and GSM T.38 compliant Web management for easy configuration and installation TFTP and HTTP firmware upgrade support Multiple SIP accounts, associated with physical line ports, each account corresponding to one of the multiple SIP profile Multiple SIP profiles, max of 3 profiles per system. Each profile hosts 0 to multiple number of SIP accounts, depending on user need One stage and two stage dialing Two stage dialing means when after dialing the number to the GXW, be it from VoIP to GXW or from PSTN to GXW, a second dial-tone prompts users to input the final destination number to finish final dialing. One stage dialing means user only hear dial-tone once and input a final destination number along with a pre-fix. One stage dialing need SIP server to support SIP call forward via a dial-plan. VoIP to PSTN call setup and teardown Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. PSTN to VoIP call setup and teardown Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. One stage dialing requires user to configure Off-Hook Auto Dial to a SIP Number. Support: G711, G723, G729, and GSM Line echo canceller g.168 support Flexible DTMF transmission method User Interface of In-audio, RFC2833, and SIP Info Round-robin port scheduling to ensure available lines to access PSTN networks Configurable channel dialing to improve dial-out reliability - digit length: default 100ms - digit volume: gain [-31,0]dB, default -11dB - dial pause between digits: default 100ms - wait for dial-tone: yes/no, default yes (1 for Yes, 2 for No) - one-stage ( use 1 ) or 2 stage (use 2) dialing: default of 2 stage dialing - Syntax: ch (or chan or channel) x-y: val; ch Configurable PSTN Termination - Enable current disconnect: default of disabled. Some special PBXs and CO lines use line power drop to indicate PSTN hang-up. When this is the configuration, please consult your PSTN line service provider for the correct PSTN disconnect method. - AC termination impedance: default North America. This impedance works with parameters of Busy/Re-order tone in Call Progress Table. Users have to set BUSY/REORDER tone values to enable this parameter. - Busy or re-order tones: following busy or reorder tone of call progress tones is used to teardown regular PSTN call if detected Configurable call progress/termination tones via pattern matching - Dial-tone: f1/f2(350/440), v1/v2( -11/ -11), on1/off1(0/0), on2/off2(0/0) - Ring back tone: f1/f2(default 440/480), on/off(default 2s/4s) - Busy tone: f1/f2(480/620), on/off(0.5/0.5s), duration (8s) - Re-order tone: f1/f2( 480/620 ), on/off(25/25), duration (default 8s) - Confirmation tone: f1/f2(350/440), on/off(0.1/0.1s), duration (default 8s) - Usage Syntax: - ch x-y: f1(or freq1 or frequency1)=val1@vol1, f2 (or freq2 or frequency2) = val2@vol2, c (or cad or cadence) = on1/off1-on2/off2-on3/off3; ch3: - x,y - 0-9 digit. - Configure Channel voice settings, - Voice volume: gain control, [-31, 31], default 1 dB - Audio input gain: [-31, 31], default 0 dB - Silence Suppression: 1 - enabled, 2 - disabled, default is 1 - Line echo cancellation: 1 - enabled, 2 - disabled; default is 1 Configure other channel settings, PSTN Silence Timeout, default 60 sec. This serves as a last measure to address PSTN run-away calls. It is not supposed to replace above regular PSTN disconnect methods. DTMF Method via : default value is in-audio 1 - in-audio 2 - RFC2833 3 - in-audio and RFC2833 4 - SIP Info 5 - in-audio and RFC2833 338 Sangoma A104x PCI Express PRI ISDN Card 734.4300 926.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a104x-pci-express-pri-isdn-card-p-338.html http://www.voipon.co.uk/images/sangoma_a104_pci_express_small.gif new Availability: In Stock The sandwich DSP card on the "d" model features Octasic's certified carrier-grade algorithms providing carrier grade echo cancellation and Voice Quality Enhancement (VQE) functions. Supporting 32-672 channels G.168-2002 echo cancellation, a minimum of 1024 taps for a 128ms tail/channel on all channel densities, the system also supports Octasic music protection, acoustic echo control and adaptive noise reduction. The "d" model also features on-board DTMF decoding and tone recognition. The A104 is the, quad port version of Sangoma's range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A104 provides full speed 132 Mbps PCI bus transfer with FPGA and DSP based processing to unload the host CPU in demanding environments such as soft PBX/IVR voice applications. Compatible with both the 3.3v and 5v PCI bus, A104 cards operate in all commercially available motherboards sharing IRQs properly with themselves and all other PCI compatible devices, so you never have to worry about hardware compatibility issues. Like all the Sangoma AFT Series, the A104 is field upgradeable to take advantage of the hardware and software improvements as they become available. Optional Modes The A104 and drivers fully support TDM voice gateways for the Asterisk? , Yate? , OPAL? PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. The A104 and drivers fully support TDM voice gateways for the Asterisk? , Yate? , OPAL? PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDACT (patent pending) technology is integrated to drastically reduce the load due to software echo cancellation, which is the largest component of CPU load in a typical soft PBX system. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Full channelized mode to act as major network hub for sub-DS1 remotes. The A104 can support any configuration of up to 124 remote 64kbps connections carrying Frame Relay, PPP or HDLC data. Timeslots can be concatenated to support remote fractional T1/E1/J1 sites in any combination. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Mixed Voice/Data mode: Robust combination of router/PBX functions in one server. WAN data connection is supported by Sangoma's standard WANPIPE? routing stack , completely independently of TDM voice application for total system reliability. WANPIPE? supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Technical Specifications Quad port T1/E1/J1 card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features related to voice and/or data to be added when they become available. DSP card on the A104d: o G.168-2002 echo cancellation in hardware o 1024 taps/128ms tail per channel on all channel densities o DTMF decoding and tone recognition Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Operating systems Linux (all versions, releases and distributions from 1.0 up). Higher level protocols Asterisk Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty FIVE years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 339 Sangoma A108 PCI PRI ISDN Card 1171.0100 1476.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a108-pci-pri-isdn-card-p-339.html http://www.voipon.co.uk/images/sangoma_a108_pcix_small.gif new Availability: In Stock The A108 is the octal port version of Sangoma's family of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of voice and data over T1, E1 and J1. The A108 provides up to 16.4Mbps of full duplex data throughput or 240 voice calls over eight T1 and or E1 lines to support high performance PCI-based routing and telephony systems. Advanced clocking features allow E1 and T1 lines to be mixed with full synchronization. As part of Sangoma's AFT range of products, the A108 makes use of the same high performance PCI interface that has been deployed in large quantities all over the world. The available A108d card includes Octasic's DSP hardware and certified algorithms providing carrier-grade echo cancellation and Voice Quality Enhancement (VQE) functions. Like all the Sangoma AFT family, the A108 is field upgradeable to take advantage of hardware and software improvements as they become available. Technical Specifications Eight T1/E1 ports with a single PCI interface for high performance voice and data applications. TDM clocking mode: Network timing can be passed from a network -connected DS0 to any or all of the other ports. Both T1 and E1 are supported simultaneously, making it possible to mix T1 channel banks and E1 networks with full channel synchronization. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Support for the Asterisk®, OPAL® Yate, FreeSwitch® PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. All Sangoma's AFT products, including the A108 card use the same base PCI interface card, and the same professionally engineered firmware on the same family of Field Programmable Gate Arrays. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper interrupt sharing without manual tuning. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers. High quality, tested RJ45 cables included. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Autosense compatibility with 5v and 3.3v PCI busses. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. EDAC® (patent pending) technology will be integrated to drastically reduce the cost of echo cancellation. Field upgradeable hardware allows for new TDM-related functions to be added as they become available. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols such as non-byte aligned monosynch or bisynch. WAN data connection is supported by Sangoma's standard WANPIPE® routing stack, and is completely independent of TDM voice application for total system reliability. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Optional DSP daughterboard on the A108d - G.168-2002 echo cancellation in hardware - 1024 taps/128ms tail per channel on all 256 channels - DTMF decoding and tone recognition - Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. T1/E1 Status alarms RED: Telco Red Alarm condition YEL: Receive Telco Yellow Alarm ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame RAIV: Receive Loss of Signal Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows, FreeBSD, Open BSD, NetBSD, Solaris Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Higher level protocols Asterisk, Yate, OPAL Open PBX/IVR, IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Certification FCC Part 15 Class A, FCC Part 68, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 340 Linksys SPA-2102 Analog Telephone Adaptor 42.0000 46.95 Linksys Linksys Analog Adaptors http://www.voipon.co.uk/linksys-spa2102-analog-telephone-adaptor-p-340.html http://www.voipon.co.uk/images/linksys_2102.jpg new Availability: In Stock Inexpensive, easy to install and simple-to-use, the SPA2102 connects a standard telephone or fax machine to IP-based data networks. VoIP service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet. The SPA2102 features two POTS (Plain Old Telephone Service) ports to connect existing analog phones, fax machines, or to a PBX system. The SPA2102 includes 2 100BaseT RJ45 Ethernet interfaces (LAN-WAN) to connect to a home or office LAN, as well as an Ethernet connection to a broadband modem or router (WAN). Each SPA2102 service line can be independently configured via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in telephones, conference speakerphones, and fax machines as well as control their migration to IP with an extremeley affordable, reliable solution. Installed by the end user and remotely provisioned, configured and maintained by the service provider, each SPA2102 converts voice traffic into data packets for transmission over an IP network. Compact in design, the SPA2102 can be used in consumer and business VoIP service offerings including a full-featured IP Centrex environment. The SPA2102 uses international standards for voice and data networking for reliable voice and fax operation. Features Telephony Two voice ports (RJ11) for analog phones or Fax machines Impedance Agnostics - 8 Configurable Settings Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Caller ID with Name/Number (Multi-national Variants) Caller ID Blocking Call Forwarding: No answer, Busy, All Do Not Disturb Call Transfer Three-way Conference Calling with Local Mixing Message Waiting Indication - Visual and Tone Based Call Return Call Back on Busy Call Blocking with Toll Restriction Delayed Disconnect Distinctive Ringing - Calling and Called Number Off-hook Warning Tone Selective/Anonymous Call Rejection Hot line and Warm Line Calling Speed Dialing of 8 Numbers/Addresses Music on Hold Fax: G.711 Pass Through or Real Time Fax over IP via T.38 Package Contents 1 - SPA2102 Phone Adapter Unit 1 - 5v Power Adapter 1 - RJ45 Ethernet Cable 1 - Quick Installation Guide Product Data Toll Quality Voice and Carrier-Grade Feature Support The SPA2102 delivers clear, high-quality voice communica-tion in a variety of network con-ditions. Excellent voice quality in challenging, changeable IP network environments is made possible via our advanced im-plementation of standard voice coding algorithms. The SPA2102 is interoperable with common telephony equipment like fac-simile, voicemail, PBX/KTS and interactive voice response sys-tems. Large-Scale Deployment and Management The SPA2102 offers all the key features and capabilities with which service providers can provide customized services to their subscribers. The SPA2102 can be remotely provisioned and supports dynamic, in-service software upgrades. A secure profile upload saves providers the time, expense and hassle of managing and pre-configuring or re-configur-ing customer premises equip-ment (CPE) for deployment. Ironclad Security The SPA2102 supports secure, encryption-based methods for communication, provisioning and servicing. Linksys SPA Series Analogue Telephone Adapter (ATA) Comparison Chart Model Service Lines Active Calls 3-Way Call Conferences Network Ports PSTN (FXO) Ports Phone (FXS) Ports SPA1001 2 2 1 1 0 1 PAP2T 2 4 2 1 0 2 SPA2102 2 4 2 2 0 2 SPA3102 2 3 1 2 1 1 SPA400 N/A N/A N/A 1 4 0 342 VXI Parrott Headset 14.5 14.50 VXI Corporation VoIP Accessories http://www.voipon.co.uk/vxi-parrott-headset-p-342.html http://www.voipon.co.uk/images/vxi_parrott_cp150.jpg new Availability: This product has been discontinued The VXI Parrott CP150 is a comfortable cost-effective monaural headset suitable for use with the Sipura/Linksys range of IP phones. It has a 2.5mm jack plug which means it can also be used with most mobile phones and any phone with a 2.5mm jack headset socket. The noise cancelling microphone helps block out background noise for improved audio quality. The flexible microphone boom adjusts for left or right side wear and the adjustable headband won't loosen after you position it. Lightweight and stable, the CP150 sits securely on your head while you're on the move. Features Comfortable for long wearing Flexible microphone boom Noise cancelling microphone Left or right side microphone placement Adjustable no-slip headband for stable fit Works well with SPA-841 and SPA-941 phones Compatible with most phones with a 2.5mm jack socket 343 Ingate SIParator 19 615 615.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-siparator-19-p-343.html http://www.voipon.co.uk/images/ingate_siparator_18.jpg new Availability: In Stock Leveling the playing field for smaller enterprises, Ingate® Systems offers the Ingate SIParator® 19, a powerful tool that offers small businesses, branch offices and home workers complete support for IP communications based on Session Initiation Protocol (SIP). With the SIParator 19, these businesses can leverage the same productivity and cost-savings benefits of Voice over IP (VoIP) and other IP-based communications as large corporations. The SIParator works seamlessly with your existing firewall to allow the flow of the SIP traffic. While traditional firewalls block SIP traffic - including mission-critical applications like VoIP - the SIParator resolves this problem, working in tandem with your current security solutions. It also solves the Network Address Translation (NAT) traversal issues inherent in SIP communications, and offers both far- and near-end NAT traversal to extend the SIP capabilities within the corporate network to remote workers. With Ingate products, enterprises can use VoIP and other live communications on the LAN and globally over the Internet or private IP networks. The Ingate SIParator 19 has three ports and offers a 30 Mbit/s throughput. Small and versatile, the 19 is perfect for smaller office environments - with no fan; the SIParator is virtually silent, which means there's no need for a separate server room. The management interface for the products is the same Web-based Graphical User Interface (GUI) that has been cited by Ingate customers and the media for ease-of-use. Included with the Ingate SIParator 19 are ten SIP user licenses that can be used for the registration of SIP users on the built-in SIP registrar. Five SIP traversal licenses are also included, allowing up to five calls to traverse at the same time. Additional SIP user and traversal licenses can be purchased at any time. All Ingate SIParators are fully featured and can be maintained by the network security administrator utilizing the GUI. Ingate SIParators include an encrypted Virtual Private Network (VPN) termination module. The SIParator 19 can be configured as a part of the DMZ or in a standalone mode. In both cases, the benefits of SIP-based communications can be added to the network quickly and easily. Choose the Right Features for Your Network Ingate offers several add-on software modules that allow you to tailor the SIParators to meet the specific demands of your business. Ingate Quality of Service (QoS) sets priorities to different kinds of data and allocates bandwidth for varied purposes - for instance, giving priority to VoIP. Ingate Remote SIP Connectivity extends the SIP capabilities of the enterprise to employees working remotely (home office workers, road warriors, etc.). Remote SIP Connectivity manages the traversal of the remote NAT from a central place and also includes a STUN server. Ingate VoIP Survival software module adds a whole new dimension to the hosted VoIP service by securing full redundancy in a SIP-based hosted IP-PBX environment all the way out to the customer premise. It serves as a customer premise backup to enhance the reliability and availability of a VoIP application platform. Add Global VoIP Connectivity to your IP-PBX The SIParator 19 opens up a world of possibilities and cost savings when used with a SIP based IP-PBX. Businesses can route telephone calls via IP, not only between branch offices and home workers, but also to offices and other users using SIP-based Internet telephony. No longer limited to telephony voice, communication can also include video, instant messaging, presence and more. In addition, the SIParator 19 makes it possible for home workers, road warriors and even branch offices to belong the same central IP-PBX - with the highest level of security. The SIParator also affords the possibility to set up a private VoIP network, if preferred. Advanced IP-PBX functions are supported, including such as call transfer, call hold, and voicemail. Do More with Microsoft LCS 2005 Adding value to Microsoft Live Communications Server (LCS) 2005, Ingate SIParators allow voice and video applications to work outside the LAN. In addition, Ingate makes LCS available to mobile users through Remote SIP Connectivity. Trusted Network Security for VoIP The Ingate SIParator SIP Proxy architecture grants fully secure traversal of the SIP traffic. The ports for the media streams are only opened between the specific parties of a call and only for the duration of the call. The SIP proxy inspects the SIP packets before sending them on. TLS encryption ensures privacy when communicating, making call eavesdropping, call hijacking and call spoofing harder to do. Ingate also supports authentication of users and servers. Free Software Upgrades for the First Year The Ingate SIParator 19 has no data user restrictions. Software upgrades are free for the first year. Thereafter, an annual licensing fee will apply. New software versions can be downloaded quickly and easily online from the Ingate website. 344 Ingate SIParator 45 2144 2144.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-siparator-45-p-344.html http://www.voipon.co.uk/images/ingate_siparator_45.jpg new Availability: In Stock The Ingate SIParator 45 and 45+ are powerful tools that offer businesses complete support for IP communications, such as Voice over IP (VoIP), based on Session Initiation Protocol (SIP). With the SIParator , businesses can harness the productivity and cost-savings benefits of VoIP and other IP-based communications while maintaining current investments in security technology. The SIParator works seamlessly with your existing firewall to allow the flow of SIP traffic. While traditional firewalls block SIP traffic &ndash; including mission-critical applications like VoIP &ndash; the SIParator resolves this problem, while working in tandem with your current security solutions. Ingate SIParators also solve the Network Address Translation (NAT) traversal issues inherent in SIP communications, and offer both far- and near-end NAT traversal to extend the SIP capabilities within the corporate network to remote workers. With Ingate products, enterprises can use VoIP and other live communications on the LAN and globally over the Internet or private IP networks. The Ingate SIParator 45 has four ports and can handle up to 150 concurrent RTP-sessions. For companies needing more capacity Ingate has developed the SIParator 45+. This SIParator can, for example, handle up to 240 concurrent RTP-sessions. The hardware and the functionality are otherwise the same both for the Ingate SIParator 45 and the Ingate SIParator 45+. Included with the Ingate SIParator 45 and 45+ are ten SIP user licenses that can be used for the registration of SIP users on the built-in SIP registrar. Five SIP traversal licenses are also included, allowing up to five calls to traverse at the same time. Additional SIP user and traversal licenses can be purchased at any time. All Ingate SIParators are fully featured and can be maintained by the network security administrator utilizing the GUI. Ingate SIParators include an encrypted Virtual Private Network (VPN) termination module. The SIParator 45 and 45+ can be configured as a part of the DMZ or in a standalone mode. In both cases, the bene?ts of SIP-based communications can be added to the network quickly and easily. Choose the Right Features for Your Network Ingate offers several add-on software modules that allow you to tailor the SIParators to meet the specific demands of your business. Ingate Quality of Service (QoS) sets priorities to different kinds of data and allocates bandwidth for varied purposes &ndash; for instance, giving priority to VoIP. Ingate Remote SIP Connectivity extends the SIP capabilities of the enterprise to employees working remotely (home of?ce workers, road warriors, etc.). Remote SIP Connectivity manages the traversal of the remote NAT from a central firewall and also includes a STUN server. Ingate VoIP Survival adds a whole new dimension to hosted VoIP service by securing full redundancy in a SIP-based hosted IP-PBX environment all the way out to the customer premises. It serves as a backup to enhance the reliability and availability of a VoIP application platform. Add Global VoIP Connectivity to your IP-PBX The SIParator 45 and 45+ open up a world of possibilities and cost savings when used with a SIP-based IP-PBX. Businesses can route telephone calls via IP, not only between branch offices and home workers, but also to offices and other users using SIP-based Internet telephony. No longer limited to telephony voice, communication can also include video, instant messaging, presence and more. In addition, the SIParator 45 and 45+ make it possible for home workers, road warriors and even branch offices to belong the same central IP-PBX &ndash; with the highest level of security. The SIParators also afford the possibility to set up a private VoIP network, if preferred. Advanced IP-PBX functions are supported, including such as call transfer, call hold and voicemail. Do More with Microsoft LCS 2005 Adding value to Microsoft Live Communications Server (LCS) 2005, Ingate SIParators allow voice and video applications to work outside the LAN. In addition, Ingate makes LCS available to mobile users through Remote SIP Connectivity. Trusted Network Security for VoIP The SIParator SIP proxy architecture grants fully secure traversal of the SIP traffc. The ports for the media streams are only opened between the specific parties of a call and only for the duration of the call. The SIP proxy inspects the SIP packets before sending them on. TLS encryption ensures privacy when communicating, making call eavesdropping, call hijacking and call spoofing harder to do. Ingate also supports authentication of users and servers. Free Software Upgrades for the First Year The Ingate SIParators 45 and 45+ have no data user restrictions. Software upgrades are free for the ?rst year. Thereafter, an annual licensing fee will apply. New software versions can be downloaded quickly and easily online from the Ingate website. 345 Ingate SIParator 45 Plus 3844 3844.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-siparator-45-plus-p-345.html http://www.voipon.co.uk/images/ingate_siparator_45_plus.jpg new Availability: In Stock The Ingate SIParator 45 and 45+ are powerful tools that offer businesses complete support for IP communications, such as Voice over IP (VoIP), based on Session Initiation Protocol (SIP). With the SIParator , businesses can harness the productivity and cost-savings benefits of VoIP and other IP-based communications while maintaining current investments in security technology. The SIParator works seamlessly with your existing firewall to allow the flow of SIP traffic. While traditional firewalls block SIP traffic &ndash; including mission-critical applications like VoIP &ndash; the SIParator resolves this problem, while working in tandem with your current security solutions. Ingate SIParators also solve the Network Address Translation (NAT) traversal issues inherent in SIP communications, and offer both far- and near-end NAT traversal to extend the SIP capabilities within the corporate network to remote workers. With Ingate products, enterprises can use VoIP and other live communications on the LAN and globally over the Internet or private IP networks. The Ingate SIParator 45 has four ports and can handle up to 150 concurrent RTP-sessions. For companies needing more capacity Ingate has developed the SIParator 45+. This SIParator can, for example, handle up to 240 concurrent RTP-sessions. The hardware and the functionality are otherwise the same both for the Ingate SIParator 45 and the Ingate SIParator 45+. Included with the Ingate SIParator 45 and 45+ are ten SIP user licenses that can be used for the registration of SIP users on the built-in SIP registrar. Five SIP traversal licenses are also included, allowing up to five calls to traverse at the same time. Additional SIP user and traversal licenses can be purchased at any time. All Ingate SIParators are fully featured and can be maintained by the network security administrator utilizing the GUI. Ingate SIParators include an encrypted Virtual Private Network (VPN) termination module. The SIParator 45 and 45+ can be configured as a part of the DMZ or in a standalone mode. In both cases, the bene?ts of SIP-based communications can be added to the network quickly and easily. Choose the Right Features for Your Network Ingate offers several add-on software modules that allow you to tailor the SIParators to meet the specific demands of your business. Ingate Quality of Service (QoS) sets priorities to different kinds of data and allocates bandwidth for varied purposes &ndash; for instance, giving priority to VoIP. Ingate Remote SIP Connectivity extends the SIP capabilities of the enterprise to employees working remotely (home of?ce workers, road warriors, etc.). Remote SIP Connectivity manages the traversal of the remote NAT from a central firewall and also includes a STUN server. Ingate VoIP Survival adds a whole new dimension to hosted VoIP service by securing full redundancy in a SIP-based hosted IP-PBX environment all the way out to the customer premises. It serves as a backup to enhance the reliability and availability of a VoIP application platform. Add Global VoIP Connectivity to your IP-PBX The SIParator 45 and 45+ open up a world of possibilities and cost savings when used with a SIP-based IP-PBX. Businesses can route telephone calls via IP, not only between branch offices and home workers, but also to offices and other users using SIP-based Internet telephony. No longer limited to telephony voice, communication can also include video, instant messaging, presence and more. In addition, the SIParator 45 and 45+ make it possible for home workers, road warriors and even branch offices to belong the same central IP-PBX &ndash; with the highest level of security. The SIParators also afford the possibility to set up a private VoIP network, if preferred. Advanced IP-PBX functions are supported, including such as call transfer, call hold and voicemail. Do More with Microsoft LCS 2005 Adding value to Microsoft Live Communications Server (LCS) 2005, Ingate SIParators allow voice and video applications to work outside the LAN. In addition, Ingate makes LCS available to mobile users through Remote SIP Connectivity. Trusted Network Security for VoIP The SIParator SIP proxy architecture grants fully secure traversal of the SIP traffc. The ports for the media streams are only opened between the specific parties of a call and only for the duration of the call. The SIP proxy inspects the SIP packets before sending them on. TLS encryption ensures privacy when communicating, making call eavesdropping, call hijacking and call spoofing harder to do. Ingate also supports authentication of users and servers. Free Software Upgrades for the First Year The Ingate SIParators 45 and 45+ have no data user restrictions. Software upgrades are free for the ?rst year. Thereafter, an annual licensing fee will apply. New software versions can be downloaded quickly and easily online from the Ingate website. 346 Ingate SIParator 90 9894 9894.00 Ingate Systems Ingate Firewalls/SIParators http://www.voipon.co.uk/ingate-siparator-90-p-346.html http://www.voipon.co.uk/images/ingate_siparator_90.jpg new Availability: In Stock The Ingate SIParator® 90 is a powerful tool that offers large enterprises a controlled and secured migration to VoIP (Voice over IP) and other live communications, based on Session Initiation Protocol (SIP). With the SIParator, even the largest of businesses, with branch of? ces around the world and remote workers, can easily harness the productivity and cost-saving benefits of VoIP and other IP-based com-munications while maintaining current investments in security technology. Available only from Ingate® Systems, the SIParator works seamlessly with your existing firewall to allow the low of SIP traffic to reach the user in the enterprise, no matter where he or she is located, as long as there is an Internet connection. While traditional firewalls are not SIP-capable and, therefore, block SIP traffic - including mission-critical applications like VoIP - the SIParator resolves this problem, and enables SIP-based communication to traverse the firewall, while working in tandem with your current security solutions. Ingate SIParators solve the Network Address Translation (NAT) traversal issues inherent in SIP communications, and offer both far- and near-end NAT traversal to extend the SIP capabilities within the corporate network to remote workers. With Ingate products, enterprises can use VoIP and other live communications on the LAN and globally over the Internet or private IP networks. Ingate SIParator 90 The Ingate SIParator 90 has eight interfaces, two of which are mini Gbic that can be used fiber optic interfaces for greater flexibility. The SIParator 90 is capable of handling 1200 concurrent VoIP calls (e.g. RTP-sessions), a critical capability for large enterprises. All Ingate SIParators are fully featured, supporting stateful inspection and packet filtering with rules defined and maintained by the network security administrator utilizing the GUI. Trusted Security for VoIP, IP Applications The Ingate SIParator's SIP proxy architecture grants fully secure tra-versal of the SIP traffic. The ports for the media streams are only opened between the specific parties of a call and only for the duration of the call. The SIP proxy inspects the SIP packets before sending them on. TLS and SRTP encryption of traffic (both signaling and media) secure privacy when communicating, making call eaves-dropping, call hijacking and call spoofing harder to do. Ingate also supports authentication of users and servers. Choose the Right Features for Your Network Ingate offers several add-on software modules that allow you to tailor the Ingate SIParator 90 to meet the specific demands of your business. Ingate Quality of Service (QoS) sets priorities to different kinds of data and allocates bandwidth for varied purposes - for instance, giving priority to VoIP. Ingate Remote SIP Connectivity extends the SIP capabilities of the enterprise to employees working remotely (home office workers, road warriors, etc.). Remote SIP Connectivity manages the traversal of the remote NAT from a central location and also includes a STUN server. Ingate VoIP Survival adds a whole new dimension to the hosted VoIP service or branch offices using the IP-PBX at the main office by securing redundancy if the connection to the VoIP service or IP-PBX should fail. It serves as a backup to enhance the reliability and availability of a VoIP application platform. Ingate Advanced SIP Routing provides the ability to fullly control and route SIP traffic in an advanced and flexible manner. The module can handle least cost routing, enabling enterprises to make global calls for local fees. It also removes all geographic boundaries as calls to ie. support or customer service teams can be routed to the person who is available at the moment, no matter where he or she is located. Ingate's scalable traversal license system allows for the purchase of as many licenses as necessary to handle the needs of the growing enterprise. Additional licenses can be purchased at any time. Add Global VoIP Connectivity to your IP-PBX The SIParator 90 opens up a world of possibilities and cost savings when used with a SIP based IP-PBX. Businesses can route telephone calls via IP, not only between branch offices and home workers, but also to of? ces and other users using SIP-based Internet telephony. No longer limited to telephony voice, communication can also include video, instant messaging, presence and more. In addition, the SIParator 90 makes it possible for home workers, road warriors and even branch offices to belong the same central IP-PBX -- with the highest level of security. The 90 also affords the possibility to set up a private VoIP network, if preferred. Advanced IP-PBX functions are supported, including such as call transfer, call hold, and voicemail. Do More with Microsoft LCS 2005 Adding value to Microsoft Live Communications Server (LCS) 2005, Ingate SIParators allow voice and video applications to work outside the LAN. In addition, Ingate makes LCS available to mobile users through Remote SIP Connectivity. The SIParator also has upport for Microsoft encrypted media/Microsoft RTP (Realtime Transport Protocol). Free Software Upgrades for the First Year The Ingate SIParator 90 has no data user restrictions. Software upgrades are free for the first year. Thereafter, an annual licensing fee will apply. New software versions can be downloaded quickly and easily online from the Ingate website. 347 Redfone foneBRIDGE2 Dual T1/E1 Ethernet Bridge 594.0000 698.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2-dual-t1e1-ethernet-bridge-p-347.html http://www.voipon.co.uk/images/redfone_fonebridge2_e1_t1_smaller.jpg new Availability: In stock The foneBRIDGE2 Dual is a 2 Port E1/T1 Ethernet Bridge Asterisk PBX T1/E1 Redundancy, High Availability & Load Balancing in an economical Enterprise class solution foneBRIDGE2 is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black-box \"appliance\" designed to streamline installation and enable redundant design of Asterisk based VoIP systems. foneBRIDGE2 eliminates the need to install proprietary TDM hard-ware cards in approved/compatible server configurations. Instead, foneBRIDGE2 terminates T1/E1 PRI lines on the trunk side and pro-vides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities. Engineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, foneBRIDGE2 provides low-latency delivery of your critical voice traffic Enables T1/E1 Load Balancing Share trunk resources among multiple Asterisk servers through the load balancing capabilities of foneBRIDGE2 Failover & Asterisk HA Enabled High-Availability through rapid failover capabilities ensures your critical telephony services are always on and available Flexible Configuration Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment Solid State Embedded Appliance No moving components Low heat and power consumption High MTBF (mean time before failure) Target Applications T1/E1 PRI Trunk termination Legacy PBX-to-Asterisk integration High-Availability/Failover Asterisk clusters Channel Bank connectivity Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) Standards Telephony PRI Switch Compatibility- EuroISDN, AT&T 4ESS, DMS 100, Lu-cent 5E, NI1/NI2; Network or CPE Line Interface- Dual or Quad T1/E1 (RJ45), per port configur-able Line Encoding- AMI/B8ZS for T1, AMI/HDB3 for E1 Super Frame (SF) and Extended Super Frame (ESF) Robbed Bit Signaling (RBS/CAS) Ethernet 10/100-BASE-TX Half/Full-duplex 2 x Dedicated RJ45 Specifications Electrical: DC 500mA Max @ 5V (2.5W) Environmental: 0 to 50 deg C operating Physical: Dimensions: 5.00" x 6.50" x 1.00" Weight: 1 pound Mounting: Flange Mount or Desktop 348 Redfone foneBRIDGE2 Quad T1/E1 Ethernet Bridge 925.0000 948.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2-quad-t1e1-ethernet-bridge-p-348.html http://www.voipon.co.uk/images/redfone_fonebridge2_e1_t1_smaller.jpg new Availability: In stock The foneBRIDGE2 Quad is a 4 Port E1/T1 Ethernet Bridge Asterisk PBX T1/E1 Redundancy, High Availability & Load Balancing in an economical Enterprise class solution foneBRIDGE2 is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP systems. foneBRIDGE2 eliminates the need to install proprietary TDM hard-ware cards in approved/compatible server configurations. Instead, foneBRIDGE2 terminates T1/E1 PRI lines on the trunk side and pro-vides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities. Engineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, foneBRIDGE2 provides low-latency delivery of your critical voice traffic Enables T1/E1 Load Balancing Share trunk resources among multiple Asterisk servers through the load balancing capabilities of foneBRIDGE2 Failover & Asterisk HA Enabled High-Availability through rapid failover capabilities ensures your critical telephony services are always on and available Flexible Configuration Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment Solid State Embedded Appliance No moving components Low heat and power consumption High MTBF (mean time before failure) Target Applications T1/E1 PRI Trunk termination Legacy PBX-to-Asterisk integration High-Availability/Failover Asterisk clusters Channel Bank connectivity Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) Standards Telephony PRI Switch Compatibility- EuroISDN, AT&T 4ESS, DMS 100, Lu-cent 5E, NI1/NI2; Network or CPE Line Interface- Dual or Quad T1/E1 (RJ45), per port configur-able Line Encoding- AMI/B8ZS for T1, AMI/HDB3 for E1 Super Frame (SF) and Extended Super Frame (ESF) Robbed Bit Signaling (RBS/CAS) Ethernet 10/100-BASE-TX Half/Full-duplex 2 x Dedicated RJ45 Specifications Electrical: DC 500mA Max @ 5V (2.5W) Environmental: 0 to 50 deg C operating Physical: Dimensions: 5.00" x 6.50" x 1.00" Weight: 1 pound Mounting: Flange Mount or Desktop 349 Siemens C460IP Dect SIP Phone 58.9000 67.00 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-c460ip-dect-sip-phone-p-349.html http://www.voipon.co.uk/images/siemens_c460ip-sml.jpg new Availability: In Stock The Siemens Gigaset C460IP DECT IP phone is a dual mode phone system which allows DECT telephones to make VoIP call over an Internet connection, or land line calls over a fixed line connection (such as a BT phone line). One VoIP call and one fixed line call can be made at the same time with multiple handsets. With the Gigaset C460IP you don't need a PC to start making Internet phone calls. One DECT handset can be used to make and receive VoIP and land line calls, switch between the two with a single key press. The C460IP contains a DECT base-station and a single handset, in total up to 6 GAP-compatible DECT handsets can be registered with the single base station. Simply plug the C460IP into a broadband Internet connection and a phone connection and in a few simple steps you can start phoning over the Internet. Key Features One phone for Internet and land line calls One VoIP and one land line call simultaneously on multiple handsets No PC required High quality colour display on handset Register up to 6 GAP-compatible handsets Transfer calls between (C460IP) handsets UK Caller ID Multiple polyphonic ringtones Personal address book with 100 entries SMS messaging on handset (provider dependent) Configuration using handset or built-in web server Long battery life: -over 150 hours standby -over 10 hours talk time Typical handset range: up to 50 metres indoors up to 300 metres outdoors Handset Display 101 x 80 pixel backlit display 4096 colours 3-line display Handset Keypad Amber backlit keypad Dedicated illuminated keys for hands-free and message waiting Flash key (R) with long press for pause (P) Keypad Protection with long press of hash key (#) Acoustics Hands-free speaking using separate speaker and adjustable volume Adjustable handset volume 2.5mm headset connector Power Supply Switching Type with Modular Wall Plug Clip - County/Region Specific DC Input Voltage: +5 VDC at 2.0 A Max Power Consumption: 5 WATTs Power Adaptor: 100-240v- 50-60Hz (26-34VA) AC Input, 1.8m cord Call Features Speed dial using keys 2 - 9 Automatic prefix of dialling codes Caller ID display Selectable ringer with polyphonic ringtones Ringer output in handset and base station Directory - 100 name and numbers Illuminated message waiting (MWI) key Missed and made calls log SMS Features Transmission of SMS (provider dependent) with up to 160 characters per message Storage of up to 15 messages in inbox and 5 in outbox SMS protocol 1 Alarm Features Alarm with selectable ringer melody Snooze function Activation by dedicated alarm button DECT System Features (multiple handsets) VoIP and land line calls simultaneously DECT transfer of external calls between handsets, blind or attended 3-way conference (1 external, 2 internal participants) Networking Ethernet (802.3) RJ-45 connection Land line (PSTN) RJ-11 connection SIP VoIP support to RFC3261 Multiple voice codec support: G.711 G.726 G729 A/B DHCP client Web (HTTP) server for configuration Quality-of-Service: ToS, DiffServ PPPoE Termination Handset General Information Standby time: over 150 hours Talk time: over 10 hours 2x NiMH AAA Batteries Indoor range: up to 50 metres Outdoor range: up to 300 metres Dimensions: Length: 140.7mm Width: 53.2mm Height: 28mm Base Station Information 230VAC Power supply Paging button Connection of up to 6 DECT handsets Hybrid functionality: VoIP and PSTN access Package Contents Gigaset handset and charger 2x NiMH AAA batteries Gigaset standalone DECT base statio Telephone cable Ethernet cable 2x power supply units (charger and base station) Quick install guide   Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features All units supplied as base and 1 handset - Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. All units can connect to both a standard BT phone line and have an ethernet connection for VoIP calling. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice   350 Siemens C46 Handset 32.0000 33.50 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-c46-handset-p-350.html http://www.voipon.co.uk/images/siemens_c46-sml.jpg new Availability: In Stock Handset Display Handset with illuminated colo ur display with 4096 colours and 4 lines (3 lines + 1 line with 2 softkeys) In Idle/ standby state Date and Time displayed (buffered in case of power loss) State of charging and field intensity displayed Display of a screensaver: big display of time in digital clock style and colour pictures In talk state Display of call duration Keypad of handset (selection) Keypad illumination Dedicated illuminated keys for Handsfree+Message waiting Flash-key (R) with longpress for pause (P) Keypad protection on/off by longpress of # key Acoustics Handsfree speaking with separate speaker activated by illuminated handsfree-key and with adjustable volume 3 earpiece volumes Headset chinch 2.5mm Receiving Calls CLIP and CNIP displayed Selectable Ringer with - min. 3 polyphonic (16 voices) - 10 standard melodies - adjustable volume Ringer output - in handset via handsfree speaker Directory Phone directory: up to 100 names and numbers Additional features Alarm clock with selectable ringer melody+snooze function Technical Features Handset Standby-time: > 150 hours Talk time: > 10 hours Battery: 2 x NiMH AAA Indoor range: up to 50 meters Outdoor range: up to 300 meters Dimensions (Retail version) Length 145 mm, Width 53 mm, Height 28 mm Package contents 1 Handset 1 Belt clip 2 x NiMH AAA 1 Power supply unit 1 Telephone cord 1 User manual Key Features DECT and GAP compatible One phone for Internet and land line calls 5 additional phones can be connected up to the Siemens C460IP One VoIP and one land line call simultaneously on multiple handsets High quality colour display on handset Transfer calls between handsets UK Caller ID Personal address book with 100 entries SMS messaging on handset (provider dependent) Configuration using handset or built-in web server Long battery life   Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 351 Intertex IX68 ADSL 134.95 134.95 Intertex Intertex Routers http://www.voipon.co.uk/intertex-ix68-adsl-p-351.html http://www.voipon.co.uk/images/intertex_ix68_regular.jpg new Availability: Discontinued The Intertex IX68 range is no longer available, it has been replaced by the Intertex SurfinBird IX78 ADSL AIR GW2. This can be found here The Intertex IX68 ADSL has 4 Ethernet Ports, SIP Firewall, ADSL Router, and ADSL 2+ modem The Intertex IX68 broadband router is the first in the world to allow SIP communication. It has a very general and well-proven SIP support by integrating a SIP proxy and a SIP registrar, dynamically controlling the NAT and firewall. Conventional firewalls and NAT routers cannot even be configured to handle SIP between a LAN and the Internet! Microsoft® Windows® Messenger as well as the Live Communications Server are based on SIP and are expected to rapidly bring Presence, In-stant Messaging, Voice, Video and Data Collaboration to the enterprise and everyone's desktop. The SIP Switch streamlines the usage of email-like SIP ad dresses used by soft SIP clients and numbers used by SIP phones. It routes common phone calls to PSTN gateways, does ENUM look up, can forward calls and can fork incoming calls to several phones or to voice mail. You can also get your own local numbers, dial 0 (or 9) to reach the outside world and other functions familiar in the old telephony world. In addition you get a set of new features only available in the IP world. All the functions and features you need and more: Easy installation and maintenance Provisioning Display and front panel keys Auto login and keep-alive Email notification Dynamics DNS client Quality of service Network monitoring Pre-Configured There are preset configurations for numerous broadband services. Easy IP Address Handling IP addresses are retrieved and distributed automatically via DHCP (or set manually). Only a single IP address is required from your provider. Auto Login and Keep-Alive No more frequent logging in and entering of your password! - The IX68 logs into your broadband service and keeps the connection alive. Flexible Interface Usage All interfaces can be bridged to form a secured and protected Local Area Network (LAN). One of the Ethernet ports, and the wireless interface, can be used as separate subnets or a DMZ. Network Status The status of the PCs and IP telephones on your network is monitored and clearly displayed. SIP Communication on Your LAN Live person-to-person communication, such as voice, video, presence, instant messaging and more is expected to be the next large usage of the Internet after email and web surfing! SIP is the Internet protocol for such applications. The SIP Switch - The Local IP PBX In addition to including a complete SIP server, the IX68 can also be equipped with the SIP Switch. It makes the IX68 your local PBX for soft SIP clients like Microsoft® Windows® Messenger and SIP telephones, integrating them with each other, the SIP world and the old telephone network via gateways. 352 Intertex IX68 ADSL AIR 169.95 169.95 Intertex Intertex Routers http://www.voipon.co.uk/intertex-ix68-adsl-air-p-352.html http://www.voipon.co.uk/images/intertex_ix68.jpg new Availability: Discontinued The Intertex IX68 range is no longer available, it has been replaced by the Intertex SurfinBird IX78 ADSL AIR GW2. This can be found here The Intertex IX68 ADSL Air has 4 Ethernet Ports, Wireless SIP Firewall, ADSL Router at 154 Mbps, and ADSL 2+ modem The Intertex IX68 broadband router is the first in the world to allow SIP communication. It has a very general and well-proven SIP support by integrating a SIP proxy and a SIP registrar, dynamically controlling the NAT and firewall. Conventional firewalls and NAT routers cannot even be configured to handle SIP between a LAN and the Internet! Microsoft® Windows® Messenger as well as the Live Communications Server are based on SIP and are expected to rapidly bring Presence, In-stant Messaging, Voice, Video and Data Collaboration to the enterprise and everyone's desktop. The SIP Switch streamlines the usage of email-like SIP ad dresses used by soft SIP clients and numbers used by SIP phones. It routes common phone calls to PSTN gateways, does ENUM look up, can forward calls and can fork incoming calls to several phones or to voice mail. You can also get your own local numbers, dial 0 (or 9) to reach the outside world and other functions familiar in the old telephony world. In addition you get a set of new features only available in the IP world. All the functions and features you need and more: Easy installation and maintenance Provisioning Display and front panel keys Auto login and keep-alive Email notification Dynamics DNS client Quality of service Network monitoring Pre-Configured There are preset configurations for numerous broadband services. Easy IP Address Handling IP addresses are retrieved and distributed automatically via DHCP (or set manually). Only a single IP address is required from your provider. Auto Login and Keep-Alive No more frequent logging in and entering of your password! - The IX68 logs into your broadband service and keeps the connection alive. Flexible Interface Usage All interfaces can be bridged to form a secured and protected Local Area Network (LAN). One of the Ethernet ports, and the wireless interface, can be used as separate subnets or a DMZ. Network Status The status of the PCs and IP telephones on your network is monitored and clearly displayed. SIP Communication on Your LAN Live person-to-person communication, such as voice, video, presence, instant messaging and more is expected to be the next large usage of the Internet after email and web surfing! SIP is the Internet protocol for such applications. The SIP Switch - The Local IP PBX In addition to including a complete SIP server, the IX68 can also be equipped with the SIP Switch. It makes the IX68 your local PBX for soft SIP clients like Microsoft® Windows® Messenger and SIP telephones, integrating them with each other, the SIP world and the old telephone network via gateways. 353 Intertex IX68 ADSL GW2 189.95 189.95 Intertex Intertex Routers http://www.voipon.co.uk/intertex-ix68-adsl-gw2-p-353.html http://www.voipon.co.uk/images/intertex_ix68_regular.jpg new Availability: Discontinued The Intertex IX68 range is no longer available, it has been replaced by the Intertex SurfinBird IX78 ADSL AIR GW2. This can be found here The Intertex IX68 ADSL GW2 has 4 Ethernet Ports, SIP Firewall, ADSL Router and Gateway and ADSL 2+ modem The Intertex IX68 broadband router is the first in the world to allow SIP communication. It has a very general and well-proven SIP support by integrating a SIP proxy and a SIP registrar, dynamically controlling the NAT and firewall. Conventional firewalls and NAT routers cannot even be configured to handle SIP between a LAN and the Internet! Microsoft® Windows® Messenger as well as the Live Communications Server are based on SIP and are expected to rapidly bring Presence, In-stant Messaging, Voice, Video and Data Collaboration to the enterprise and everyone's desktop. The SIP Switch streamlines the usage of email-like SIP ad dresses used by soft SIP clients and numbers used by SIP phones. It routes common phone calls to PSTN gateways, does ENUM look up, can forward calls and can fork incoming calls to several phones or to voice mail. You can also get your own local numbers, dial 0 (or 9) to reach the outside world and other functions familiar in the old telephony world. In addition you get a set of new features only available in the IP world. All the functions and features you need and more: Easy installation and maintenance Provisioning Display and front panel keys Auto login and keep-alive Email notification Dynamics DNS client Quality of service Network monitoring Pre-Configured There are preset configurations for numerous broadband services. Easy IP Address Handling IP addresses are retrieved and distributed automatically via DHCP (or set manually). Only a single IP address is required from your provider. Auto Login and Keep-Alive No more frequent logging in and entering of your password! - The IX68 logs into your broadband service and keeps the connection alive. Flexible Interface Usage All interfaces can be bridged to form a secured and protected Local Area Network (LAN). One of the Ethernet ports, and the wireless interface, can be used as separate subnets or a DMZ. Network Status The status of the PCs and IP telephones on your network is monitored and clearly displayed. SIP Communication on Your LAN Live person-to-person communication, such as voice, video, presence, instant messaging and more is expected to be the next large usage of the Internet after email and web surfing! SIP is the Internet protocol for such applications. The SIP Switch - The Local IP PBX In addition to including a complete SIP server, the IX68 can also be equipped with the SIP Switch. It makes the IX68 your local PBX for soft SIP clients like Microsoft® Windows® Messenger and SIP telephones, integrating them with each other, the SIP world and the old telephone network via gateways.   354 Intertex IX68 ADSL AIR GW2 178 178.00 Intertex Intertex Routers http://www.voipon.co.uk/intertex-ix68-adsl-air-gw2-p-354.html http://www.voipon.co.uk/images/intertex_ix68.jpg new Availability: Discontinued The Intertex IX68 range is no longer available, it has been replaced by the Intertex SurfinBird IX78 ADSL AIR GW2. This can be found here The Intertex IX68 ADSL Air has 4 Ethernet Ports, SIP Firewall, ADSL Router, Wireless Gateway and ADSL 2+ modem The Intertex IX68 broadband router is the first in the world to allow SIP communication. It has a very general and well-proven SIP support by integrating a SIP proxy and a SIP registrar, dynamically controlling the NAT and firewall. Conventional firewalls and NAT routers cannot even be configured to handle SIP between a LAN and the Internet! Microsoft® Windows® Messenger as well as the Live Communications Server are based on SIP and are expected to rapidly bring Presence, In-stant Messaging, Voice, Video and Data Collaboration to the enterprise and everyone\'s desktop. The SIP Switch streamlines the usage of email-like SIP ad dresses used by soft SIP clients and numbers used by SIP phones. It routes common phone calls to PSTN gateways, does ENUM look up, can forward calls and can fork incoming calls to several phones or to voice mail. You can also get your own local numbers, dial 0 (or 9) to reach the outside world and other functions familiar in the old telephony world. In addition you get a set of new features only available in the IP world. All the functions and features you need and more: Easy installation and maintenance Provisioning Display and front panel keys Auto login and keep-alive Email notification Dynamics DNS client Quality of service Network monitoring Pre-Configured There are preset configurations for numerous broadband services. Easy IP Address Handling IP addresses are retrieved and distributed automatically via DHCP (or set manually). Only a single IP address is required from your provider. Auto Login and Keep-Alive No more frequent logging in and entering of your password! - The IX68 logs into your broadband service and keeps the connection alive. Flexible Interface Usage All interfaces can be bridged to form a secured and protected Local Area Network (LAN). One of the Ethernet ports, and the wireless interface, can be used as separate subnets or a DMZ. Network Status The status of the PCs and IP telephones on your network is monitored and clearly displayed. SIP Communication on Your LAN Live person-to-person communication, such as voice, video, presence, instant messaging and more is expected to be the next large usage of the Internet after email and web surfing! SIP is the Internet protocol for such applications. The SIP Switch - The Local IP PBX In addition to including a complete SIP server, the IX68 can also be equipped with the SIP Switch. It makes the IX68 your local PBX for soft SIP clients like Microsoft® Windows® Messenger and SIP telephones, integrating them with each other, the SIP world and the old telephone network via gateways.   355 Pirelli Discus DualPhone DP L10 124.0000 137.00 Pirelli Pirelli SIP/GSM Telephone http://www.voipon.co.uk/pirelli-discus-dualphone-dp-l10-p-355.html http://www.voipon.co.uk/images/pirelli_dualphone_dp_l10.jpg new Availability: In Stock Pirelli's DualPhone unites Wi-Fi and VoIP technology in one handset. Now you can deliver the convenience of a mobile and a broadband cordless in ONE phone. The Pirelli DiscusTM DualPhone DP-L10 is a dual-mode GSM/SIP Wi-Fi handset, combining tri-band capabilities with the support of SIP VoIP via an integrated WLAN 802.11g interface. The DualPhone works like a standard cell phone - any GSM SIM card will work and cell phone features such as SMS, MMS, e-mail, and Internet browsing are easily enabled. When at home or in office, the DualPhone automatically switches to the Wi-Fi-enabled fixed connection; making the same mobile services available indoors at a higher speed and a lower cost to you and your customers.The DiscusTM DualPhone is developed on industry-standard protocols such as 3GPP, 802.11 and SIP guaranteeing interoperability. It also features Pirelli's enhanced software that makes dual-mode technology a viable commercial solution for you. Our software also supports seamless roaming, zero-touch registration and remote manageability.The DualPhone DP-L10 is designed and qualified to operate in a multi-vendor, standards-based environment.Pirelli's SIP Wi-Fi service-oriented portfolio includes: Pirelli's AWR WLAN extender, a self-configuring Wi-Fi repeater supporting extended indoor coverage and seamless WLAN roaming Pirelli's Multiplay Access Gateways, delivering dual-mode phone QoS and service provisioning Pirelli's PMP Management Platform, a TR-069 remote management system which supports the provisioning, diagnostics and software upgrades for the entire Fixed Mobile Convergence suite (DualPhone, AWR WLAN Extender and Multiplay Access Gateways) Feature Benefits Dual Mode Handset (Wi-Fi/GSM) GSM tri-band (900/1800/1900MHz) Wi-Fi 802.11b/g SIP SMS, MMS, WAP, E-mail OTAP and Remote Management 356 Sangoma A102 PCI PRI ISDN Card 401.0000 509.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a102-pci-pri-isdn-card-p-356.html http://www.voipon.co.uk/images/sangoma_a102_pcix_small.gif new Availability: In Stock The A102 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Operational modes Data only T1/E1 and fractional T1/E1, single channel HDLC per line. Can be used as a hub for sub-DS1 remotes. The A101c and A102c can support any configuration of up to 62 DS0s carrying Frame Relay, PPP or HDLC data. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform. Channelized mode supporting individual DMA into voice timeslots plus on-board HDLC support of PRI channel for soft PBX implementations that can use these features. Mixed Voice/Data mode Combination of router/PBX functions in one server.Both 8 bit (64kbps per channel) and 7 bit (56kbps per channel) board-level HDLC support.WAN data connection is supported by Sangoma's standard WANPIPE? routing stack providing certified Frame Relay, PPP, HDLC and X.25. RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Technical Specification Available as Single T1/E1 port (A101) or Dual T1/E1 port (A102) with daughterboard (as shown in photograph). Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Power: 800mA peak, operational 300mA max at +3.3v or 5v. MTBF: > 1 Million hours. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. Certification FCC Part 15 Class A, FCC Part 68, CE. T echnical certifications in Russia, Malaysia Production quality ISO 9002 Warranty Five years parts and labour. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/ 2000/ XP, FreeBSD, Open BSD,NetBSD, Solaris Voice applications Asterisk, Yate, OPAL Open PBX/IVR Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame Diagnostic tools WANPIPEMON, SNMP, System logs 357 Sangoma A102x PCI Express PRI ISDN Card 451.6700 569.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a102x-pci-express-pri-isdn-card-p-357.html http://www.voipon.co.uk/images/sangoma_a102_pci_express_small.gif new Availability: In Stock The A102 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Operational modes Data only T1/E1 and fractional T1/E1, single channel HDLC per line. Can be used as a hub for sub-DS1 remotes. The A101c and A102c can support any configuration of up to 62 DS0s carrying Frame Relay, PPP or HDLC data. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform. Channelized mode supporting individual DMA into voice timeslots plus on-board HDLC support of PRI channel for soft PBX implementations that can use these features. Mixed Voice/Data mode Combination of router/PBX functions in one server.Both 8 bit (64kbps per channel) and 7 bit (56kbps per channel) board-level HDLC support.WAN data connection is supported by Sangoma's standard WANPIPE® routing stack providing certified Frame Relay, PPP, HDLC and X.25. RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Technical Specification Available as Single T1/E1 port (A101) or Dual T1/E1 port (A102) with daughterboard (as shown in photograph). Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Power: 800mA peak, operational 300mA max at +3.3v or 5v. MTBF: > 1 Million hours. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. Certification FCC Part 15 Class A, FCC Part 68, CE. T echnical certifications in Russia, Malaysia Production quality ISO 9002 Warranty Five years parts and labour. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/ 2000/ XP, FreeBSD, Open BSD,NetBSD, Solaris Voice applications Asterisk, Yate, OPAL Open PBX/IVR Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame Diagnostic tools WANPIPEMON, SNMP, System logs 358 Sangoma A102DX PCI Express PRI ISDN Card 685.9600 865.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a102dx-pci-express-pri-isdn-card-p-358.html http://www.voipon.co.uk/images/sangoma_a102_pci_express_small.gif new Availability: In Stock The A102DX PCIx has hardware based echo cancellation across 60 channels The A102 is Sangoma's next generation hardware designed for optimum support of data and voice over T1 and E1. Operational modes Data only T1/E1 and fractional T1/E1, single channel HDLC per line. Can be used as a hub for sub-DS1 remotes. The A101c and A102c can support any configuration of up to 62 DS0s carrying Frame Relay, PPP or HDLC data. Raw bitstream interfaces can be used to support arbitrary non standard line protocols such as non-byte aligned monosynch or bisynch. Voice modes Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. Block mode raw bit-stream interface for integration with the Asterisk Open Source PBX/IVR platform. Channelized mode supporting individual DMA into voice timeslots plus on-board HDLC support of PRI channel for soft PBX implementations that can use these features. Mixed Voice/Data mode Combination of router/PBX functions in one server.Both 8 bit (64kbps per channel) and 7 bit (56kbps per channel) board-level HDLC support.WAN data connection is supported by Sangoma's standard WANPIPE® routing stack providing certified Frame Relay, PPP, HDLC and X.25. RAIV: Receive Loss of Signal YEL: Receive Telco Yellow Alarm Line protocols Voice CAS and PRI, ATM, Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Technical Specification Available as Single T1/E1 port (A101) or Dual T1/E1 port (A102) with daughterboard (as shown in photograph). Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Power: 800mA peak, operational 300mA max at +3.3v or 5v. MTBF: > 1 Million hours. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50C. Line decoding: HDB3, AMI, B8ZS. Framing: CRC4, non-CRC4, ESF, D4T1/E1. Clocking mode: Normal, Master. Certification FCC Part 15 Class A, FCC Part 68, CE. T echnical certifications in Russia, Malaysia Production quality ISO 9002 Warranty Five years parts and labour. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/ 2000/ XP, FreeBSD, Open BSD,NetBSD, Solaris Voice applications Asterisk, Yate, OPAL Open PBX/IVR Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. T1/E1 Status alarms ALOSV: Loss of Signal alarm LOS: Receive Loss of Signal ALTLOS: Alternate loss of Signal Status OOF: Out of Frame RED: Telco Red Alarm condition AIS: Alarm Indication Signal OOSMFV: Loss of Signaling Multiframe OOCMFV: Loss of CRC Multiframe OOOFV: Out of Off-Line Frame Diagnostic tools WANPIPEMON, SNMP, System logs 360 Eicon Diva Server V-Analog-4P 671 671.00 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-vanalog4p-p-360.html http://www.voipon.co.uk/images/eicon_diva_server_v-analog-4p.jpg new Availability: In Stock Eicon Networks™ Diva Server V-Analog are high-performance, PC-based telephony adapters that provide rich media processing capabilities for either four or eight analog trunk interfaces. They are the perfect choice for enterprises looking for small size voice, speech and conferencing platforms to get started, yet allow for easy expansion. Powerful Digital Signal Processors (DSP) - one dedicated to each communication channel, ensures real-time voice processing reducing latency and improving overall system performance. Thus Diva Server V-Analog enables legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as voice portals and speech-enabled contact centers. World-class voice processing Onboard DSPs perform full set of voice processing functions Facilitates speech-enabled applications Full duplex operation and highest recognition accuracy Robust Voice over IP (VoIP) technology For integrating with emerging IP-Telephony applications Consistent Diva Server programming interface Design once - use on all! State-of-the-art operating systems supported Both Microsoft Windows as well as Linux Superior Scalability and Flexibility Up to 8 adapters can be operated concurrently in a server Easy to Install and Configure Plug and Play, no need to manually configure your server 5 year warranty, subject to product registration 361 Eicon Diva Server V-Analog-8P 868 868.00 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-vanalog8p-p-361.html http://www.voipon.co.uk/images/eicon_diva_server_v-analog-8p.jpg new Availability: In Stock Eicon Networks&trade; Diva Server V-Analog are high-performance, PC-based telephony adapters that provide rich media processing capabilities for either four or eight analog trunk interfaces. They are the perfect choice for enterprises looking for small size voice, speech and conferencing platforms to get started, yet allow for easy expansion. Powerful Digital Signal Processors (DSP) - one dedicated to each communication channel, ensures real-time voice processing reducing latency and improving overall system performance. Thus Diva Server V-Analog enables legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as voice portals and speech-enabled contact centers. World-class voice processing Onboard DSPs perform full set of voice processing functions Facilitates speech-enabled applications Full duplex operation and highest recognition accuracy Robust Voice over IP (VoIP) technology For integrating with emerging IP-Telephony applications Consistent Diva Server programming interface Design once - use on all! State-of-the-art operating systems supported Both Microsoft Windows as well as Linux Superior Scalability and Flexibility Up to 8 adapters can be operated concurrently in a server Easy to Install and Configure Plug and Play, no need to manually configure your server 5 year warranty, subject to product registration 362 Eicon Diva Server V-4PRI-120 7323.23 7323.23 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-v4pri120-p-362.html http://www.voipon.co.uk/images/eicon_diva_server_v-4pri.jpg new Availability: In Stock The Eicon Networks Diva Server V-4PRI board is an exceptionally powerful PC-based telephony adapter that provides rich media processing capabilities for up to 120 voice channels over E1 interfaces or up to 96 voice channels over T1 interfaces. It is the perfect choice for enterprises and service providers looking for a large size voice, speech and conferencing platform that can easily be expanded to multiple adapters per system. Extremely Powerful Digital Signal Processors (DSP) &ndash; one dedicated to six communication channels, ensures real-time voice processing reducing latency and improving overall system performance. Thus the Diva Server V-4PRI enables both legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as; voice portals and speech-enabled contact centers. Up to 120 voice channels using four E1 or four ISDN PRI interfaces Up to 96 voice channels using four T1 interfaces World-class media processing &ndash; voice activity detection, DTMF and tone handling, echo cancellation Efficient integration with speech (ASR and TTS) engines Enhanced Switching and Conferencing support Robust Voice over IP (VoIP) technology built in Powerful onboard RISC CPU and 20 DSPs ensure high performance, offloading server processing Extensive, open and well documented API Tightly integrated with most recent Microsoft Windows 2000, XP and Server 2003 Full Linux support - including the latest SuSE, RedHat and Debian versions, as well as via source level RPM and DEB Ease of installation guaranteed by Plug and Play conformance Easy upgrade with free software downloads from Eicon 5 year warranty, subject to product registration 363 Eicon Diva Server V-2PRI-60 5039.73 5039.73 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-v2pri60-p-363.html http://www.voipon.co.uk/images/eicon_diva_server_v-2pri.jpg new Availability: In Stock The Eicon Networks Diva Server V-4PRI board is an exceptionally powerful PC-based telephony adapter that provides rich media processing capabilities for up to 120 voice channels over E1 interfaces or up to 96 voice channels over T1 interfaces. It is the perfect choice for enterprises and service providers looking for a large size voice, speech and conferencing platform that can easily be expanded to multiple adapters per system. Extremely Powerful Digital Signal Processors (DSP) - one dedicated to six communication channels, ensures real-time voice processing reducing latency and improving overall system performance. Thus the Diva Server V-4PRI enables both legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as; voice portals and speech-enabled contact centers. Up to 60 voice channels using four E1 or four ISDN PRI interfaces Up to 48 voice channels using four T1 interfaces World-class media processing - voice activity detection, DTMF and tone handling, echo cancellation Efficient integration with speech (ASR and TTS) engines Enhanced Switching and Conferencing support Robust Voice over IP (VoIP) technology built in Powerful onboard RISC CPU and 10 DSPs ensure high performance, offloading server processing Extensive, open and well documented API Tightly integrated with most recent Microsoft Windows 2000, XP and Server 2003 Full Linux support - including the latest SuSE, RedHat and Debian versions, as well as via source level RPM and DEB Ease of installation guaranteed by Plug and Play conformance Easy upgrade with free software downloads from Eicon 5 year warranty, subject to product registration 364 Eicon Diva Server V-PRI E1/30 3159.3 3159.30 Eicon Eicon BRI Cards http://www.voipon.co.uk/eicon-diva-server-vpri-e130-p-364.html http://www.voipon.co.uk/images/eicon_server_v-pri_e1_30.jpg new Availability: In Stock Eicon Networks&rsquo; Diva Server V-PRI/E1 boards are exceptionally powerful, PC-based telephony adapters that provide rich media processing capabilities for up to 30 voice channels over E1 or ISDN PRI interface. It is the perfect choice for enterprises and service providers looking for a large size voice, speech and conferencing platform that can easily be expanded to multiple adapters per system. Powerful Digital Signal Processors (DSP) &ndash; one dedicated to each communication channel, ensures real-time voice processing reducing latency and improving overall system performance. Thus Diva Server V-PRI/E1 enables both legacy and IP-based computer telephony (CTI) solutions, as well as leading edge voice business applications, such as; voice portals and speech-enabled contact centers. Up to 30 voice channels over E1 or ISDN PRI interface World-class media processing &ndash; voice activity detection, DTMF and tone handling, echo cancellation Efficient integration with speech (ASR and TTS) engines Enhanced Switching and Conferencing support Robust Voice over IP (VoIP) technology built in Powerful onboard CPU and DSPs ensure high performance, offloading server processing Extensive, open and well documented API Tightly integrated with most recent Microsoft Windows 2000, XP and Server 2003 Full Linux support - latest SuSE and RedHat versions Ease of installation guaranteed by Plug and Play conformance Easy upgrade with free software downloads from Eicon 5 year warranty, subject to product registration 387 Draytek Vigor2800VG 176 176.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2800vg-p-387.html http://www.voipon.co.uk/images/draytek_2800G.jpg new Availability: Discontinued The Draytek 2800 series has been discontinued, and replaced by the 2820 series. The replacement for the 2800VG is the Draytek Vigor 2820Vn . Draytek Vigor 2800VG VPN ADSL 2/2+ 108Mbps Wireless Router w/Voice Over IP The DrayTek Vigor2800VG is a combined Voice Over IP ADSL router for Internet access, firewall, VPN device and Ethernet switch. The DrayTek Vigor Vigor2800VG includes features such as VLAN, QoS, Content filtering, VPN support for up to 32 tunnels and a USB printer port and support for all current ADSL technologies. Targeting requirement for residential, SOHO (Small office and Home Office) and business users, the DrayTek Vigor 2800VG is an ADSL2/2+ enabled integrated access device. With downstream speed up to 12Mbps (ADSL2) or 24Mbps (ADSL2+). The DrayTek Vigor 2800VG provides exceptional bandwidth* for internet access. (* note: the available bandwidth also depends on the Internet Service Provider) Selectable QoS Assurance The DrayTek Vigor 2800VG supports selectable QoS (Quality of Service). This enables you to select specific protocols/services to have guaranteed levels of your Internet bandwidth. For example, if you need POP3 email to have priority, you could specify that 50% of your download (or upload, or both) bandwidth is reserved for POP3 when required. When it's not being used by POP3, the bandwidth is available for all other regular traffic. The DrayTek Vigor2800VG is very flexible on QoS - you can set several groups of services to have different priorities and bandwidth reservations. Voice Over IP Voice-over IP (VoIP), is one of the hottest communications technologies being adapted for transmission over broadband technologies. If you regularly make long-distance phone calls, chances are you've already used IP telephony without even knowing it. VoIP is the transmission of telephone calls over a data network like one of the many networks that make up the Internet. While you may of heard of VoIP, what you may not know is that many traditional telephone companies are already using it in the connections between their regional offices. The DrayTek 2800VG allows a simple, cost-effective solution to take advantage of this exicting technology. A typical DrayTek 2800VG solution is below. The calls between the two sites above are free of charge, making use of your existing always-on ADSL connection, but cost isn't the only advantage; using VOIP means that you have additional call capacity in your home or office, without tying up your regular phone line. You can also make calls to regular phone lines, via a PSTN gateway. Enhanced Wireless Security The DrayTek Vigor2800VG built-in Wireless Access Point enables wireless connectivity for wireless clients supporting the 802.11g wireless protocol, providing a total wireless bandwidth of up to 54Mb/s. Twin extra-gain aerials ensuring maximum coverage range and signal diversity The DrayTek Vigor 2800VG supports industry standard WEP and WPA/TKIP and WPA2/AES (802.11i) Encryption methods for encrypting your wireless LAN traffic. If you prefer, the Vigor2800VG can use "VPN over WLAN" to increase the level of wireless encryption, using DES/3DES encryption. You can lock the router down further so that only PCs whose unique hardware ('MAC') address is authorised can access the router. "SSID Stealthing" makes your base more difficult to other people to scan for. The Vigor2800G also supports 802.1x Authentication to allow a permitted wireless user database on your Radius server. To enhance the security and performance of the wireless network, the DrayTek Vigor2800VG feature wireless LAN isolation, WDS (Wireless Distribution System), and Universal VLAN. Vigor2800VG Features : ADSL2+ Router & Firewall Compatible with ADSL, ADSL2, ADSL2+ (up to 24Mb/s) Easy to use user interface for setup and control Four-Port Ethernet autosensing 10/100BaseT Switch VPN Server for up to 16 simultaneous tunnels Stateful Packet Inspection for NAT and Fully routed connections Internet URL Content Filtering (Blacklist or Whitelist) Parental Control with Surfcontrol® Block IM & P2P Applications QoS Assurance for mission-critical applications/services Time Scheduling for Internet Access Virtual LAN (VLAN) - Segment Ethernet ports into distinct common/isolated groups Bandwidth Throttling - Restrict speed on each Ethernet port USB Printer Port - share regular printer centrally SNMPv2, Syslog, uPNP & Radius Support Voice-over-IP Features: Twin VoIP Telephone ports Connect any standard telephone (cordless or corded) SIP Compliant Voice codecs supported: 8Kb/s-64Kb/s Multiple Simultaneous SIP server registrations Wireless LAN Features: Super-G, 802.11g and 802.11b Wireless LAN Access Up to 108Mb/s Wireless LAN total bandwdith Several independent levels of Wireless Security WEP, WPA & WPA2 Wireless Encryption Wireless VLAN - Isolate WLAN from Wired LAN     388 Linksys SPA-922 Single Line SIP Phone 64.3000 66.90 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa922-single-line-sip-phone-p-388.html http://www.voipon.co.uk/images/linksys_spa922.jpg new Availability: In Stock Stylish and functional in design, the SPA-922 IP phone is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large-scale IP Centrex deployment. The SPA922 leverages industry leading VoIP technology from Linksys to deliver an upgradeable high quality IP phone that is unparalleled in features, value, and support. Standard features on the SPA-922 include dual switched Ethernet ports, 802.3af PoE, a high resolution graphical display, speakerphone and a 2.5 mm head-set port. The SPA922 supports one line with two call appearances and provides support for three way conferencing, attended call transfer, and placing a call on hold to answer an incoming call. The line can be configured as a unique phone number (or extension), or can be configured to share a number that is assigned to multiple phones. Full featured one-line business class IP phone supporting Power over Ethernet 802.3af Connect directly to an Internet Telephone Service Provider or connect to an IP PBX Dual switched Ethernet ports, Speakerphone, Caller ID, Call Hold, Conferencing, and more Easy installation with secure remote provisioning. Menu based and web based configuration. Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the SPA-922 has been tested to ensure comprehensive interoperability with equipment enabling service providers to quickly roll-out competitive, feature rich services to their customers. With hundreds of features and configurable sevice parameters, the SPA922 addresses the requirements of traditional business users while leveraging the advantages of IP telephony. Features such as easy station moves, presence, and shared line appearances (across local and geographically dispersed locations) are just some of the many advantages of the SPA-922. Carrier-Grade Security, Provisioning, and Management The SPA922 uses standard encryption protocols to provide secure remote provisioning and unobtrusive in-service software upgrades. Linksys secure remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high quality support to their subscribers. Remote provisioning also saves service providers the hassle and expense of managing, preloading, and re-configuring customer premise equipment (CPE). Package Contents SPA-922 IP Phone, Handset, and Stand Handset Cord RJ45 Ethernet Cable Quick Installation Guide Power Supply supplied separately Telephony Features One Voice Line with Two Call Appearances Backlit Pixel Based Display: 128x64 Monochrome Graphical Liquid Crystal Display (LCD) Line Status - Active Line Indication, Name and Number Menu Driven User Interface Shared Line Appearance ** Speakerphone Call Hold for the SPA922 Music on Hold ** Call Waiting Caller ID Name and Number and Outbound Caller ID Blocking Outbound Caller ID Blocking Call Transfer - Attended and Blind Three Way Call Conferencing with Local Mixing Connects to External Conference Bridge for Multi-party Conferencing Automatic Redial of Last Calling and Last Called Numbers On-Hook Dialing Call Pick Up - Selective and Group ** Call Park and UnPark **Call Swap Call Back on Busy Call Blocking - Anonymous and Selective Call Forwarding - Unconditional, No Answer, On Busy Hot Line and Warm Line Automatic Calling Call Logs (60 entries each): Made, Answered, and Missed Calls Redial from Call Logs Personal Directory with Auto-dial (100 entries) Do Not Disturb (callers hear line busy tone) Digits Dialed with Number Auto-Completion Anonymous Caller Blocking URI (IP) Dialing Support (Vanity Numbers) On Hook Default Audio Configuration (Speakerphone and Headset) Multiple Ring Tones with Selectable Ring Tone per Line Called Number with Directory Name Matching Call Number using Name - Directory Matching or via Caller ID Subsequent Incoming Calls with Calling Name and Number Date and Time with Intelligent Daylight Savings Support Call Duration and Start Time Stored in Call Logs Call Timer Name and Identity (Text) Displayed at Start UpDistinctive Ringing Based on Calling and Called Number Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com Speed Dialing, Eight Entries Configurable Dial/Numbering Plan Support Intercom ** Group Paging ** NAT Traversal, including STUN Support DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy Syslog, Debug, Report Generation, and Event Logging Secure Call Encrypted Voice Communication Support Built-in Web Server for Administration and Configuration with Multiple Security Levels for the SPA922 Automated Remote Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP) Optionally Require Admin Password to Reset Unit to Factory Defaults ** Feature requires support by call server Hardware Features Pixel Based Display: 128x64 Monochrome LCD Graphical Display with back light Dedicated Illuminated Buttons for: Audio Mute On/Off Headset On/Off Speakerphone On/Off Four Way Rocking Directional Knob for Menu Navigation Voice Mail Message Waiting Indicator Light Voice Mail Message Retrieval Button Dedicated Hold Button Settings Button for Access to Feature, Set-up, and Configuration Menus Volume Control Rocking Up/Down Knob Controls Handset, Headset, Speaker, Ringer Standard 12-Button Dialing Pad High Quality Handset and Cradle Built-In High Quality Microphone and Speaker Headset Jack - 2.5 millimeter LED Test Function Two Ethernet LAN Ports with Integrated Ethernet Switch - 100BaseT RJ-45 802.3af Compliant Power over Ethernet (PoE) Data Networking MAC Address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) ARP - Address Resolution Protocol DNS - A Record (RFC 1706), SRV Record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) I CMP - Internet Control Message Protocol (RFC792) TCP - Transmission Control Protocol (RFC793) UDP - User Datagram Protocol (RFC768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349) VLAN Tagging 802.1p/q - Layer 2 QoS SNTP - Simple Network Time Protocol (RFC 2030) Voice Gateway SIPv2 - Session Initiation Protocol Version 2 (RFC 3261, 3262, 3263, 3264) SIP Proxy Redundancy - Dynamic via DNS SRV, A Records Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks - NAT (including STUN) SIPFrag (RFC 3420) for the SPA922 Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP Codec Name Assignment Voice Algorithms: -G.711 (A-law and &#956;-law) -G.726 (16/24/32/40 kbps) -G.729 A -G.723.1 (6.3 kbps, 5.3 kbps) Dynamic Payload Support Adjustable Audio Frames Per Packet DTMF: In-band and Out-of-Band (RFC 2833) (SIP INFO) Flexible Dial Plan Support with Inter-Digit Timers IP Address / URI Dialing Support Call Progress Tone Generation Jitter Buffer - Adaptive Frame Loss Concealment VAD - Voice Activity Detection with Silence Suppression Attenuation / Gain Adjustments MWI - Message Waiting Indicator Tones VMWI - Voice Mail Waiting Indicator - Via NOTIFY, SUBSCRIBE Caller ID Support (Name and Number) Third Party Call Control (RFC 3725) Provisioning, Administration & Maintenance Integrated Web Server Provides Web Based Administration and Configuration Telephone Key Pad Configuration via Display Menu / Navigation Automated Provisioning and Upgrade via HTTPS, HTTP, TFTP Asynchronous Notification of Upgrade Availability via NOTIFY Non-intrusive, In-Service Upgrades Report Generation and Event Logging Statistics Transmitted in BYE Message Syslog and Debug Server Records - Configurable Per Line Physical Interfaces 2 100baseT RJ-45 Ethernet Port (IEEE 802.3) Handset: RJ-7 Connector for the SPA922 Built-in Speakerphone and Microphone Headset 2.5 mm Port The Linksys SPA 922 requires a Power adapter if your network does not support PoE (power over ethernet). Please select the appropriate option from the drop-down menu below.   Linksys SPA-Series VoIP Telephone (SIP) Comparison Table SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernet Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 389 Atcom AT-530 IAX IP Telephone 42.0000 46.00 Atcom Atcom IAX IP Telephone http://www.voipon.co.uk/atcom-at530-iax-ip-telephone-p-389.html http://www.voipon.co.uk/images/atcom_at-530.jpg new Availability: In Stock AT530 series IP phone is an internet based voice network phone terminal supporting power supply through Ethernet. AT530 series IP phone adopts multiple voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service. AT530 IP phone supports SIP and IAX2 protocol, offering two 10/100Mbps Ethernet interface with built in router. AT-530 can accommodate both internet access and telephone connection on a single line, thus effectively using the existing broadband resources. It is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service. AT530 IP phone adopts the latest Ethernet power supply technology to not only save user investments, simplify network deployment, but also provide centralized power management.(accessory option). Key features: Support two sip servers running at the same time. Redundancy sip server capable. NAT, Firewall.DHCP client and server. Support PPPoE, (used for ADSL, cable modem connecting). Support major G7.xxx CODEC. VAD,CNG. G.165 compliant 16ms echo cancellation E.164 dial plan and customized dial rules Hotline. Call Forward, Call Transfer, 3-way conference calls Call ID display DND(Do Not Disturb),Black List,Limit List Data Features: Static/Dynamic WAN-IP-Addressing PPPoE Management: Web, telnet and keypad management. Adjustable user password and super password Upgrade firmware through HTTP, FTP or TFTP. Telnet remote management. Upload/download setting file Auto-provision. Safe mode provide reliability Phone book, maximum 100 entries. Atcom AT-530 Interface: Two RJ45 ports, one for WAN, one LAN. Power port 390 Grandstream GXW-4004 Analog FXS Gateway 102.0000 114.00 Grandstream Grandstream Gateways http://www.voipon.co.uk/grandstream-gxw4004-analog-fxs-gateway-p-390.html http://www.voipon.co.uk/images/GXW-4004 small.jpg new Availability: In stock The GXW Analog FXS IP Gateway series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXS analog gateways options. The GXW FXS series, comes in three different port configurations: 4 port , 8 port , and 24 port . Designed and tested for full interoperability with leading IP-PBXs, Softswitches and most SIP-based environments, the GXW technology ensures manageability, a simple configuration, superb voice quality and feature rich functionality. The GXW Analog IP Gateway series is based on open industry standards. Main Benefits: Simple and flexible configuration options as a VoIP-FXS Gateway IP enabler for analogue phones, faxes and legacy PBX systems Secure and easy management using Web browser, automated provisioning tools, and integrated IVR High quality voice and fax communication with significant cost savings More Information on GXW-4004: 4 FXS ports with PSTN lifeline Two 10M/100Mbps network ports (switched or routed) Multiple SIP server profiles (2 per system) and independent account per port Supports audio codecs: G.711, G.723, G.726, G.729 A/B/E, G.728, iLBC T.38 Fax G.168 Echo Cancellation 391 Grandstream GXW-4008 Analog FXS Gateway 155.0000 179.00 Grandstream Grandstream Gateways http://www.voipon.co.uk/grandstream-gxw4008-analog-fxs-gateway-p-391.html http://www.voipon.co.uk/images/GXW-4008 small.JPG new Availability: In stock The GXW Analog FXS IP Gateway series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXS analog gateways options. The GXW FXS series, comes in three different port configurations: 4 port , 8 port , and 24 port . Designed and tested for full interoperability with leading IP-PBXs, Softswitches and most SIP-based environments, the GXW technology ensures manageability, a simple configuration, superb voice quality and feature rich functionality. The GXW Analog IP Gateway series is based on open industry standards. Main Benefits: Simple and flexible configuration options as a VoIP-FXS Gateway IP enabler for analogue phones, faxes and legacy PBX systems Secure and easy management using Web browser, automated provisioning tools, and integrated IVR High quality voice and fax communication with significant cost savings More Information on GXW-4008: 8 FXS ports with PSTN lifeline Two 10M/100Mbps network ports (switched or routed) Multiple SIP server profiles (2 per system) and independent account per port Supports audio codecs: G.711, G.723, G.726, G.729 A/B/E, G.728, iLBC T.38 Fax G.168 Echo Cancellation 392 Sangoma A200 FXO FXS Analogue Card PCI 75.0000 95.00 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-p-392.html http://www.voipon.co.uk/images/sangoma_a200_pcix_small.jpg new Availability: In Stock PCI Type: PCIx Sangoma's A200 4 port FXO/FXS card delivers superior audio qualities and is expandable to 24 ports in a 2U form factor, with optional carrier-grade echo cancellation. As you need them, additional REMORA cards can be added to the base four port A200 card. A single PCI slot hosts connection for up to 24 ports and ensures common synchronous clocking for all channels. The A200 AFT architecture is shared with Sangoma's A101, A102, A104 and A108 cards ensuring common 3.3v/5v, high performance and universal PCI compatibility. Like all the Sangoma AFT Series, the A200 and REMORA system has field upgradeable firmware to take advantage of enhancements as they become available. Optionally, the A200 supports Sangoma's echo cancellation and voice enhancement DSP daughterboard for carrier grade echo cancellation and voice enhancement. Architecture The A200 consists of a REMORA daughterboard mounted on the AFT PCI card. The REMORA card has two sockets each which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS lines respectively. Up to five additional REMORA daughterboards can be mounted in empty slot positions beside the A200 assembly, connected to the A200 by a backplane bus connector. Wiring Connections The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2m cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other. For those who need to hard wire the A200 system, Sangoma provides a kit of 12 RJ11/4 plugs and a crimping tool. Technical specifications From 2 to 24 ports supported, mixing FXO and FXS interfaces as required. Support for the Asterisk, Yate, FreeSwitch, OPAL, PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Single synchronous PCI interface for all 24 FXO/FXS ports. Four RJ11 ports per REMORA card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. Short 2U compatible mounting clips available for installation in 2U rack-mount servers. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5v and 3.3v PCI busses. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features can be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Optional DSP daughterboard on the A200d G.168-2002 echo cancellation in hardware 1024 taps/128ms tail per channel on all channel densities DTMF decoding and tone recognition Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2m cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other. For those who need to hard wire the A200 system, Sangoma provides a kit of 12 RJ11/4 plugs and a crimping tool. 393 Sangoma A200 FXO FXS Analogue Card PCI Express 101.0000 130.00 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html http://www.voipon.co.uk/images/sangoma_a200_pci_express_small.jpg new Availability: In Stock PCI Type: PCI Express Sangoma's A200 4 port FXO/FXS card delivers superior audio qualities and is expandable to 24 ports in a 2U form factor, with optional carrier-grade echo cancellation. As you need them, additional REMORA cards can be added to the base four port A200 card. A single PCI slot hosts connection for up to 24 ports and ensures common synchronous clocking for all channels. The A200 AFT architecture is shared with Sangoma's A101, A102, A104 and A108 cards ensuring common 3.3v/5v, high performance and universal PCI compatibility. Like all the Sangoma AFT Series, the A200 and REMORA system has field upgradeable firmware to take advantage of enhancements as they become available. Optionally, the A200 supports Sangoma's echo cancellation and voice enhancement DSP daughterboard for carrier grade echo cancellation and voice enhancement. Architecture The A200 consists of a REMORA daughterboard mounted on the AFT PCI card. The REMORA card has two sockets each which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS lines respectively. Up to five additional REMORA daughterboards can be mounted in empty slot positions beside the A200 assembly, connected to the A200 by a backplane bus connector. Wiring Connections The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2m cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other. For those who need to hard wire the A200 system, Sangoma provides a kit of 12 RJ11/4 plugs and a crimping tool. Technical specifications From 2 to 24 ports supported, mixing FXO and FXS interfaces as required. Support for the Asterisk, Yate, FreeSwitch, OPAL, PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Single synchronous PCI interface for all 24 FXO/FXS ports. Four RJ11 ports per REMORA card. Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. Short 2U compatible mounting clips available for installation in 2U rack-mount servers. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5v and 3.3v PCI busses. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features can be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Optional DSP daughterboard on the A200d G.168-2002 echo cancellation in hardware 1024 taps/128ms tail per channel on all channel densities DTMF decoding and tone recognition Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. The A200 and Remora cards incorporate four, 4 pin RJ11 narrow jacks such as used in telephone handsets. Each A200/Remora is shipped with four 2m cables terminating in a narrow RJ11/4 plug at one end and a telephone-standard RJ11/6 plug at the other. For those who need to hard wire the A200 system, Sangoma provides a kit of 12 RJ11/4 plugs and a crimping tool. 394 Sangoma A400 FXO FXS Analogue Card PCI 132.0000 166.00 Sangoma Sangoma A400 Remora http://www.voipon.co.uk/sangoma-a400-fxo-fxs-analogue-card-pci-p-394.html http://www.voipon.co.uk/images/sangoma_a400_pcix_small.jpg new Availability: In Stock PCI Type: PCIx The A400 and REMORA system support up to 48 FXO/FXS ports The A400 Series is the high density version of the popular A200 series analog cards. Identical in operation and configuration, and using the same FXO and FXS modules, the A400 system supports twelve ports per main board and REMORA, as compared to four ports for the A200. As you need them, additional REMORA cards can be added to the base twelve port A400 card. A single PCI or PCI Express slot host connection can support up to 48 FXO/FXS ports with common synchronous clocking for all channels. Like all the Sangoma AFT Series, the A400 and REMORA system has field upgradeable firmware to take advantage of enhancements as they become available. Optionally, the A400 supports Sangoma's echo cancellation and voice enhancement DSP daughterboard for carrier grade echo cancellation a n d voice enhancement. Architecture The A400 consists of an A400 REMORA daughterboard mounted on the AFT PCI /PCI Express. The A400 REMORA card has six sockets each which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS lines respectively. Up to four additional A400 REMORA daughterboards can be mounted in empty slot positions beside the A400 assembly, connected to the A400 by a backplane bus connector. Each 12 port A400 REMORA card is connected by means of a standard 12 line color coded telephone cable terminating at the card in a robust DB25 connector, and ready for hard wiring into a punch block at for the PSTN connection. Technical specifications From 2 to 48 ports supported, mixing FXO and FXS interfaces as required. Support for the Asterisk, Yate, FreeSwitch, OPAL, PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Single synchronous PCI interface up to 48 FXO/FXS ports. Connection by means of a standard 2 line colour coded telephone cable for connection to a punch block. Dimensions: 2U height form factor: 290mm x 55mm for use in a 2U chassis. Short 2U compatible mounting clips available for installation in 2U rackmount servers. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5v and 3.3v PCI busses. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper nterrupts sharing. Intelligent hardware:Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Optional DSP daughterboard on the A400d G.168-2002 echo cancellation in hardware. 1024 taps/128ms tail per channel on all channel densities. DTMF decoding and tone recognition. Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. 395 Sangoma A400 FXO FXS Analogue Card PCI Express 158.0000 200.00 Sangoma Sangoma A400 Remora http://www.voipon.co.uk/sangoma-a400-fxo-fxs-analogue-card-pci-express-p-395.html http://www.voipon.co.uk/images/sangoma_a400_pcix_express_small.jpg new Availability: In Stock PCI Type: PCI Express The A400 and REMORA system support up to 48 FXO/FXS ports The A400 Series is the high density version of the popular A200 series analog cards. Identical in operation and configuration, and using the same FXO and FXS modules, the A400 system supports twelve ports per main board and REMORA, as compared to four ports for the A200. As you need them, additional REMORA cards can be added to the base twelve port A400 card. A single PCI or PCI Express slot host connection can support up to 48 FXO/FXS ports with common synchronous clocking for all channels. Like all the Sangoma AFT Series, the A400 and REMORA system has field upgradeable firmware to take advantage of enhancements as they become available. Optionally, the A400 supports Sangoma\'s echo cancellation and voice enhancement DSP daughterboard for carrier grade echo cancellation a n d voice enhancement. Architecture The A400 consists of an A400 REMORA daughterboard mounted on the AFT PCI /PCI Express. The A400 REMORA card has six sockets each which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS lines respectively. Up to four additional A400 REMORA daughterboards can be mounted in empty slot positions beside the A400 assembly, connected to the A400 by a backplane bus connector. Each 12 port A400 REMORA card is connected by means of a standard 12 line color coded telephone cable terminating at the card in a robust DB25 connector, and ready for hard wiring into a punch block at for the PSTN connection. Technical specifications From 2 to 48 ports supported, mixing FXO and FXS interfaces as required. Support for the Asterisk, Yate, FreeSwitch, OPAL, PBX/IVR projects, as well as other Open Source and proprietary PBX/Switch/IVR/VoIP gateway applications. Single synchronous PCI interface up to 48 FXO/FXS ports. Connection by means of a standard 2 line colour coded telephone cable for connection to a punch block. Dimensions: 2U height form factor: 290mm x 55mm for use in a 2U chassis. Short 2U compatible mounting clips available for installation in 2U rackmount servers. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5v and 3.3v PCI busses. Fully PCI 2.2 compliant, compatible with all commercially available motherboards, proper nterrupts sharing. Intelligent hardware:Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradeable so that new features related to voice and/or data can be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Temperature range: 0 - 50C. Optional DSP daughterboard on the A400d G.168-2002 echo cancellation in hardware. 1024 taps/128ms tail per channel on all channel densities. DTMF decoding and tone recognition. Voice quality enhancement: Octasic music protection, acoustic echo control and adaptive noise reduction. 396 Sangoma A200-S FXS Module (2 Ports) 39.0000 53.00 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200s-fxs-module-2-ports-p-396.html http://www.voipon.co.uk/images/sangoma_a200-O.jpg new Availability: In Stock Sangoma A200 FXS module; 2 ports Two port FXS module for Sangoma A200 and REMORA series range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic. A200 REMORA cards have two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively. 397 Sangoma A200-O FXO Module (2 Ports) 43.0000 59.00 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200o-fxo-module-2-ports-p-397.html http://www.voipon.co.uk/images/sangoma_a200-s.jpg new Availability: In Stock Sangoma A200 FXO module; 2 ports Two port FXS module for Sangoma A200 and REMORA series range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic. A200 REMORA cards have two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively. 398 Sangoma A400-O FXO Module (2 Ports) 43.0000 59.00 Sangoma Sangoma A400 Remora http://www.voipon.co.uk/sangoma-a400o-fxo-module-2-ports-p-398.html http://www.voipon.co.uk/images/sangoma_a200-s.jpg new Availability: In Stock Sangoma A400 FXO module: 2 ports Two port FXS module for Sangoma A400 and REMORA series range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic. A400 REMORA cards have two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively. 399 Sangoma A400-S FXS Module (2 Ports) 39.0000 53.00 Sangoma Sangoma A400 Remora http://www.voipon.co.uk/sangoma-a400s-fxs-module-2-ports-p-399.html http://www.voipon.co.uk/images/sangoma_a200-O.jpg new Availability: In Stock Sangoma A400 FXS module: 2 ports Two port FXS module for Sangoma A400 and REMORA series range of Advanced, Flexible Telecommunications (AFT) hardware designed for optimum support of analog voice traffic. A400 REMORA cards have two sockets each of which can accept an FXO-2 or FXS-2 module. Each FXO-2 or FXS-2 module supports two FXO or FXS ports respectively. 400 Sangoma A200RABP2 Slave Card 36.0000 48.00 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200rabp2-slave-card-p-400.html http://www.voipon.co.uk/images/sangoma_a200rabp2.jpg new Availability: In Stock Sangoma A200 Remora Slave Card Sangoma's A200 4 port FXO/FXS card delivers superior audio qualities and is expandable to 24 ports in a 2U form factor, with optional carrier-grade echo cancellation. As you need them, additional REMORA cards can be added to the base four port A200 card. A single PCI slot hosts connection for up to 24 ports and ensures common synchronous clocking for all channels.   The Sangoma A200 card supports upto 5 slave cards, providing 24 ports (12 modules).   401 Sangoma A400RABP2 Slave Card 110.0000 140.00 Sangoma Sangoma A400 Remora http://www.voipon.co.uk/sangoma-a400rabp2-slave-card-p-401.html http://www.voipon.co.uk/images/sangoma_a400rabp2.jpg new Availability: In Stock The A400 Series is the high density version of the popular A200 series analog cards. Identical in operation and configuration, and using the same FXO and FXS modules, the A400 system supports twelve ports per main board and REMORA, as compared to four ports for the A200. As you need them, additional REMORA cards can be added to the base twelve port A400 card. A single PCI or PCI Express slot host connection can support up to 48 FXO/FXS ports with common synchronous clocking for all channels. Be aware, you can add a maximum of 3 slave cards to the existing Sangoma Card providing a total of 24 modules (48 ports) Each slave card supports 6 modules (12 ports).     402 DrayTEK Vigor 2100VG 85 85.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor-2100vg-p-402.html http://www.voipon.co.uk/images/draytek-2100v-small.jpg new Draytek Vigor 2100VG Wireless Broadband Router with Voice over IP The DrayTek Vigor 2100VG is an broadband firewall/router - ideal for sharing your Cable Modem between several PCs, with a built-in VoIP (Voice over IP) telephone port. Even if you have just one PC, the router provides the firewall facilities. The phone port enables you to use your existing broadband connection to make VoIP calls to any other compatible device, anywhere in the world and anywhere on the Internet, free of charge. You can receive calls too of course, and all with a standard (analogue) telephone which you connect into the socket on the back of the router. Having the Vigor2100VG on your broadband connection effectively gives you an extra phone line - with no extra line rental, so when family or co-workers are tying up the main line you can still make calls using the phone on your Vigor2100VG - ideal for teleworkers who need that extra flexibility. The Vigor2100VG is visually pleasing too; its curved lines and brushed silver finish enable it to fit into any environment stylishly. Automatic QoS Assurance Traffic levels from your local users to the Internet can vary; if someone else on your router is downloading, that will affect your speeds. Mostly that's quite tolerable - you just get your data a little slower but you wouldn't normally notice. With VoIP, things are different. A voice call has to be digitised, transmitted to the remote end and the turned back into an analogue waveform (sound!) in real-time. If part of a packet is delayed then the sound becomes jerky and intermittent which at best is annoying and in worst cases intolerable. The Vigor2100VG firstly uses efficient codecs designed to make the best use of available bandwidth, but secondly includes automatic QoS Assurance. QoS (Quality of Service) Assurance reserves part of your Internet bandwidth for voice calls whilst a voice call is active (the reserved bandwidth is available for regular use if there is no Voice call active). This means that, regardless of what else other people are doing on your Network, you will always have the necessary inbound and outbound bandwidth reserved exclusively for Voice. Voice Over IP Using VoIP is simple. You connect any standard telephone to one of the two phone ports on the unit. In the simplest usage mode, you lift the handset and dial the IP address of the remote unit ( * is used in place of the normal dot). The router then contacts the remote router and the phone connected at the other site then rings. The remote user lifts his handset to answer the call and the two parties can then talk to each other. More conveniently, you use the phone book to dial remote users with a short code and a SIP registar/proxy to locate users automatically without needing to know their IP address. Vigor 2100VG Main Features : Broadband Router & Firewall for cable-modems Ideal for use with Broadband from NTL or Telewest Blueyonder Ideal for the SoHo user (Small Office/Home) 802.11g and 802.11b Wireless LAN Access (Vigor2100VG only) Enables you to connect wireless computers & laptops around your home Several independent levels of Wireless Security (Vigor2100VG only) Voice-over-IP Faciliy using your normal telephone PSTN Passthrough for your regular phone line Waiting POP3 Email notifcation by front panel LED Four port 10/100BaseT Switch (expandable) Internet Content Filtering Automatic QoS (Quality of Service) Assurance for voice calls Phone ports compatible with any standard telephone VoIP Voice calls are carried over existing ADSL connection Automatic Fallback to your analogue line during ADSL failure VPN Passthrough for VPN client/server running behind the router Automatic support for popular multimedia applications including Netmeeting & MSN Messenger for multiple LAN users. Dynamic DNS Posting, compatible with popular services DNS Proxy/Cache & DHCP Server     403 Draytek Vigor 2700 ADSL2 Router 85 85.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor-2700-adsl2-router-p-403.html http://www.voipon.co.uk/images/draytek-2700-small.jpg new Draytek Vigor 2700 ADSL2 + Router The DrayTek Vigor 2700 ADSL2+ Router is to be the optimum ADSL and firewall sharing device for Small office and Home (SoHo) user. The Vigor2700 router perfectly combines quality of service with ease of use. Crammed full of advanced features, this attractively packaged router offers great value for money. The Vigor2700 router has direct ADSL support, compatible with ADSL, ADSL2 and ADSL2+ for line speeds up to 24Mb/s. ADSL2+ boasts increased speed (and higher speed over a given distance) than original ADSL. Feature Highlights: Built-in ADSL modem - plugs directly into your ADSL-enabled line Compatible with ADSL, ADSL2, ADSL2+ up to 24Mb/s downstream Universal Plug'n'Play (uPnP) Firewall - Help protect your PCs from external attacks NAT Port redirection, forwarding and IP-DMZ VPN Passthrough for VPN client/server running behind the router 4-Port 10/100BaseT Ethernet Switch (with automatic uplink detection) for PC/Mac connection Automatic support for popular multimedia applications including MSN Messenger Dynamic DNS Posting, compatible with popular DDNS providers Easy configuration, monitoring & control from web-interface DNS Proxy/Cache & DHCP Server   Specifications: ADSL Interface compliant with ADSL, ADSL2, ADSL2+ Auto-Rate Negotiation (512Kb/s-24Mb/s Downstream) Up to 1Mb/s Upstream Comprehensive Routing Facilities including Internet Protocols : PPP (RFC 1661), PPPoA (RFC2364) LAN Protocols : ARP, Proxy ARP, IP, ICMP, IGMP, UDP, TCP Authentication Protocols : PAP, CHAP, MS-CHAP DNS Proxy & Cache Universal Plug'n'Play (uPnP) support Firewall Facilities Keepstate packet monitoring Protection against DoS/DDos attacks & IP Anti-Spoofing Default Port Blocking through NAT User definable packet filtering Built-in Ethernet switch featuring Four port 10/100BaseT auto-selecting (IEEE802.3) Automatic selection of full/half duplex Automatic selection of regular/uplink mode (no more finding the right cable!) Auto detects connection to another switch or hub Dual-colour LED for each port to indicate operating speed NAT (Network Address Translation) featuring Port forwarding - individual ports (with optional translation) and ranges of ports Open internal servers up for external access DMZ - For passthrough of all protocols/data to a single internal client DHCP Server - User configurable Easy to use Web-based User Interface, management with optional IP address restriction. Management Logging/Monitoring via Syslog Web based interface Telnet Interface Flash-Upgradable memory for firmware upgrades/enhancements by TFTP (utility included) Support for local Class C network (up to 253 local users) or smaller subnets. Physical Characteristics Metallic Blue ABS Case Flash Upgradable firmware (across LAN using DrayTek TFTP utility) Dimensions (mm) : L 218 x W 158 x H 36 Weight : 359 grams approx Maximum Power Consumption : 10W approx Client O/S Compatibility : Windows 95, 98, ME, NT4, 2000, XP, Linux & MacOS8/9/X Warranty : Three (3) year manufacturer's RTB warranty Box contents : Router, UK PSU, Ethernet cable (1.8M), ADSL RJ11 Line cable (1.8M), Quickstart manual, Utility & Documentation CD-ROM, Wall-mounting Screws       404 Draytek Vigor 2700VG ADSL2 + VoIP Router 120 120.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor-2700vg-adsl2-voip-router-p-404.html http://www.voipon.co.uk/images/DRAYTEK-2700VG-SMALL.jpg new Availability: In Stock The DrayTek Vigor 2700 Series ADSL routers are designed to be the ideal ADSL-sharing and firewall device for to the SoHo (Small Office / Home) user. They combine performance, features and price into an attractive, relible and easy-to-use product, with the pedigree and thoughtful design DrayTek are renowned for. Vigor 2700VG Highlights Built-in ADSL modem - plugs directly into your ADSL-enabled line Compatible with ADSL, ADSL2, ADSL2+ up to 24Mb/s downstream Universal Plug'n'Play (uPnP) Firewall - Help protect your PCs from external attacks NAT Port redirection, forwarding and IP-DMZ VPN Passthrough for VPN client/server running behind the router 4-Port 10/100BaseT Ethernet Switch (with automatic uplink detection) for PC/Mac connection Automatic support for popular multimedia applications including MSN Messenger Dynamic DNS Posting, compatible with popular DDNS providers Easy configuration, monitoring & control from web-interface DNS Proxy/Cache & DHCP Server Wireless LAN Interface (Vigor2700G and Vigor2700VG only): 802.11b & 802.11g Compatibility WEP & WPA Wireless Security & Encryption Voice-over-IP Facilities (Vigor2700V & Vigor2700VG only): Phone socket for any regular telephone SIP-compliant Analogue Phone Line integration, switching & failover. Make voice calls free-of-charge to any other compatible VoIP user, worldwide Make low cost calls to any phone worldwide at low rates Receive calls on your own real telephone number ADSL Compatibility The Vigor2700 router series has direct ADSL support, compatible with ADSL, ADSL2 and ADSL2+ for line speeds up to 24Mb/s. The Vigor2700 series is compatible with all standard ISPs in the UK/NI and Channel Islands. ADSL2+ gives increased speed (and higher speed over a given distance) than original ADSL. VoIP Capability ('V' Models only) The Vigor2700VG model provides Voice-over-IP capability. A phone port on the rear allows you to connect any regular telephone (cordless or corded) to make and receive voice calls using your Internet connection. Pre-set 'QoS assurance' automatically gives priority to VoIP traffic so that call quality is maximised from your available Internet bandwidth. Selectable codecs allow you to tune your ideal VoIP parameters for best call quality and bandwidth use. Vigor 2700 Specification ADSL Interface compliant with : ADSL, ADSL2, ADSL2+ Auto-Rate Negotiation (512Kb/s-24Mb/s Downstream). Up to 1Mb/s Upstream. Comprehensive Routing Facilities including : Internet Protocols : PPP (RFC 1661), PPPoA (RFC2364) LAN Protocols : ARP, Proxy ARP, IP, ICMP, IGMP, UDP, TCP Authentication Protocols : PAP, CHAP, MS-CHAP DNS Proxy & Cache Universal Plug'n'Play (uPnP) support Firewall Facilities : Keepstate packet monitoring Protection against DoS/DDos attacks & IP Anti-Spoofing Default Port Blocking through NAT User definable packet filtering Built-in Ethernet switch featuring : Four port 10/100BaseT auto-selecting (IEEE802.3) Automatic selection of full/half duplex Automatic selection of regular/uplink mode (no more finding the right cable!) Auto detects connection to another switch or hub Dual-colour LED for each port to indicate operating speed NAT (Network Address Translation) featuring : Port forwarding - individual ports (with optional translation) and ranges of ports Open internal servers up for external access DMZ - For passthrough of all protocols/data to a single internal client DHCP Server - User configurable Easy to use Web-based User Interface, management with optional IP address restriction. Management Logging/Monitoring via : Syslog Web (Browser) based interface Telnet Interface Flash-Upgradable memory for firmware upgrades/enhancements by TFTP (utility included) Support for local Class C network (up to 253 local users) or smaller subnets. Physical Characteristics: Metallic Blue ABS Case Flash Upgradable firmware (across LAN using DrayTek TFTP utility) Dimensions (mm) : L 218 x W 158 x H 36 Weight : 359 grams approx Maximum Power Consumption : 10W approx Client O/S Compatibility : Windows 95, 98, ME, NT4, 2000, XP, Linux & MacOS8/9/X Warranty : Three (3) year manufacturer's RTB warranty Box contents : Router, UK PSU, Ethernet cable (1.8M), ADSL RJ11 Line cable (1.8M), Quickstart manual, Utility & Documentation CD-ROM, Wall-mounting crews, Aerial ('G' models only) 406 Vegastream Vega 50 Europa Gateway 4 x FXO 320.0000 354.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-4-x-fxo-p-406.html http://www.voipon.co.uk/images/vega_50_europa_4xfxo.png new Availability: Late October The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 4 Port FXO BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80") x 44.45mm (1.75") x 237.06mm (9.33") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 407 Vegastream Vega 50 Europa Gateway 8 x FXO 589.0000 654.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-8-x-fxo-p-407.html http://www.voipon.co.uk/images/vega_50_europa_8xfxo.png new Availability: Late October The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 8 Port FXO BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80") x 44.45mm (1.75") x 237.06mm (9.33") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 408 Vegastream Vega 50 Europa Gateway 4 x FXS 2 x FXO 290.0000 324.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-4-x-fxs-2-x-fxo-p-408.html http://www.voipon.co.uk/images/vega_50_europa_4xfxs_2xfxo.png new Availability: In stock The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 4 Port FXS + 2 x FXO BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80\") x 44.45mm (1.75\") x 237.06mm (9.33\") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 409 Vegastream Vega 50 Europa Gateway 8 x FXS 2 x FXO 534.0000 594.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-8-x-fxs-2-x-fxo-p-409.html http://www.voipon.co.uk/images/vega_50_europa_8xfxs_2xfxo.png new Availability: In stock The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 8 Port FXS + 2 x FXO BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80") x 44.45mm (1.75") x 237.06mm (9.33") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 410 Vegastream Vega 50 Europa Gateway 2 x BRI 4 Channels 410.0000 499.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-2-x-bri-4-channels-p-410.html http://www.voipon.co.uk/images/vega_50_europa_2xbri_4_channels.png new Availability: In stock The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 2 Port BRI (4 channels) BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80") x 44.45mm (1.75") x 237.06mm (9.33") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 411 Vegastream Vega 50 Europa Gateway 4 x BRI 8 Channels 550.0000 695.00 Vegastream Vegastream Vega 50 Europa http://www.voipon.co.uk/vegastream-vega-50-europa-gateway-4-x-bri-8-channels-p-411.html http://www.voipon.co.uk/images/vega_50_europa_4xbri_8_channels.png new Availability: In stock The Vega 50 Europa gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, ISDN, analog phonesand the PSTN to IP networks. The Vega 50 Europa supports up to 10 analog ports, and either 4 or 8 basic rate ISDN channels on 4 interfaces. The Europa is available factory configured in a number of different configurations. The following gateway is configured as follows: 4 Port BRI (8 channels) BRI interfaces can each be independently configured as network orterminal-facing. The Vega 50 Europa gateway can, therefore, beconnected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing The Vega 50 Europa can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 Europa supports ETSI BRI and has proveninteroperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Lifeline PSTN Backup All variants equipped with any number of FXS ports are also fitted with two FXO ports. The two FXO ports provide a hard-wired bypassto allow local PSTN calls to be made under power or IP failure conditions Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.711 (a-law/µ-law) (64 kbps) G.729a (8kbps) G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 8 VoIP channels Telephony Interface ETSI BRI 2 or 4 powered S/T interfaces Point to point or point to multipoint Each Interface can be configured NT or TE 8 or 4 FXS ports 600R, 900R or CTR-21 line impedance for the 2 backup FXO ports fitted in FXS variant. Features Identification Caller ID presentation Caller ID screening guarantees connection only from authenticated call sources H.323 gatekeeper registration> SIP Registration and Digest Authentication SIP Registration and Digest Authentication Billing, Operations and Maintenance HTTP web server - with access to user guide Radius Accounting Remote firmware upgrade: - Auto code upgrade - Auto configuration upgrade SNMP MIB 1&2 TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) Call Quality Adaptive jitter removal Comfort noise generation Silence suppression> 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) NAT Traversal Environmental 0°-40°C 0-90% humidity (non-condensing) Front Panel Display LED: POWER LAN: Link, Activity Call Activity Physical Dimensions 299.73mm (11.80") x 44.45mm (1.75") x 237.06mm (9.33") width/height/depth Weight: 1.02kgs (unit only) Power External 30W AC-DC adapter (TBD) Program Storage Code and configuration data are stored in FLASH and executed from RAM 412 DrayTek Vigor 2800i 166 166.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor-2800i-p-412.html http://www.voipon.co.uk/images/2800I-SMALL.jpg new Availability: Discontinued The Draytek 2800 series has been discontinued, and replaced by the 2820 series. The 2820 series can be found here . DrayTek Vigor 2800i Firewall VPN ADSL 2 Router The Vigor2800i model offers all of the same facilities as the standard Vigor2800 model but has an ISDN interface in addition to ADSL. This can connect to any ISDN2e or BT Highway/Midband line. The ISDN interface provides dial-backup in the event of your main ADSL connection being interrupted. Alternatively, the ISDN interface can be used on its own (as your main Internet connection) if you do not have an ADSL feed to connect to the Vigor2800i. When operating on ISDN, the Vigor2800i can be used both for shared Internet access and direct-dial ISDN LAN-to-LAN Wide Area Networking (dial in teleworkers or dial-in/out branch links) at the same time (one function on each B channel). Feature Highlights: Easy Internet-sharing via ADSL, ADSL 2 connection Robust firewall to help protect your network from external attacks Powerful VPN facilities provide deployment of linked branch offices and teleworkers Comprehensive business features such as VLAN, Bandwidth Control, Advanced Content Filtering Wireless LAN data rate up to 108Mbps with Super G technology   Routing Features Internet Protocols : PPPoA, PPPoE Static routing and Dynamic Routing with RIP v1 and v2 DHCP Server, DHCP Client, DHCP Relay, DHCP over IPSec DNS Cache/Proxy Configurable MTU Size Dynamic DNS Firewall and Security Features Stateful Packet Inspection Firewall Packet filtering based on port, source IP address, destination IP address, MAC address, (ICMP/TCP/UDP) DoS, DDos attacks prevention (IP Spoofing, Land Attack, Smurf Attack, Ping of Death, TCP SYN flooding) E-mail alert and logging via syslog Disable Firewall Option Content Filtering Instand Messenger/P2P blocking URL key word blocking Java applet, cookies, Active X, compressed, executable, multimedia file blocking Time schedule VPN (Virtual Private Network) Features Support up to 32 simultaneous tunnels High-performance IPSec 3DES encryption MPPE, DES (56-bit) and 3DES (168-bit) encryption, AES (256-bit) encryption Authenication MD5 & SHA-1 Diffie-Hellman Group 768-bit & 1024-bit Key Management: Auto Internet Key Exchange (IKE) w/ Perfect Forward Secrecy (PFS) PKI (X.509) certificate support Operation Mode: Main & Aggressive Single-session Virtual Private Network (VPN) Pass-through (IPSec, L2TP), PPTP Radius Support for dial-in teleworker profiles NAT (Network Address Translation) Features Simultanous Non-NAT & NAT mode Many-to-One (NAT) Many-to-Many (Multi-NAT) Full Routing (Non-NAT) ISDN Features Compatible with ISDN2e, BT's Home/Business Highway & BT Midband® lines Uses ISDN for shared Internet access (dial-on-demand) Support for 64Kb/s and 128Kb/s (Multilink-PPP) Automatic ISDN backup for Internet access during ADSL failure Bandwidth-on-demand (automatically switches between 64Kb/s and 128Kb/s) Direct ISDN Dial-up LAN-to-LAN connectivity (to another ISDN site) Remote 'teleworker' direct dial-in access to your LAN (from a remote ISDN line) Remote activation of ISP dial-up (dials ISP on receipt of recognised Caller ID) Applications & Gaming Features Port Forwarding UPnP Support Single DMZ Support Management Features Web Based Interface (HTTP/HTTPS) Simple Installation Wizard CLI (Command Line Interface, Telnet/SHH) System performance and status monitoring Built-in Diagnostic Function Administration access control Remote Management via Web services Syslog Support Configuration backup/restore Firmware upgrade via TFTP/FTP SNMP management MIB-II Support Bandwidth Control and Quality of Service Wireless rate-control Wired & Wireless VLAN Support 4 VLANs Class-based bandwidth guarantee by user-defined traffic categories Support of four priority-levels Support of DiffServ Codepoint classifying Printer Server USB Printer Server USB Port Version 1.1 Compatible with most printers with a USB Port Unidirectional Only LAN Ports 4 RJ-45 10/100 Ethernet switch Auto Sensing / Manual Selection Auto Uplink WAN Ports RJ-11 ADSL Port Built-in ADSL 2/2+ Modem G.dmt: 8Mbps downstream, 832Kbps upstream G.lite: 1.5Mbps downstream, 512Kbps upstream ADSL2: 12Mbps downstream, 1Mbps upstream ADSL2+: 24Mbps downstream, 1Mbps upstream ISDN Bri Port Package Contents Draytek Vigor 2800i Firewall VPN ADSL 2 Router w/ISDN Backup UK Power adapter RJ11 Telephone Cable RJ45 3m Ethernet cable Installation guide     413 Linksys SPA-962 IP Telephone 131.0000 137.90 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-spa962-ip-telephone-p-413.html http://www.voipon.co.uk/images/spa-962-small.jpg new Availability: In Stock A Highly Functional Business VoIP Phone. The SPA-962 VoIP telephone is a must for businesses using a hosted IP telephony service, an IP PBX, or a large scale IP Centrex deployment. The SPA962 utilizes industry leading VoIP technology from Linksys to deliver a high quality IP Phone that is unparalleled in features, value, and support. Up to Six Configurable Phone Lines Some of the many features on the SPA962 include six active lines, dual switched Ethernet ports, 802.3af PoE support and a high resolution colour display. Each line can be independently configured to use a unique phone number (or extension), or can be configured to use a shared number that is assigned to multiple phones. Interoperable The SPA962 has been tested to ensure comprehensive interoperability with equipment from VoIP infrastructure. With hundreds of features and configurable service parameters, the SPA962 addresses the requirements of traditional business whilst utilizing the advantages of IP Telephony. Technical Data Key features continued: On-Hook Dialing Call Pick Up - Selective and Group Call Park and UnPark Call Swap Call Back on Busy Call Blocking - Anonymous and Selective Call Forwarding - Unconditional, No Answer, On Busy Hot Line and Warm Line Automatic Calling Call Logs (60 entries each): Made, Answered,and Missed Calls Redial from Call Logs Personal Directory with Auto-dial (100 entries) Do Not Disturb (callers hear line busy tone) URI (IP) Dialing Support (Vanity Numbers) On Hook Default Audio Configuration (Speaker phone and Headset) Multiple Ring Tones with Selectable Ring Tone per Line Called Number with Directory Name Matching Call Number using Name - Directory Matching or via Caller ID Subsequent Incoming Calls with Calling Name and Number Date and Time with Intelligent Daylight Savings Support Call Duration and Start Time Stored in Call Logs Call Timer Name and Identity (Text) Displayed at Start Up Distinctive Ringing Based on Calling and Called Number Ten User Downloadable Ring Tones - Ring Tone Generator Free from www.linksys.com Speed Dialling Configurable Dial/Numbering Plan Support - per Line Intercom ** Group Paging ** DNS SRV and Multiple A Records for Proxy Lookup and Proxy Redundancy Syslog, Debug, Report Generation, and Event Logging Secure Call Encrypted Voice Communication Support - SIP over TLS, and SRTP Built-in Web Server for Administration and Configuration with Multiple Security Levels Automated Provisioning, Multiple Methods. Up to 256 Bit Encryption: (HTTP, HTTPS, TFTP) Optionally Require Admin Password to Reset Unit to factory Defaults ** Feature requires support by SIP server Hardware: 320 x 240 True colour, Four inch, Liquid Crystal Display (LCD) Four Illuminated Call Appearance Line buttons with tricolour LED's LED Indicates Line State - Active, Idle, On-Hold, Unregistered Line LED Configurable to 13 Different States (On/off, Colour, Flash) Dedicated Illuminated buttons for - Audio Mute On/off Headset On/Off Speaker phone On/Off Four Soft Key Buttons Four way Rocking Directional Knob for Menu Navigation Support for up to two attendant Consoles; adds up to 64 programmable buttons Voice Mail Message Waiting Indicator Light Voice Mail retrieval button Dedicated Hold Settings button for Access to Feature, Set-up, and configuration Menus Volume Control Rocking Up/Down Controls Handset, speaker, Ringer Standard 12-Button Dialling Pad Headset Jack - 2.5 millimetre LED Test Function Two Ethernet LAN Ports with Integrated Ethernet Switch - 100BaseT RJ-45 802.3af Compliant Power over Ethernet (PoE) Optional 5 volt DC Universal (100-240 Volt) (ORDERED SEPARATELY) Regulatory Compliance & Security: FCC, CE, Class B Canadian ICES-003, A-Tick Certification Password Protected System, pre-set to factory default Password Protected Access to Administrator and User level Features HTTPS with Factory Installed Client Certificate HTTP Digest - Encrypted Authentication via MD5 (RFC 1321) Up to 256-bit Encryption Documentation: Quick-Start Installation and Configuration Guide User Guide Administration Guide Package contents: 1 - SPA-962 IP Phone, Handset and Stand 1 - Handset Cord 1 - RJ45 Ethernet Cable 1 - Quick installation Guide Environmental: Dimensions: 8" x 7.63" x 7.50" (203 x 194 x 191 mm) W x H x D Unit weight: 2.4lbs (1.088 kg) Operating Temp: 41?~113?F (5?~45?C) Storage Temp: 10~90% Non-Condensing Storage Temp: 10-90% Non-Condensing Linksys SPA Series VoIP Telephone (SIP) Comparison Table SPA Model Voice Lines Ethernet Ports High Resolution Display Power Over Ethernet Mains Power Supply SPA901 1 1 N N Y SPA921 1 1 Y N Y SPA922 1 2 Y Y N SPA941 2-4 1 Y N Y SPA942 2-4 2 Y Y N SPA962 6 2 Y Colour Y N 414 Atcom AT-530 IAX IP Telephone POE Version 44.0000 49.00 Atcom Atcom IAX IP Telephone http://www.voipon.co.uk/atcom-at530-iax-ip-telephone-poe-version-p-414.html http://www.voipon.co.uk/images/AT-530-POE-SMALL.jpg new Availability: In Stock The AT-530 with POE from ATCOM is a powerful yet affordable SIP/IAX2 IP phone. Superb voice quality along with the ability to connect two SIP servers simultaneously, built in firewall and many other features including built in answer phone service and supports sidetone. Key features: Supports a maximum of two sip servers running at the same time. Redundant sip service capable - Can automatically switch to secondary service if the primary one fails. Answer phone (Can record three messages each at 90 seconds in length) NAT, Firewall. WAN and LAN ports can operate as two independent networks, or in bridge mode (both network cards on the same LAN) DHCP client and server. Supports PPPoE, (used for ADSL, cable modem connecting). Support major G7.xxx CODEC. VAD,CNG. G.165 compliant 16ms echo cancellation E.164 dial plan and customized dial rules Hotline (If configured will auto dial a predetermined number when off hook state is determined) Do not disturb Auto answer Ban all outgoing calls Call Forward Call Transfer 3-way conference calls Call ID display DND (Do Not Disturb) Black List (Block specific incoming numbers from calling) Limit List (Block specific outgoing numbers from being dialled) Mute Handsfree Speed dial (9 keys on phone) VPN using L2TP and UDP Sidetone support Data Features: Static/Dynamic WAN-IP-Addressing PPPoE Management: Web, telnet and keypad management. Adjustable user password and super password Upgrade firmware through HTTP, FTP or TFTP Telnet remote management and diagnostics Upload/download setting file Auto-provisioning Safe mode feature for phone recovery and diagnostics Phone book with maximum 100 entries. Security Built in Firewall MMI Feature (Limit devices with the ability to connect to the phone) Interface Two RJ45 ports, one for WAN, one LAN (bridge mode available to ensure both NICs can be on the same LAN) Power port       415 Snom 370 IP Telephone 168.0000 179.00 Snom Snom IP Telephones http://www.voipon.co.uk/snom-370-ip-telephone-p-415.html http://www.voipon.co.uk/images/snom_370_small.jpg new Availability - In stock The Snom 370 has a large graphical, high-definition display and offers a much improved and additional presentation of call-lists, caller information and address-books. Call information can be customised with ease via XML to show the information the user wants displayed. Via an in built mini browser, a user has direct access to their own applications from a display screen. As well as the user being able to customise the design of the display they can also view news-tickers and other useful information, access central and public phone directories. The snom 370 provides many audio devices at the same time, e.g., a user can use the headset, handset and loud speaker concurrently. The Snom 370 IP telephone offers with its increased memory capacity in addition to all the needed office functionality such as status indicator, choice of trunk line, group lines, transfer, call-pickup or conferencing (3-way conference bridge) more flexibility for individual functions and programs. The Snom's increased memory capacity will also enable the display of graphics and high resolution images to depict the status of contacts (e.g., busy, off-line, on-line) similar to Instant-Messenger. The Snom 370 can also play music and media files. Ease of use, high audio quality, surpassed security and interoperability make the snom 370 highly suitable for private users, SME's, home offices, and ISP applications. Technical Specifications: - Tilt able, high-def graphical display (240x128 Pixel) - Large LED (red) for incoming calls - 47 keys, 13 LEDs - 12 programmable function keys - Speaker phone - Power over Ethernet (PoE) - Dual-Ethernet connectivity - Support for several audio devices - Additional keypads with 42 programmable function keys - Codecs: G.711, G.729A, G.726, G.723.1, G.722, GSM 6.10 (Full-rate) - Security: SIPS/SRTP, TLS - SIP RFC3261 - STUN, NAT, ENUM, ICE - National Language Support Snom IP Phone Comparison Table Phone Power over ethernet Display SIP Identities Programmable Keys with LEDs Large LED for incoming calls Extension Keyboard Available Wireless headset adaptor Snom 300 Y 2 Line 16 Characters 4 6 N N N Snom 320 Y Hinged 2 line character display with graphical field 12 12 N Y Y Snom 360 Y Hinged, backlit graphical display (128x64 pixels) 12 12 N Y Y Snom 370 Y Hinged, backlit, high-definition graphical display (240x128 pixels) 12 12 Y Y Y 416 Siemens S450IP Dect SIP Phone 83.0000 87.50 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-s450ip-dect-sip-phone-p-416.html http://www.voipon.co.uk/images/siemens_s450ip.jpg new Availability: In stock The Gigaset S450 IP puts everything at your fingertips. Switch between making internet calls and fixed-line calls - no PC required! In addition to plug-and-play simplicity, this VoIP-ready phone tells you if your friends are online and even notifies you if you've received any e-mails. The Gigaset S450 IP proves that cutting-edge technology can also be brilliantly easy to use. Without the need for a PC, you simply plug-in the Gigaset S450 IP to your phone connection and in a few steps you can start phoning over the internet to your friends and family around the world. Conveniently select the internet telephony provider you prefer on the handset's illuminated display. No matter what, you're assured Gigaset's Quality of Service (QoS) for superb internet telephony quality. VoIP doesn't get any easier than that. Highlights * Dual mode: Easy switch from VoIP calls to fixed line calls by single keypress * 2 internet calls in parallel or 1 fixed-line call and 1 VoIP call in parallel with multiple handsets * Multiline: assign up to 4 VoIP numbers* for multihandset use * E-mail notification with time, date and subject* S450 IP Features * Load contacts from PC (Outlook) * Instant Messaging (buddy list, chatting, presence status) * Supports key messengers (e.g. MSN, Yahoo, AOL) * Easy configuration of internet telephony without a PC * Increased virus protection thanks to protected operating system * CLIP* function displays caller's name * SMS* with up to 640 characters * Phonebook for up to 150 entries     Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features All units supplied as base and 1 handset - Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. All units can connect to both a standard BT phone line and have an ethernet connection for VoIP calling. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 418 Aastra 53i IP Phone 81.0000 88.00 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-53i-ip-phone-p-418.html http://www.voipon.co.uk/images/aastra_53i_sml.jpg new Availability: In stock Great Value and Quality in an Entry-level Business IP Telephone The Aastra 53i offers powerful features and flexibility in a standards-based, carrier-grade business IP phone. With a sleek design and 3 line LCD display, the 53i is fully interoperable with leading PBXs, offering advanced XML capability to access custom applications with support for up to 9 calls simultaneously. Part of the Aastra family of IP phones, the 53i is perfectly suited for all business telephone requirements. Key Features Enhanced Call Management With extensive capacity for personal directories, redial lists and call logs, the Aastra 53i can improve efficiency by providing more call information at the touch of a button. Features such as shared call and bridged-line-appearances, call forwarding, call transfer, call waiting, intercom and local 3-way conference increase call flexibility and control. XML Browser The Aastra 53i incorporates XML browser capabilities, allowing access to customised services and applications. Development guides available from Aastra enable you to create custom internal service applications. This feature provides unlimited potential to customise the Aastra 53i to meet the specific needs of your systems and business using the display and keypad. Excellent Audio Quality All Aastra IP phones offer full-duplex speakerphone with excellent voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product. Simplified Deployment & Implementation From initial installation and configuration to future enhancements and upgrades, Aastra IP telephones are designed to save your business time and money. Dual auto-sensing switched Ethernet ports eliminate the need for additional wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet simplifies deployment with centralised power and backup. Configuration files, easily created using any text editor, can be used to configure phones individually or centrally. Feature Highlights Up to 9 call appearance lines Shared call and bridged line appearances XML browser Multi-proxy support Distinctive ringing, priority alerting Personal directory (200 entries) Call forward/Call transfer/Call waiting Caller and calling line information Callers log (200 entries) Redial list (last 100 numbers) Local 3-way conference Intercom with auto-answer Do Not Disturb Live dial pad or pre-dial support Digium's Asterisk Business Edition - supports Busy Lamp Field (BLF), Bridged Line Appearances (BLA), Intercom BroadSoft's BroadWorks enhanced SIP - supports Shared Call, Appearances, Busy Lamp Field (BLF), CommPilot Call Manager Nortel's MCS 5100/5200 SIP supports Conferencing, Voicemail, and NAT Traversal Sylantro's SIP - supports Bridged Line Appearances (BLA), ComCierge, Call Park/Pickup, Conference 419 Aastra 55i IP Telephone 109.0000 121.85 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-55i-ip-telephone-p-419.html http://www.voipon.co.uk/images/aastra_55i_sml.jpg new Availability: In stock High Quality, Cost Effective Mid-level Expandable, Business IP Phone The Aastra 55i offers powerful features and versatility in a standards-based, carrier-grade expandable business IP phone. With a sleek design, 144 x 75 pixel backlit LCD display and 6 dynamic context-sensitive softkeys, the 55i is fully interoperable with leading IP-telephony platforms.The 55i incorporates advanced XML capability to access custom applications and support for up to 9 calls simultaneously. The 55i is extremely well suited to business telephone requirements and users who require more one-touch feature keys and XML based programs. Key Features   Enhanced Call Management With extensive storage capacity for personal directories, call logs and redial lists and 12 programmable keys, the Aastra 55i can improve efficiency by providing more call information at the press of a button. Features such as shared call and bridged line appearances, call forward, call transfer, call waiting, intercom and local 3-way conference increase call flexibility and control. XML Browser The Aastra 55i incorporates XML browser capabilities and a large display with dynamic softkeys to easily access customised services and applications. It is possible to create internal service applications using development guides available from Aastra. This provides unlimited potential to customise the Aastra 55i to meet your specific business needs and application requirements using the display and keypad. Expandability The Aastra 55i supports up to 3 Aastra 536M modules, each offering 36 keys with LED indicators to create a feature-rich attendant console. Excellent Audio Quality All Aastra voIP phones offer full-duplex speakerphone capabilities with exceptional voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product. Simplified Deployment & Implementation From initial implentation and configuration to future enhancements and upgrades, Aastra IP phones are designed to save your business time and money. Dual auto-sensing switched Ethernet ports remove the need for additional wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet enables easy deployment with centralised power and backup. Configuration files can be easily created using any text editor to configure phones individually or centrally. Feature Highlights Shared call and bridged line appearances Up to 9 call appearance lines Multi-proxy support Distinctive ringing, priority alerting XML browser Call forward/Call transfer/Call waiting Personal directory (200 entries) Caller and calling line information Callers log (200 entries) Local 3-way conference Intercom with auto-answer Do Not Disturb Redial list (last 100 numbers) Live dial pad or pre-dial support Digium's Asterisk Business Edition - supports Busy Lamp Field (BLF), Bridged Line Appearances (BLA), Intercom BroadSoft's BroadWorks enhanced SIP - supports Shared Call, Appearances, Busy Lamp Field (BLF), CommPilot Call Manager N ortel's MCS 5100/5200 SIP supports Conferencing, Voicemail, and NAT Traversal Sylantro's SIP - supports Bridged Line Appearances (BLA), ComCierge, Call Park/Pickup, Conference 420 Aastra 57i IP Phone 122.0000 145.00 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html http://www.voipon.co.uk/images/aastra_57i_sml.jpg new Availability: In stock The Aastra 57i - Amazing Value & Quality in a Fully-Featured, Expandable, Business IP Phone The Aastra 57i phone offers many powerful features and great flexibility in a standards based, carrier-grade fully featured expandable business voIP telephone. With a sleek, elegant design, large 144 x 128 pixel backlit graphical LCD display and 6 dynamic context-sensitive softkeys, the 57i is fully interoperable with leading IP-telephony platforms, offering advanced XML capability to access bespoke applications with support for up to 9 calls simultaneously. The Aastra 57i IP phone is ideally suited for executives and business telephone users who require more one touch feature keys and a large screen to fully utilise XML based programs. Key Features XML Browser The Aastra 57i is equipped with XML browser capabilities and an extra-large display with dynamic softkeys to quickly access customised services and applications. Internal service applications can be created using development guides available from Aastra; this provides unlimited potential to customise the Aastra 57i to meet specific business needs using the display and keypad. Advanced Call Management With extensive capacity for personal directories, call logs, redial lists and 12 programmable keys, the Aastra 57i can increase efficiency by providing more call information at the touch of a button. Features such as shared call and bridged line appearances, call forward, call waiting, call transfer, intercom and local 3-way conference increase call flexibility and control. Expandability The Aastra 57i has two different module extension options. It supports up to 3 Aastra 536M modules, each offering 36 keys with LED indicators to create a feature rich attendant console. Alternatively, it will also support up to three of the more advanced Aastra 560M modules, each offering 60 keys with a screen based LCD display and LED system. Superior Audio Quality All Aastra IP telephones offer full-duplex speakerphone capabilities with excellent voice clarity. With years of experience in telecoms and telephone design, Aastra can ensure superior voice quality on every product. Simplified Deployment & Implementation From initial deployment and configuration to subsequent enhancements and upgrades, the Aastra family of IP phones is designed to save businesses time and money. Dual auto-sensing switched Ethernet ports eliminate extra wiring and simplify installations. Integrated IEEE 802.3af Power-over-Ethernet allows easy deployment with centralised power and backup. Easily created configuration files, using any text editor, can be used to configure phones centrally or individually. Feature Highlights Up to 9 call appearance lines Shared call and bridged line appearances XML browser Personal directory (200 entries) Multi-proxy support Distinctive ringing, priority alerting Call forward/Call transfer/Call waiting Caller and calling line information Callers log (200 entries) Intercom with auto-answer Local 3-way conference Redial list (last 100 numbers) Do Not Disturb Live dial pad or pre-dial support Digium's Asterisk Business Edition - supports Busy Lamp Field (BLF), Bridged Line Appearances (BLA), Intercom BroadSoft's BroadWorks enhanced SIP - supports Shared Call, Appearances, Busy Lamp Field (BLF), CommPilot Call Manager Nortel's MCS 5100/5200 SIP supports Conferencing, Voicemail, and NAT Traversal Sylantro's SIP - supports Bridged Line Appearances (BLA), ComCierge, Call Park/Pickup, Conference 421 Sangoma A200 Backplane (2 Connector, 8 ports) 8.51 8.51 Sangoma Sangoma A200 Remora http://www.voipon.co.uk/sangoma-a200-backplane-2-connector-8-ports-p-421.html http://www.voipon.co.uk/images/sangoma_a200_bp2.jpg new Availability: In Stock Connector bridges are available in several sizes: 2,3,4,5 and 6 slots. 422 Linksys SPA400 Analogue Telephone Adapter 158.95 158.95 Linksys Linksys IP PBX http://www.voipon.co.uk/linksys-spa400-analogue-telephone-adapter-p-422.html http://www.voipon.co.uk/images/spa400.jpg new Availability: In Stock Internet Telephony Gateway with 4 FXO Ports Advanced Multi-Port PSTN Gateway Solution for the Linksys Voice System Functions as an Analogue Line Gateway for a Linksys Voice System VoIP Network Integrated Voicemail Application Server with up to 32 Voicemail Accounts Perfectly Suited to Connect up to 4 Analogue Lines Enables Linksys Voice System Users to Leave and Playback Voicemail Messages Designed only to be deployed with the Linksys SPA9000 Easy installation support using the SPA9000 Setup Wizard The SPA400 features the ability to connect up to four (4) standard analogue telephones lines to a Linksys Voice System (LVS) VoIP network and includes the additional benefit of an integrated voicemail application. The SPA400 utilizes multiple analogue phone lines and automatically routes calls to and from your existing PSTN telephone service. Designed to be implemented with the LVS IP Telephony System, the SPA400 enables cost-conscience business users to leverage the high value features generally found only on more expensive PBX equipment. The SPA400 also includes an integrated voicemail application supporting up to 32 voicemail accounts or boxes with customized greetings, providing LVS users the ability to receive and playback voicemail messages from their Linksys IP Phones. The SPA400 supports four (4) RJ-11 FXO ports to connect to the PSTN and also includes one 10/100 BaseT RJ-45 Ethernet interface that is used to connect to the local IP network. Compact in design, the SPA400 is perfectly suited for a small business in providing reliable operation for both VoIP and PSTN voice communication service. Installed by the service provider, reseller, or end user, the SPA400 enables transparent PSTN gateway functionality for a VoIP telephony network and is a key element in building a solid business communications system. Linksys SPA Series Analogue Telephone Adapter (ATA) Comparison Chart Model Service Lines Active Calls 3-Way Call Conferences Network Ports PSTN (FXO) Ports Phone (FXS) Ports SPA1001 2 2 1 1 0 1 PAP2T 2 4 2 1 0 2 SPA2102 2 4 2 2 0 2 SPA3102 2 3 1 2 1 1 SPA400 N/A N/A N/A 1 4 0 423 Digium TDM802B 2 Port FXO Module 191.1000 213.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm802b-2-port-fxo-module-p-423.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 424 Digium TDM801B 1 Port FXO Module 145.7000 167.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm801b-1-port-fxo-module-p-424.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 425 Digium TDM803B 3 Port FXO Module 236.4000 259.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm803b-3-port-fxo-module-p-425.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 426 Digium TDM8S4B 4 Port FXO Module 282.2000 305.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm8s4b-4-port-fxo-module-p-426.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 427 Digium TDM805B 1 Quad Port FXO & 1 Port FXO Module 216.1000 318.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm805b-1-quad-port-fxo-1-port-fxo-module-p-427.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 428 Digium TDM806B 1 Quad Port FXO & 2 Port FXO Module 305.7000 364.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm806b-1-quad-port-fxo-2-port-fxo-module-p-428.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 429 Digium TDM808B 2 Quad Port FXO Module 329.8000 424.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm808b-2-quad-port-fxo-module-p-429.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 430 Digium TDM810B 1 Port FXS Module 140.3000 162.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm810b-1-port-fxs-module-p-430.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 431 Digium TDM811B 1 Port FXS & 1 Port FXO Module 185.7000 208.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm811b-1-port-fxs-1-port-fxo-module-p-431.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 432 Digium TDM812B 1 Port FXS & 2 Port FXO Module 231.1000 254.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm812b-1-port-fxs-2-port-fxo-module-p-432.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 433 Digium TDM813B 1 Port FXS & 3 Port FXO Module 276.4000 299.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm813b-1-port-fxs-3-port-fxo-module-p-433.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 434 Digium TDM814B 1 Port FXS & 1 Quad Port FXO Module 255.1000 313.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm814b-1-port-fxs-1-quad-port-fxo-module-p-434.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 435 Digium TDM820B 2 Port FXS Module 180.4000 202.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm820b-2-port-fxs-module-p-435.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 436 Digium TDM821B 2 Port FXS & 1 Port FXO Module 225.8000 248.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm821b-2-port-fxs-1-port-fxo-module-p-436.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 437 Digium TDM822B 2 Port FXS & 2 Port FXO Module 271.1000 294.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm822b-2-port-fxs-2-port-fxo-module-p-437.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 438 Digium TDM824B 2 Port FXS & 1 Quad Port FXO Module 295.2000 354.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm824b-2-port-fxs-1-quad-port-fxo-module-p-438.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\\\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\\\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\\\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\\\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 439 Digium TDM830B 3 Port FXS Module 220.4000 243.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm830b-3-port-fxs-module-p-439.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 440 Digium TDM831B 3 Port FXS & 1 Port FXO Module 265.8000 289.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm831b-3-port-fxs-1-port-fxo-module-p-440.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 441 Digium TDM84SB 4 Port FXS Module 260.5000 283.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm84sb-4-port-fxs-module-p-441.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 442 Digium TDM840B 1 x Quad Port FXS Module 199.1000 264.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm840b-1-x-quad-port-fxs-module-p-442.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\\\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\\\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\\\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\\\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 443 Digium TDM841B 1 x Quad Port FXS & 1 Port FXO Module 244.4000 310.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm841b-1-x-quad-port-fxs-1-port-fxo-module-p-443.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 444 Digium TDM842B 1 x Quad Port FXS & 2 Port FXO Module 289.8000 356.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm842b-1-x-quad-port-fxs-2-port-fxo-module-p-444.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 445 Digium TDM844B 1 x Quad Port FXS & 1 x Quad Port FXO Module 313.8000 416.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm844b-1-x-quad-port-fxs-1-x-quad-port-fxo-module-p-445.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 446 Digium TDM850B 1 x Quad Port FXS & 1 Port FXS Module 239.0000 305.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm850b-1-x-quad-port-fxs-1-port-fxs-module-p-446.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 447 Digium TDM851B 1 x Quad Port FXS & 1 Port FXS & 1 Port FXO Modul 284.4000 351.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm851b-1-x-quad-port-fxs-1-port-fxs-1-port-fxo-modul-p-447.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 448 Digium TDM860B 1 x Quad Port FXS & 2 Port FXS Module 279.1000 345.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm860b-1-x-quad-port-fxs-2-port-fxs-module-p-448.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 449 Digium TDM880B 2 x Quad Port FXS Module 297.8000 408.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm880b-2-x-quad-port-fxs-module-p-449.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 451 SIP / IAX Call Credit 500 500.00 VoIP Call Credit http://www.voipon.co.uk/sip-iax-call-credit-p-451.html http://www.voipon.co.uk/images/500.gif new Availability: Instant Activation Enjoy all the benefits of VoIP with our PSTN termination services with VoIPon Call Credit. What do I get? With VoIPon Call Credit you receive free registration to our SIP subscription service (you get a SIP address) with which you can make PSTN calls and a host of other features: Low UK Call Rates ( more info ) Low International Call Rates ( more info ) Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Scheduled Dialplan ( more info ) SMS Call Back ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) Online Billing Information How does it work? Simply add this item and your credit will be updated accordingly. Call credit will be allocated to your VoIPon account. If this is the first time shopping with us, you can choose to register with VoIPon in the create account page. UK Charges - Summary Call Type Peak* Off-peak Weekend Geographic 1.2 0.9 0.9 Mobile - from 11.4 11.4 11.4 Local rate (0845) 6.0 6.0 6.0 National rate (0870) 7.0 7.0 7.0 * Peak time is 8am to 6pm. You can call ALL these countries for as little as 1.5 pence per minute! Argentina Australia Belgium Canada China Cyprus Denmark France Germany Greece Hong Kong Hungary Israel Italy Luxembourg Netherlands New Zealand Norway Poland Portugal Singapore Spain Taiwan USA Don't Forget ...... The above countries are what we call lo-cost which, at 1.5p per minute is fantastic value BUT, if you are calling to another VoIP user, the call is FREE! Many other countries are much cheaper to call using VoIP, some examples are: Brazil 2.8ppm Finland 2.8ppm Gibraltar 3.7ppm India 6.0ppm Kenya 9.8ppm Malta 5.6ppm Monaco 2.1ppm Nigeria 9.1ppm Russia 3.7ppm Zimbabwe 5.3ppm 452 Junghanns duoBRI 2.0 PCI ISDN card 210.0000 245.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-duobri-20-pci-isdn-card-p-452.html http://www.voipon.co.uk/images/junghanns_duobri_small.jpg new Availability: In stock The duoBRI® 2.0 PCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world. Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The duoBRI® 2.0 PCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. Multiple duoBRI® 2.0 PCI ISDN cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge B channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. B channels on different cards can be actively switched). Target applications ISDN BRI PBX ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking Requirements CPU 500+ Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 2 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge PCM bus connectors daisy chaining of max. 8 cards 4 dual-color LEDs (layer 1 state indicators) Active channel switching (across multiple cards over the external PCM bus) Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) euroISDN protocol stack suitable for 3.3 volts and 5.0 volts 32 bit PCI slots Hardware watchdog Optional onboard S0 bus power feeding module available 453 Junghanns unoBRI miniPCI Card 185.0000 210.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-unobri-minipci-card-p-453.html http://www.voipon.co.uk/images/junghanns_unobri_minipci.jpg new Availability: In Stock The unoBRI ® miniPCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world (see Application 1). Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The unoBRI® miniPCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. The miniPCI form factor is the ideal base to build highly integrated and cost effective Voice over IP gateways and PBX solutions. Target applications ISDN BRI PBX (embedded) ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking IAD with 1 BRI interfaces Requirements CPU 200+ Mhz (AMD Geode tested) RAM 32+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) Features 1 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge 1 dual-color LEDs (layer 1 state indicators)active channel switching (across multiple cards over the external PCM bus) TRB3/TRB3-A1 certified Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) EuroISDN protocol stack Suitable for miniPCI III slots 454 Junghanns duoBRI miniPCI Card 215.0000 245.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-duobri-minipci-card-p-454.html http://www.voipon.co.uk/images/junghanns_duobri_minipci.jpg new Availability: In Stock The duoBRI ® miniPCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world (see Application 1). Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The duoBRI® miniPCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. The miniPCI form factor is the ideal base to build highly integrated and cost effective Voice over IP gateways and PBX solutions. Target applications ISDN BRI PBX (embedded) ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking IAD with 2 BRI interfaces Requirements CPU 200+ Mhz (AMD Geode tested) RAM 32+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) Features 2 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge 2 dual-color LEDs (layer 1 state indicators) active channel switching (across multiple cards over the external PCM bus) TRB3/TRB3-A1 certified Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) EuroISDN protocol stack Suitable for miniPCI III slots 455 Junghanns quadBRI miniPCI Card 275.0000 315.00 Junghanns Junghanns BRI Cards http://www.voipon.co.uk/junghanns-quadbri-minipci-card-p-455.html http://www.voipon.co.uk/images/junghanns_quadbri_minipci.jpg new Availability: In Stock The quadBRI® miniPCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN TA) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world (see Application 1). Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The quadBRI® miniPCI ISDN brings powerful ISDN BRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. All 4 BRI ports can be configured for TE or NT mode individually by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned. The miniPCI form factor is the ideal base to build highly integrated and cost effective Voice over IP gateways and PBX solutions. Target applications ISDN BRI PBX (embedded) ISDN least cost routers Voice over IP BRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking IAD with 4 BRI interfaces Requirements CPU 200+ Mhz (AMD Geode tested) RAM 32+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) Features 4 Basic Rate Interface ports (I.421) for TE and NT mode DTMF detection Conference bridge 4 dual-color LEDs (layer 1 state indicators) active channel switching (across multiple cards over the external PCM bus) TRB3/TRB3-A1 certified Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) EuroISDN protocol stack Suitable for miniPCI III slots 456 Junghanns unoGSM PCI Card 640 640.00 Junghanns Junghanns GSM Cards http://www.voipon.co.uk/junghanns-unogsm-pci-card-p-456.html http://www.voipon.co.uk/images/junghanns_unogsm_pci.jpg new Availability: In Stock The Junghanns.NET GSM® PCI series cards provide scalable connectivity to GSM networks for your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. Multiple Junghanns.NET GSM® PCI series cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge voice channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. voice channels on different cards can be actively switched). The Junghanns.NET GSM® PCI series cards can be connected to any other Junghanns.NET card to build a real TDM switched PBX. Target applications GSM connectivity for PBX mobile PBX GSM VoIP gateway SMS gateway GSM callback services Requirements +1000 Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 1 GSM dualband channels (quadband available soon) 1 SIM card per GSM channel optional SIM expansion boards available (2 SIM per channel, swapable during operation) DIGITAL audio quality (noise suppression, echocancel) fast call setup DTMF detection GSM data connections GPRS data connections PCM bus connectors daisy chaining of max. 8 cards 1 dual-color LEDs (network state indicators) active channel switching (across multiple cards over the external PCM bus) suitable for 3.3 volts and 5.0 volts 32 bit PCI slots 457 Junghanns duoGSM PCI Card 754.0000 840.00 Junghanns Junghanns GSM Cards http://www.voipon.co.uk/junghanns-duogsm-pci-card-p-457.html http://www.voipon.co.uk/images/junghanns_duogsm.jpg new Availability: In Stock The Junghanns.NET GSM® PCI series cards provide scalable connectivity to GSM networks for your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. Multiple Junghanns.NET GSM® PCI series cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge voice channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. voice channels on different cards can be actively switched). The Junghanns.NET GSM® PCI series cards can be connected to any other Junghanns.NET card to build a real TDM switched PBX. Target applications GSM connectivity for PBX mobile PBX GSM VoIP gateway SMS gateway GSM callback services Requirements +1000 Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 2 GSM dualband channels (quadband available soon) 1 SIM card per GSM channel optional SIM expansion boards available (2 SIM per channel, swapable during operation) DIGITAL audio quality (noise suppression, echocancel) fast call setup DTMF detection GSM data connections GPRS data connections PCM bus connectors daisy chaining of max. 8 cards 2 dual-color LEDs (network state indicators) active channel switching (across multiple cards over the external PCM bus) suitable for 3.3 volts and 5.0 volts 32 bit PCI slots 458 Junghanns quadGSM PCI Card 1001.0000 1115.00 Junghanns Junghanns GSM Cards http://www.voipon.co.uk/junghanns-quadgsm-pci-card-p-458.html http://www.voipon.co.uk/images/junghanns_quadgsm.jpg new Availability: In Stock The Junghanns.NET GSM® PCI series cards provide scalable connectivity to GSM networks for your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. Multiple Junghanns.NET GSM® PCI series cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge voice channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. voice channels on different cards can be actively switched). The Junghanns.NET GSM® PCI series cards can be connected to any other Junghanns.NET card to build a real TDM switched PBX. Target applications GSM connectivity for PBX mobile PBX GSM VoIP gateway SMS gateway GSM callback services Requirements +1000 Mhz RAM 64+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 4 GSM dualband channels (quadband is now available) 1 SIM card per GSM channel optional SIM expansion boards available (2 SIM per channel, swapable during operation) DIGITAL audio quality (noise suppression, echocancel) fast call setup DTMF detection GSM data connections GPRS data connections PCM bus connectors daisy chaining of max. 8 cards 4 dual-color LEDs (network state indicators) active channel switching (across multiple cards over the external PCM bus) suitable for 3.3 volts and 5.0 volts 32 bit PCI slots 459 Junghanns singleE1 PCI ISDN card 425.0000 490.00 Junghanns Junghanns PRI cards http://www.voipon.co.uk/junghanns-singlee1-pci-isdn-card-p-459.html http://www.voipon.co.uk/images/junghanns_singe1.jpg new Availability: In Stock The single E1® PCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN PBX) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world (see Application 1). Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The single E1® PCI ISDN brings powerful ISDN PRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. The E1 PRI port can be configured for CPE or NET operation by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN, AT&T 4ESS, DMS 100, Lucent 5E and National ISDN 2 signalling Multiple single E1® PCI ISDN cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge B channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. B channels on different cards can be actively switched). Target applications ISDN PRI PBX ISDN least cost routers Voice over IP PRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking Requirements CPU 1000+ Mhz RAM 128+ MB Linux 2.4.X or 2.6.X Kernel (2.6.Xrecommended) 32bit PCI Version 2.2 slot Features 1 Primary Rate Interface port for CPE and NET operation DTMF detection Conference bridge PCM bus connectors (PCM30, PCM64, PCM128) 2 x 4 dual-color LEDs (layer 1 state indicators) active channel switching (across multiple cards over the external PCM bus) Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) PRI ISDN protocol stack suitable for 3.3 volts and 5.0 volts 32 bit PCI slots 460 Junghanns doubleE1 PCI ISDN card 755.0000 870.00 Junghanns Junghanns PRI cards http://www.voipon.co.uk/junghanns-doublee1-pci-isdn-card-p-460.html http://www.voipon.co.uk/images/junghanns_doublee1.jpg new Availability: In Stock The double E1® PCI ISDN turns your legacy ISDN equipment (or analog equipment behind an ISDN PBX) into powerful Voice over IP devices. It provides a soft migration path from ISDN technology to the new Voice over IP world (see Application 1). Connect your ISDN PBXes at different locations with Voice over IP and drop costs for company internal calls close to zero. Transparently add least cost routing over ISDN or VoIP carriers to reduce costs on outbound calls significantly. The double E1® PCI ISDN brings powerful ISDN PRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. The E1 PRI port can be configured for CPE or NET operation by jumpers. This port configuration is detected by the driver automatically. The drivers can handle the user and network side of euroISDN, AT&T 4ESS, DMS 100, Lucent 5E and National ISDN 2 signalling Two double E1® PCI ISDN cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge B channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. B channels on different cards can be actively switched). Target applications ISDN PRI PBX ISDN least cost routers Voice over IP PRI gateways VoIP integration of ISDN equipment PBX to PBX VoIP trunking Requirements CPU 1000+ Mhz RAM 128+ MB Linux 2.4.X or 2.6.X Kernel (2.6.X recommended) 32bit PCI Version 2.2 slot Features 2 Primary Rate Interface port for CPE and NET operation DTMF detection Conference bridge PCM bus connectors (PCM30, PCM64, PCM128) 2 x 4 dual-color LEDs (layer 1 state indicators) active channel switching (across multiple cards over the external PCM bus) Point-to-Point (TE / NT) and Point-to-Multipoint (TE / NT) PRI ISDN protocol stack suitable for 3.3 volts and 5.0 volts 32 bit PCI slots 461 Junghanns ISDNguard automatic layer 1 failover switch 310.0000 355.00 Junghanns Junghanns Failover Solutions http://www.voipon.co.uk/junghanns-isdnguard-automatic-layer-1-failover-switch-p-461.html http://www.voipon.co.uk/images/junghanns_isdnguard.jpg new Availability: In Stock The Junghanns.NET ISDNguard is an automatic layer 1 failover switch that has been exclusively designed for the Asterisk® open source PBX. It can be used for various failover scenarios that require a physical reconnection of up to 4 BRI/PRI interfaces (see Application 1 and 2). The Asterisk® server provides a heartbeat to the ISDNguard over a serial (RS232) interface. If the heartbeat stops the ISDNguard will go into failover state, directly connecting the ISDN NET with the ISDN CPE ports indicated by the red failover LED. In normal operation the heartbeat stops whenever the Asterisk® server is stopped. A configuration option allows the heartbeat to continue if the Asterisk® server was cleanly shut down. Multiple ISDNguards can be cascaded so they can be switched by a single Asterisk® server with a single RS232 interface. In case of a power failure the ISDNguard will automatically fall back to failover state. Target Applications Integration of ISDN equipment PBX with redundant BRI/PRI ports Redundant VoIP gateways Layer 1 port switch LEDs Power ok (green) Heartbeat ok (green) Failover Modus (red) Requirements Linux / *BSD operating system RS232 (D-SUB 9) interface Casing 19" rack mountable 1 unit in height brushed stainless steel Interfaces 1 RS232 (heartbeat) 2 DIN5 (cascading) 4 shielded RJ45 (ISDN CPE) 4 shielded RJ45 (ISDN NET) 4 shielded RJ45 (Asterisk CPE) 4 shielded RJ45 (Asterisk NET) 4 shielded RJ45 (octoBRI ISDN) 462 Junghanns IP Power Switch 395.0000 416.00 Junghanns Junghanns Failover Solutions http://www.voipon.co.uk/junghanns-ip-power-switch-p-462.html http://www.voipon.co.uk/images/junghanns_ip_switch.jpg new Availability: In Stock The IP Switch will be attached to any TCP/IP network and is accessible over it. The plugged devices can be turned on or off independently via ethernet. Simply two steps are required for start-up. The connection to an existing power supply system as well as a TCP/IP network and the configuration of the IP settings. These can take place via DHCP or manually. After assignment of IP settings, the IP Switch may be accessed via web browser in the same network. The access can be protected by an administrator and an user password. It is also possible to access the IP Switch via command line tool (e.g. for automated or time controlled device switching). Features Easy and fast start-up Automatic configuration via DHCP Ease of use Cross platform Independent switching of the eight sockets Access protection by the use of optional administrator and user password Integrated reset function Accessible via command line tool Technical Characteristics Network connection: 10MBit 10baseT Ethernet Protocol: TCP/IP Capacity per socket: max. 2000W Over-all capacity: max. 3000W Switching voltage: 230V Operating temperature 0°C-50°C/32°F-122°F 463 Junghanns SIM Bracket 39.0000 42.00 Junghanns Junghanns GSM Cards http://www.voipon.co.uk/junghanns-sim-bracket-p-463.html http://www.voipon.co.uk/images/junghanns_sim_bracket.jpg new Availability: In Stock The SIM Bracket lets you use 4 more SIM cards with your quadGSM® or other GSM card. You can select the SIM cards through the driver. 464 Junghanns Hot Plug SIM Changer 98.0000 105.00 Junghanns Junghanns GSM Cards http://www.voipon.co.uk/junghanns-hot-plug-sim-changer-p-464.html http://www.voipon.co.uk/images/junghanns_sim_changer.jpg new Availability: In Stock The Hotplug SIM changer allowes you easily to exchange faulty or broken SIM cards externally. There is no need to power down the server and plug out the GSM card. Additionally just like the SIM Bracket the Hotplug SIM Changer lets you use 4 more SIM cards for the quadGSM® card, giving you a total of 8 SIM cards to be controlled by the driver. 465 Digium Wildcard TE120P PCI ISDN PRI Card 266.5000 319.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te120p-pci-isdn-pri-card-p-465.html http://www.voipon.co.uk/images/digium_te120P.jpg new Availability: In Stock High Performance Digium Wildcard TE120P voIP Card Digium's Wildcard TE120P is a high-performance, cost effective digital telephony interface card that makes it possible to create a seamless network, interconnecting conventional telephony systems with VoIP technologies. digium & Asterisk® The TE120P can be used to provide a wide range of PBX and IVR services to the network or handset including call conferencing, voicemail, 3-way calling, and VoIP gateways. The Wildcard TE120P is a single span, selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) card. The card utilizes Digium's VoiceBus® technology . VoiceBus technology allows the TE120P to use an industry standard bus-mastering PCI interface, to maximise system compatibility and eliminate system conflicts. The TE120P supports both voice and data modes on its single span. The card can support 12 channels dedicated to voice, routed directly to the Asterisk PBX, and 12 to data, handled by the underlying Linux OS, therefore eliminating the need for an external router. The TE120P works in both 3V and 5V slots by auto detecting the slot's voltage automatically. By utilising TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can now easily connect multiple PCs equipped with the TE120P and achieve voice quality comparable to single PBX implementations. Scalability for this product comes from adding multiple TE120Ps to each individual PC. The TE120P supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Using this card in concert with Digium's Asterisk® software, standard PC hardware, and the Linux® OS, you can upgrade your PBX to a sophisticated telephony environment capable of supporting both voice and data channels. 466 Digium TC400B Transcoder Card 786.8000 890.00 Digium Digium Transcoder Card http://www.voipon.co.uk/digium-tc400b-transcoder-card-p-466.html http://www.voipon.co.uk/images/digium_tc400b_transcoder_ca.jpg new Availability: In Stock The TC400B is a half-length, low-profile PCI 2.2-compliant card for transforming complex VoIP codecs into simple codecs. digium hardware The TC400B is a bundle of the half-length, low-profile PCI-2.2 compliant TC400P base card and the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software. Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B not only relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible. The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 96 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium's software-based G.729a codec licensing. 467 Octasic 1 License SoftEcho 5.49 5.49 Octasic Semiconductor Octasic SoftEcho http://www.voipon.co.uk/octasic-1-license-softecho-p-467.html http://www.voipon.co.uk/images/softecho.gif new Availability: In stock OctWare's SoftEcho is a simple to integrate echo cancellation application for Asterisk based IP PBX systems up to 8 channels. Available per channel, SoftEcho enables integrators to cost-effectively deliver products with industry-leading sound quality. Based on years of experience in trunk side as well as local side network echo cancellation, SoftEcho's algorithm is auto-tuning meaning that no adjustments are necessary: just install SoftEcho and your IP PBX will offer superior voice quality. SoftEcho Delivers Superior Performance by Way of: Algorithm transparency Excellent double-talk handling Fast convergence High quality background noise handling Long echo tail Performance and voice quality statistics Line Echo Cancellation (LEC) Feature List Up to 128 ms echo tail Transparent algorithm Auto-tuning algorithm Dial tone and digit transparency G.168-2004 compliant How to Download & Purchase SoftEcho product: 2 Easy Steps to Download & Purchase SoftEcho: 1. Download SoftEcho by clicking on the image below. 2. Add the required Activation-Keys to your shopping basket and purchase them. 468 Octasic 8 License SoftEcho 42 42.00 Octasic Semiconductor Octasic SoftEcho http://www.voipon.co.uk/octasic-8-license-softecho-p-468.html http://www.voipon.co.uk/images/softecho.gif new Availability: In stock OctWare's SoftEcho is a simple to integrate echo cancellation application for Asterisk based IP PBX systems up to 8 channels. Available per channel, SoftEcho enables integrators to cost-effectively deliver products with industry-leading sound quality. Based on years of experience in trunk side as well as local side network echo cancellation, SoftEcho's algorithm is auto-tuning meaning that no adjustments are necessary: just install SoftEcho and your IP PBX will offer superior voice quality. SoftEcho Delivers Superior Performance by Way of: Algorithm transparency Excellent double-talk handling Fast convergence High quality background noise handling Long echo tail Performance and voice quality statistics Line Echo Cancellation (LEC) Feature List Up to 128 ms echo tail Transparent algorithm Auto-tuning algorithm Dial tone and digit transparency G.168-2004 compliant How to Download & Purchase SoftEcho product: 2 Easy Steps to Download & Purchase SoftEcho: 1. Download SoftEcho by clicking on the image below. 2. Add the required Activation-Keys to your shopping basket and purchase them. 469 PIKA Single Span T1/E1 PRI ISDN Card with Echo Cancellation 574 574.00 Pika Technologies Pika PRI Cards http://www.voipon.co.uk/pika-single-span-t1e1-pri-isdn-card-with-echo-cancellation-p-469.html http://www.voipon.co.uk/images/pika_pri_pci_card.jpg new Availability In Stock PIKA For Asterisk T1/E1 Gateway Board PIKA for Asterisk Digital Single Span T1/E1 PRI ISDN Card with Echo Cancellation features market leading flexible port densities. Currently available in a PCI variant with PCIe coming later this year. Up to four T1/E1 network interfaces can be enabled with in field license upgrades.With 20 proven years experience in the board market, PIKA has evolved its TDM interface cards to fully integrate into an Asterisk Open Source PBX application and provides the most reliable choice for your Asterisk application. Unparalleled Flexibility Every T1/E1 gateway card ships from Pika with hardware capacity for up to 4 spans. At initial order,1, 2, 3 or 4 T1/E1 spans are enabled. However, should your Asterisk system grow over time, you can expand the port capacity by installing a cost effective span upgrade license (shipped to you by email). How simple can it get! Features & Benefits Provides low cost Primary Rate ISDN (NA and Euro) network access for Asterisk environments All on Host (AoH) echo cancellation provides DSP quality, software based Echo cancellation using a superior, low processing power algorithm. Because it is done on the host, expensive echo cancelling hardware is not required All on Host (AoH) technology removes expensive DSP resources to provide an economical board solution A single board supports cost effective growth for up to120 channels Additional T1/E1 spans can be added with software licensing eliminating the need for additional PCI slots Support for future applications such as fax can be done with simple software upgrades Installation is as simple as downloading and installing the Pika channel driver for Asterisk and plugging in the boards! Native switching between voice channels (B channels) on the same card or across multiple cards provides a lower latency connection improving voice conversation quality Pika provides no charge technical support to help you get your Asterisk application up and running. ROHS All Pika boards are ROHS compliant. Warranty Pika provides 3 warranty on all boards 470 PIKA Dual Span T1/E1 PRI ISDN Card with Echo Cancellation 674 674.00 Pika Technologies Pika PRI Cards http://www.voipon.co.uk/pika-dual-span-t1e1-pri-isdn-card-with-echo-cancellation-p-470.html http://www.voipon.co.uk/images/pika_pri_pci_card.jpg new Availability In Stock PIKA For Asterisk T1/E1 Gateway Board PIKA for Asterisk Digital Dual Span T1/E1 PRI ISDN Card with Echo Cancellation features market leading flexible port densities. Currently available in a PCI variant with PCIe coming later this year. Up to four T1/E1 network interfaces can be enabled with in field license upgrades.With 20 proven years experience in the board market, PIKA has evolved its TDM interface cards to fully integrate into an Asterisk Open Source PBX application and provides the most reliable choice for your Asterisk application. Unparalleled Flexibility Every T1/E1 gateway card ships from Pika with hardware capacity for up to 4 spans. At initial order,1, 2, 3 or 4 T1/E1 spans are enabled. However, should your Asterisk system grow over time, you can expand the port capacity by installing a cost effective span upgrade license (shipped to you by email). How simple can it get! Features & Benefits Provides low cost Primary Rate ISDN (NA and Euro) network access for Asterisk environments All on Host (AoH) echo cancellation provides DSP quality, software based Echo cancellation using a superior, low processing power algorithm. Because it is done on the host, expensive echo cancelling hardware is not required All on Host (AoH) technology removes expensive DSP resources to provide an economical board solution A single board supports cost effective growth for up to120 channels Additional T1/E1 spans can be added with software licensing eliminating the need for additional PCI slots Support for future applications such as fax can be done with simple software upgrades Installation is as simple as downloading and installing the Pika channel driver for Asterisk and plugging in the boards! Native switching between voice channels (B channels) on the same card or across multiple cards provides a lower latency connection improving voice conversation quality Pika provides no charge technical support to help you get your Asterisk application up and running. ROHS All Pika boards are ROHS compliant. Warranty Pika provides 3 warranty on all boards 471 PIKA Triple Span T1/E1 PRI ISDN Card with Echo Cancellation 774 774.00 Pika Technologies Pika PRI Cards http://www.voipon.co.uk/pika-triple-span-t1e1-pri-isdn-card-with-echo-cancellation-p-471.html http://www.voipon.co.uk/images/pika_pri_pci_card.jpg new Availability In Stock PIKA For Asterisk T1/E1 Gateway Board PIKA for Asterisk Digital Triple Span T1/E1 PRI ISDN Card with Echo Cancellation features market leading flexible port densities. Currently available in a PCI variant with PCIe coming later this year. Up to four T1/E1 network interfaces can be enabled with in field license upgrades.With 20 proven years experience in the board market, PIKA has evolved its TDM interface cards to fully integrate into an Asterisk Open Source PBX application and provides the most reliable choice for your Asterisk application. Unparalleled Flexibility Every T1/E1 gateway card ships from Pika with hardware capacity for up to 4 spans. At initial order,1, 2, 3 or 4 T1/E1 spans are enabled. However, should your Asterisk system grow over time, you can expand the port capacity by installing a cost effective span upgrade license (shipped to you by email). How simple can it get! Features & Benefits Provides low cost Primary Rate ISDN (NA and Euro) network access for Asterisk environments All on Host (AoH) echo cancellation provides DSP quality, software based Echo cancellation using a superior, low processing power algorithm. Because it is done on the host, expensive echo cancelling hardware is not required All on Host (AoH) technology removes expensive DSP resources to provide an economical board solution A single board supports cost effective growth for up to120 channels Additional T1/E1 spans can be added with software licensing eliminating the need for additional PCI slots Support for future applications such as fax can be done with simple software upgrades Installation is as simple as downloading and installing the Pika channel driver for Asterisk and plugging in the boards! Native switching between voice channels (B channels) on the same card or across multiple cards provides a lower latency connection improving voice conversation quality Pika provides no charge technical support to help you get your Asterisk application up and running. ROHS All Pika boards are ROHS compliant. Warranty Pika provides 3 warranty on all boards 472 PIKA Quad Span T1/E1 PRI ISDN Card with Echo Cancellation 874 874.00 Pika Technologies Pika PRI Cards http://www.voipon.co.uk/pika-quad-span-t1e1-pri-isdn-card-with-echo-cancellation-p-472.html http://www.voipon.co.uk/images/pika_pri_pci_card.jpg new Availability In Stock PIKA For Asterisk T1/E1 Gateway Board PIKA for Asterisk Digital Quad Span T1/E1 PRI ISDN Card with Echo Cancellation features market leading flexible port densities. Currently available in a PCI variant with PCIe coming later this year. Up to four T1/E1 network interfaces can be enabled with in field license upgrades.With 20 proven years experience in the board market, PIKA has evolved its TDM interface cards to fully integrate into an Asterisk Open Source PBX application and provides the most reliable choice for your Asterisk application. Unparalleled Flexibility Every T1/E1 gateway card ships from Pika with hardware capacity for up to 4 spans. At initial order,1, 2, 3 or 4 T1/E1 spans are enabled. However, should your Asterisk system grow over time, you can expand the port capacity by installing a cost effective span upgrade license (shipped to you by email). How simple can it get! Features & Benefits Provides low cost Primary Rate ISDN (NA and Euro) network access for Asterisk environments All on Host (AoH) echo cancellation provides DSP quality, software based Echo cancellation using a superior, low processing power algorithm. Because it is done on the host, expensive echo cancelling hardware is not required All on Host (AoH) technology removes expensive DSP resources to provide an economical board solution A single board supports cost effective growth for up to120 channels Additional T1/E1 spans can be added with software licensing eliminating the need for additional PCI slots Support for future applications such as fax can be done with simple software upgrades Installation is as simple as downloading and installing the Pika channel driver for Asterisk and plugging in the boards! Native switching between voice channels (B channels) on the same card or across multiple cards provides a lower latency connection improving voice conversation quality Pika provides no charge technical support to help you get your Asterisk application up and running. ROHS All Pika boards are ROHS compliant. Warranty Pika provides 3 warranty on all boards 473 Polycom SoundPoint IP 320 (IP320) 53.5000 57.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip-320-ip320-p-473.html http://www.voipon.co.uk/images/polycom_ip320_330.jpg new Availability: In stock Please note the the SoundPoint IP 330 and 320 are PoE and require a power supply which can be purchased seperaterly here . Entry-level IP phones with excellent sound quality, an enterprise-grade feature set and that deliver remarkable value. The SoundPoint IP 320 two-line SIP phone delivers superb sound quality and offers a wide range of supported business telephony features. While the SoundPoint IP 330 incorporates dual-port 10/100 Ethernet switch for LAN and PC connection, the SoundPoint IP 320 phone has a single 10/100 Ethernet port, therefore its intended applications include common areas, such as hallways, lobbies and break rooms, as well as various wall-mounted deployments. Exceptional Sound Quality The Polycom SoundPoint IP 330 and 320 phones feature a full-duplex IEEE 1329 Type 1-compliant speakerphone with Polycom's reknowned Acoustic Clarity Technology that delivers superb sound quality and enables echo-free and noise-free conversations that feel as natural as being there. Enterprise-Grade Feature Set The SoundPoint IP 320 and 330 phones deliver a full feature set through an intuitive user interface - encompassing both traditional business telephony features such as call hold, park, transfer, pick-up, and three-way local conferencing, as well as more advanced capabilities like shared call/bridged line appearance, support of XHTML applications and distinctive call treatment. Efficient Installation and Provisioning The SoundPoint IP 320 is engineered to make installation, configuration and upgrade as simple and efficient as possible. The phone's standard base can be reversed to become a wall mount, eliminating the need for a separate accessory. The built-in IEEE 802.3af PoE circuitry and a dual-port Ethernet switch (SoundPoint IP 330 phone only) enable flexible deployment options and savings on cabling cost. The Polycom IP 320 and 330 phones support remote, zero-touch provisioning and upgrade from a variety of servers, including FTP, TFTP, HTTP, or HTTPS. To ensure reliable, uninterrupted performance, the phones support boot and call server redundancy. Broad Interoperability - Maxium Versatility The SoundPoint IP 320 and 330 phones have been tested and certified to deliver extensive feature support and comprehensive interoperability with leading SIP-based PBX platforms by Digium, BroadSoft, Sylantro, Interactive Intelligence/Vonexus, and other Polycom Technology and Interoperability Partners2. The IP320 and IP 330 phones also interoperate with Microsoft Live Communications Server (LCS) 2005 for telephony and presence. This broad interoperability provides Polycom customers a wide choice of supported standards-based call server solutions with which the phones can be deployed.   Soundpoint IP320 Features & Benefits Excellent Sound Quality - Polycom Acoustic Clarity Technology enables crystal-clear simultaneous hands-free conversations that feel as natural as being there. Enterprise-Grade Feature Set Two lines Full-duplex speakerphone Support of shared lines, presence, 3-way local conferencing, and distinctive call treatment Easy-to-read, 102 x 33-pixel graphical LCD XHTML micro-browser for Web applications Efficient Installation and Provisioning Integrated IEEE 802.3af Power over Ethernet (PoE) support Remote, zero-touch provisioning with support of a variety of servers Broad Interoperability - Certified to support a rich set of features with a variety of leading SIP-based IP PBX and Softswitch platforms and to interoperate with Microsoft LCS for telephony and presence 474 Polycom SoundPoint IP 330 (IP330) 69.0000 75.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip-330-ip330-p-474.html http://www.voipon.co.uk/images/polycom_ip320_330.jpg new Availability: In stock Please note the the SoundPoint IP 330 and 320 are PoE and require a power supply which can be purchased seperately here . Polycom Soundpoint IP 330 - Cost effective entry-level IP phone with excellent sound quality & enterprise-grade feature set The SoundPoint IP 330 and 320 are two-line SIP phones that deliver superb sound quality and a wide range of supported business telephony features. The SoundPoint IP 330 phone, with its dual-port 10/100 Ethernet switch for LAN and PC connection, represents a cost-effective solution for cubicle workers and call centre operators who use a "hard" phone in parallel with a soft phone client running on the PC. The SoundPoint IP 320 phone has a single 10/100 Ethernet port, therefore its intended uses include common areas, such as lobbies, hallways, break rooms, as well as various wall-mounted deployments. Superior Sound Quality The SoundPoint IP 330 and 320 phones feature a full-duplex IEEE 1329 Type 1-compliant speakerphone with Polycom's reknowned Acoustic Clarity Technology that delivers excellent sound quality and provides echo-free and noise-free conversations that are as natural as being there. Enterprise-Grade Feature Set The SoundPoint IP 330 and 320 phones deliver a full feature set through an intuitive user interface which encompasses both traditional business telephony features such as call hold, pick-up, park, transfer and three-way local conferencing, and more advanced capabilities such as shared call/bridged line appearance, support of XHTML applications and distinctive call treatment. Efficient Installation and Provisioning The Polycom SoundPoint IP 330 and 320 phones are engineered to make installation, configuration and upgrading as straight-forward and efficient as possible. The phones' standard base can be reversed to become a wall mount, with no need for a separate accessory. The built-in IEEE 802.3af PoE circuitry and a dual-port Ethernet switch (SoundPoint IP 330 phone only) provide flexible deployment options and savings on cabling costs. The SoundPoint IP 330 and 320 phones support remote, zero-touch provisioning and upgrade from a variety of servers, including FTP, TFTP, HTTP, or HTTPS. In order to ensure reliable, uninterrupted performance, the phones support boot and call server redundancy. Broad Interoperability The Polycom SoundPoint IP 330 and 320 phones have been tested and certified to deliver comprehensive interoperability and extensive feature support with leading SIP-based call control platforms by Digium, BroadSoft, Sylantro, Interactive Intelligence/Vonexus and other Polycom Technology and Interoperability Partners2. The phones also interoperate with Microsoft Live Communications Server (LCS) 2005 for telephony and presence. This broad, multi-system interoperability gives Polycom customers a choice of supported standards-based call server solutions with which the phones can be deployed. Soundpoint IP 330 Features & Benefits Excellent Sound Quality Polycom Acoustic Clarity Technology enables crystal-clear simultaneous hands-free conversations that feel as natural as being there. Enterprise-Grade Feature Set Two lines Easy-to-read, 102 x 33-pixel graphical LCD Full-duplex speakerphone Support of shared lines, presence, 3-way local conferencing, and distinctive call treatment XHTML micro-browser for Web applications Efficient Installation and Provisioning Integrated IEEE 802.3af Power over Ethernet (PoE) support Remote, zero-touch provisioning with support of a variety of servers Broad Interoperability - Certified & tested to support a rich set of features with a variety of leading SIP-based IP PBX and Softswitch platforms and to interoperate with Microsoft LCS for telephony and presence 475 Polycom SoundPoint IP 550 (IP550) 160.0000 176.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip-550-ip550-p-475.html http://www.voipon.co.uk/images/polycom_ip550-ip560_sml.jpg new Availability: In stock Polycom soundpoint IP550 - Revolutionary sound quality & advanced features in a 4-line SIP phone The SoundPoint IP 550 desktop phone is a four-line SIP phone that delivers calls of unprecedented clarity and richness and supports a comprehensive range of cutting-edge features. Unrivaled Sound Quality The SoundPoint IP 550 voIP phone features Polycom's revolutionary HD Voice technology that brings life-like richness and clarity to each and every call. Polycom HD Voice technology incorporates wideband audio for more than twice the voice clarity; Polycom's patented Acoustic Clarity Technology gives crystal-clear, noise- and echo-free sound, plus best-in-class system design for high-fidelity, faithful voice reproduction. Comprehensive, Cutting-Edge Feature Set The SoundPoint IP 550 desktop IP phone provides advanced features and capabilities, such as shared call/ bridged line appearance, busy lamp field (BLF), text messaging, presence and buddy lists. The Polycom SoundPoint IP 550 phone also incorporates an integrated XHTML micro-browser that enables users to make use of productivity-enhancing Web-based applications. With support for four lines, a backlit, high-res, easy-to-read graphical display and flexible customisation options, it is clear why the SoundPoint IP 550 is able to meet the voice communication needs of the most demanding managers and professionals. Efficient Installation and Provisioning The Polycom SoundPoint IP 550 phone is engineered to make installation, configuration and upgrading as simple and efficient as possible. The phone's built-in IEEE 802.3af PoE circuitry and dual-port Ethernet switch allow flexible deployment options and savings on cabling costs. The SoundPoint IP 550 phone supports remote, zero-touch provisioning and can be upgraded from a variety of servers, including FTP, TFTP, HTTP3, or HTTPS3. In order to ensure reliable, uninterrupted performance, the 550 IP phone supports boot and call server redundancy. Broad Interoperability for Maximum Flexibility The Polycom SoundPoint IP 550 desktop phone is tested and certified to deliver comprehensive interoperability and extensive feature support with leading SIP-based call control platforms by BroadSoft, Digium, Sylantro, Interactive Intelligence/Vonexus, and other Polycom Technology and Interoperability Partners. The 550 phone also interoperates with Microsoft Live Communications Server (LCS) 2005 for telephony and presence. Whether the SoundPoint IP 550 phone is deployed in conjunction with an IP PBX or a hosted VoIP service, users can rely on it to fulfil its promise of outstanding sound quality and industry-leading SIP features. P olycom Soundpoint IP 550 Benefits & Features Revolutionary Voice Quality Polycom HD Voice technology provides life-like richness, clarity, and interactivity of voice communications. Advanced Features & Applications Four lines Backlit 320 x 160-pixel graphical grayscale LCD Busy lamp field (BLF) Presence, buddy lists Shared call/bridged line appearance XHTML micro-browser for Web applications Efficient Installation and Provisioning Remote, zero-touch provisioning with support of a variety of servers Integrated IEEE 802.3af Power over Ethernet (PoE) support Broad Interoperability Certified to support a comprehensive range of features with a variety of leading SIP-based IP PBX and Softswitch platforms and to interoperate with Microsoft LCS for telephony and presence 476 SIP / IAX Call Credit 100 100.00 VoIP Call Credit http://www.voipon.co.uk/sip-iax-call-credit-p-476.html http://www.voipon.co.uk/images/100.gif new Availability: Instant Activation Enjoy all the benefits of VoIP with our PSTN termination services with VoIPon Call Credit. What do I get? With VoIPon Call Credit you receive free registration to our SIP subscription service (you get a SIP address) with which you can make PSTN calls and a host of other features: Low UK Call Rates ( more info ) Low International Call Rates ( more info ) Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Scheduled Dialplan ( more info ) SMS Call Back ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) Online Billing Information How does it work? Simply add this item and your credit will be updated accordingly. Call credit will be allocated to your VoIPon account. If this is the first time shopping with us, you can choose to register with VoIPon in the create account page. UK Charges - Summary Call Type Peak* Off-peak Weekend Geographic 1.2 0.9 0.9 Mobile - from 11.4 11.4 11.4 Local rate (0845) 6.0 6.0 6.0 National rate (0870) 7.0 7.0 7.0 * Peak time is 8am to 6pm. You can call ALL these countries for as little as 1.5 pence per minute! Argentina Australia Belgium Canada China Cyprus Denmark France Germany Greece Hong Kong Hungary Israel Italy Luxembourg Netherlands New Zealand Norway Poland Portugal Singapore Spain Taiwan USA Don't Forget ...... The above countries are what we call lo-cost which, at 1.5p per minute is fantastic value BUT, if you are calling to another VoIP user, the call is FREE! Many other countries are much cheaper to call using VoIP, some examples are: Brazil 2.8ppm Finland 2.8ppm Gibraltar 3.7ppm India 6.0ppm Kenya 9.8ppm Malta 5.6ppm Monaco 2.1ppm Nigeria 9.1ppm Russia 3.7ppm Zimbabwe 5.3ppm 478 SIP / IAX Call Credit 50 50.00 VoIP Call Credit http://www.voipon.co.uk/sip-iax-call-credit-p-478.html http://www.voipon.co.uk/images/50.gif new Availability: Instant Activation Enjoy all the benefits of VoIP with our PSTN termination services with VoIPon Call Credit. What do I get? With VoIPon Call Credit you receive free registration to our SIP subscription service (you get a SIP address) with which you can make PSTN calls and a host of other features: Low UK Call Rates ( more info ) Low International Call Rates ( more info ) Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Scheduled Dialplan ( more info ) SMS Call Back ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) Online Billing Information How does it work? Simply add this item and your credit will be updated accordingly. Call credit will be allocated to your VoIPon account. If this is the first time shopping with us, you can choose to register with VoIPon in the create account page. UK Charges - Summary Call Type Peak* Off-peak Weekend Geographic 1.2 0.9 0.9 Mobile - from 11.4  11.4  11.4  Local rate (0845) 6.0 6.0 6.0 National rate (0870) 7.0 7.0 7.0 * Peak time is 8am to 6pm. You can call ALL these countries for as little as 1.5 pence per minute! Argentina Australia Belgium Canada China Cyprus Denmark France Germany Greece Hong Kong Hungary Israel Italy Luxembourg Netherlands New Zealand Norway Poland Portugal Singapore Spain Taiwan USA Don't Forget ...... The above countries are what we call lo-cost which, at 1.5p per minute is fantastic value BUT, if you are calling to another VoIP user, the call is FREE! Many other countries are much cheaper to call using VoIP, some examples are: Brazil 2.8ppm Finland 2.8ppm Gibraltar 3.7ppm India 6.0ppm Kenya 9.8ppm Malta 5.6ppm Monaco 2.1ppm Nigeria 9.1ppm Russia 3.7ppm Zimbabwe 5.3ppm 480 SIP / IAX Call Credit 5 5.00 VoIP Call Credit http://www.voipon.co.uk/sip-iax-call-credit-p-480.html http://www.voipon.co.uk/images/5.gif new Availability: Instant Activation Enjoy all the benefits of VoIP with our PSTN termination services with VoIPon Call Credit. What do I get? With VoIPon Call Credit you receive free registration to our SIP subscription service (you get a SIP address) with which you can make PSTN calls and a host of other features: Low UK Call Rates ( more info ) Low International Call Rates ( more info ) Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Scheduled Dialplan ( more info ) SMS Call Back ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) Online Billing Information How does it work? Simply add this item and your credit will be updated accordingly. Call credit will be allocated to your VoIPon account. If this is the first time shopping with us, you can choose to register with VoIPon in the create account page. UK Charges - Summary Call Type Peak* Off-peak Weekend Geographic 1.2 0.9 0.9 Mobile - from 11.4 11.4 11.4 Local rate (0845) 6.0 6.0 6.0 National rate (0870) 7.0 7.0 7.0 * Peak time is 8am to 6pm. You can call ALL these countries for as little as 1.5 pence per minute! Argentina Australia Belgium Canada China Cyprus Denmark France Germany Greece Hong Kong Hungary Israel Italy Luxembourg Netherlands New Zealand Norway Poland Portugal Singapore Spain Taiwan USA Don't Forget ...... The above countries are what we call lo-cost which, at 1.5p per minute is fantastic value BUT, if you are calling to another VoIP user, the call is FREE! Many other countries are much cheaper to call using VoIP, some examples are: Brazil 2.8ppm Finland 2.8ppm Gibraltar 3.7ppm India 6.0ppm Kenya 9.8ppm Malta 5.6ppm Monaco 2.1ppm Nigeria 9.1ppm Russia 3.7ppm Zimbabwe 5.3ppm 481 SIP / IAX Call Credit 25 25.00 VoIP Call Credit http://www.voipon.co.uk/sip-iax-call-credit-p-481.html http://www.voipon.co.uk/images/25.gif new Availability: Instant Activation Enjoy all the benefits of VoIP with our PSTN termination services with VoIPon Call Credit. What do I get? With VoIPon Call Credit you receive free registration to our SIP subscription service (you get a SIP address) with which you can make PSTN calls and a host of other features: Low UK Call Rates ( more info ) Low International Call Rates ( more info ) Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Scheduled Dialplan ( more info ) SMS Call Back ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) Online Billing Information How does it work? Simply add this item and your credit will be updated accordingly. Call credit will be allocated to your VoIPon account. If this is the first time shopping with us, you can choose to register with VoIPon in the create account page. UK Charges - Summary Call Type Peak* Off-peak Weekend Geographic 1.2 0.9 0.9 Mobile - from 11.4 11.4 11.4 Local rate (0845) 6.0 6.0 6.0 National rate (0870) 7.0 7.0 7.0 * Peak time is 8am to 6pm. You can call ALL these countries for as little as 1.5 pence per minute! Argentina Australia Belgium Canada China Cyprus Denmark France Germany Greece Hong Kong Hungary Israel Italy Luxembourg Netherlands New Zealand Norway Poland Portugal Singapore Spain Taiwan USA Don't Forget ...... The above countries are what we call lo-cost which, at 1.5p per minute is fantastic value BUT, if you are calling to another VoIP user, the call is FREE! Many other countries are much cheaper to call using VoIP, some examples are: Brazil 2.8ppm Finland 2.8ppm Gibraltar 3.7ppm India 6.0ppm Kenya 9.8ppm Malta 5.6ppm Monaco 2.1ppm Nigeria 9.1ppm Russia 3.7ppm Zimbabwe 5.3ppm 483 0800 UK VoIP Freephone Number 65 65.00 Incoming Numbers http://www.voipon.co.uk/0800-uk-voip-freephone-number-p-483.html http://www.voipon.co.uk/images/0800.gif new Availability: Instant Activation For a one off payment of only £65 you can choose to have a UK Freephone 0800 telephone number. This number is free of charge to the caller. Incoming calls cost 2p/min plus any divert costs. With VoIPon 0800 numbers you receive free registration to our SIP/IAX subscription service (you get a SIP address) and a host of other features: Connect via SIP / IAX ( more info - SIP) ( more info - IAX) Voicemail to Email Service ( more info ) Fax to Email ( more info ) Scheduled Dialplan ( more info ) Call divert ( more info ) VoIPon ThreeCall ( more info ) Route to an external provider ( more info ) Personalise VoIP Accounts ( more info ) Check VoIP Status ( more info ) You may also configure this number to point to different targets for daytime, evening and weekend depending on your requirements. DDIs ranges We also provide competitively priced DDIs for ranges of 10-100 numbers. Click here to discuss this with us. 484 Grandstream Handytone 502 Analog Adaptor 45.0000 47.00 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-502-analog-adaptor-p-484.html http://www.voipon.co.uk/images/grandstream-handytone-502.jpg new Availability: In Stock The HT-502 is SIP 2.0 standard compliant and features 2 FXS ports, dual 10M/100Mbps Ethernet ports with integrated high performance NAT router, port status and message waiting LED, and a base stand for vertical positioning. It supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and traditional and advanced telephony features. The HT-502 provides an enhanced level of security and provides automated provisioning using symmetric and asymmetric voice codec/RTP in any call sessions, and supports a broad range of popular voice codecs. Features & Benefits Universal Plug-in-Play (UPnP) 2 FXS ports (RJ11) w/up to 2 SIP account profiles Dual10/100 Mpbs ports (RJ45) w/integrated router Advanced features: -caller ID, call waiting, 3-way conference, blind or attended transfer -call forward, do not disturb, voicemail, MLS voice prompts -T.38 fax, flexible dial plan, direct IP calling Supports Voice Codecs: -G.711(a/u-law), G.723.1, G.729A/B, G.729E -G.726-40/32/24/16 and iLBC T.38 Fax HTTP/HTTPS/Telnet/TFTP Provisioning SIP over TCP/TLS IP connectivity for any phone and fax Web management for easy configuration and installation Offers traditional and advanced telephony features Portable and compact for use at home or on the road Grandstream ATA Comparison Table Feature Handytone 286 Handytone 386 Handytone 486 Handytone 488 Handytone 496 Ethernet Ports 1 x RJ45 (LAN) 1 x RJ45 (LAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) 2 x RJ45 (LAN/WAN) DHCP/NAT Router No No Yes Yes Yes Analogue Phone Port 1 2 1 1 2 Analogue Line Port No No No Yes No PSTN Pass-through Port No Yes Yes Yes No Remote Configuration TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP 485 Grandstream GXP 2020 IP Telephone 79.8000 84.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp-2020-ip-telephone-p-485.html http://www.voipon.co.uk/images/grandstream_2020_sml.jpg new Availability: In Stock GXP2020 Enterprise Level 6 Line IP Phone The Grandstream GXP2020 Enterprise SIP phone has a new stylish design. It addresses the need for an elegant IP handset solution for the executive office at a highly competitive price. The GXP2020 provides excellent voice clarity, an extensive and advanced call feature set, multi-language support, security protection, automated provisioning and broad compatibility with leading SIP platforms. The GXP2020 offers 6 lines, 7 programmable keys, 4 dynamic context-sensitive soft keys, dual switched 10M/100Mbps auto-sensing Ethernet ports with integrated PoE, a backlit 320x160 high resolution graphic LCD with multi-level grey scales, SRTP and TLS (pending) for privacy protection, as well as secure and automated provisioning for mass deployment. The GXP2020 is the second phone in the Grandstream GXP Enterprise SIP Phone series. The GXP2020 is ideal for both the executive office and advanced enterprise users. Features & Benefits Technical Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af) Backlit 320x160 high resolution graphic LCD with multi-level grey scales Call Features 6 lines indicators with individual SIP account profiles Multi-line support of up to 13 call appearance lines with dual-color LED indicators 7 programmable hard keys and 4 XML programmable context-sensitive soft keys Expandable to additional 112 lines through expansion key-modules Support Caller ID display or block, per call or permanent Call waiting, hold, mute, transfer (blind or attended),forward, and more Multi-party conferencing (up to 5-way) And many more enterprise grade features Handset Features Full duplex speakerphone w/advanced audio echo cancellation Secure and automated provisioning for mass deployment Headset jack (2.5 mm and RJ22) SRTP and TLS (pending) fro privacy protection And many more enterprise grade features Specifications SIP Compliant and Protocols: SIP, TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP, etc. Networking Interfaces Dual 10/100mbps Ethernet ports (switched or routed). Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG, AGC high fidelity wideband audio (G.722). Superb Audio Quality Custom Ringtone Software Convert most music files to a Grandstream ringtone. Advanced Functionality: Multi-line support, multi-party conferencing (5-way), multi-language support (MLS), headset enabled, expandable, intercom, AES encryption, etc. 486 Aastra 536M Keypad Expansion Module for 55i and 57i IP Telephone 67.0000 75.00 Aastra VoIP Accessories http://www.voipon.co.uk/aastra-536m-keypad-expansion-module-for-55i-and-57i-ip-telephone-p-486.html http://www.voipon.co.uk/images/aastra_536_exp_module.jpg new Availability: In Stock Aastra 5i Series Expansion Modules for increased Call Control and Flexibility. The Aastra 536M expansion module is designed to increase the power and flexibility of the 5i Series SIP phones. Up to three modules can be used with either the 55i, 57i or 57i CT telephone to create a powerful, feature-rich console option. The 536M module shares power and signaling with the phone, eliminating the need for additional wiring. Intended for receptionists, administrative assistants, call-centre agents, managers and executives who need to monitor and manage a large volume of calls on a regular basis, the 5i Series expansion modules are an intelligent choice for all enterprise IP environments. Key Features Flexible Features - 536M modules are equipped with LED status indicators and support a range of programmable features including Line, Speed Dial, Busy Lamp Field, Bridged Line Appearance, Shared Call Appearance and Do Not Disturb. Simplified Deployment - Sharing power and signaling with the 5i Series phone, expansion modules can be easily added without extra installation or wiring expense. Simplified configuration via WebUI or configuration file from the phone. Expandability - Up to three modules can be used with the 5i Series models 55i, 57i or 57i CT, offering up to 180 keys. Growing businesses can start with a single module and add to this as their needs increase. Product Highlights Up to 9 call appearance lines overthe phone and expansion modules Shared Call and Bridged LineAppearances (SCA & BLA) Programmable Speed dials Busy Lamp Field (BLF) Custom XML feature key support Call park and pickup Directory Do Not Disturb Flash features Callers log * Feature availability dependant on IP telephony systems 487 Aastra 560M LCD & Keypad Expansion Module for 57i IP Telephone 89.0000 101.00 Aastra VoIP Accessories http://www.voipon.co.uk/aastra-560m-lcd-keypad-expansion-module-for-57i-ip-telephone-p-487.html http://www.voipon.co.uk/images/aastra_560_exp_module.jpg new Availability: In Stock Aastra 5i Series Expansion Modules Offer increased Call Control and Flexibility. The two Aastra 5i Series expansion modules are designed to increase the power and flexibility of the 5i Series SIP telephones. Up to three modules can be used with either the 55i, 57i or 57i CT telephone to create a powerful, feature rich console option. Modules share power and signaling with the phone, eliminating the need for additional wiring. Designed for receptionists, administrative assistants, call center agents, power users, and executives who need to monitor and manage a large volume of calls on a regular basis, the 5i Series Expansion Modules provide an intelligent choice for all Enterprise IP environments. Key Features and Benefits: Flexible Features - Both the 536M and 560M modules are equipped with LED status indicators and support a variety of programmable features including Line, Speed Dial, Busy Lamp Field, Bridged Line Appearance, Shared Call Appearance and Do Not Disturb. Simplified Deployment - Sharing both power and signaling with the 5i Series phone, expansion modules can be easily added without additional installation or wiring expense. Simplified configuration via WebUI or configuration file from the phone. Expandability - Up to three modules can be used with the 5i Series models 55i, 57i or 57i CT, offering up to 180 keys. Growing businesses can start with a single module and add more as their needs increase. Feature Highlights* Up to 9 call appearance lines overthe phone and expansion modules Shared Call and Bridged LineAppearances (SCA & BLA) Busy Lamp Field (BLF) Programmable Speed dials Custom XML feature key support Call park and pickup Directory Callers log Do Not Disturb Flash features * Feature availability dependant on IP telephony systems 488 Snom Wireless Headset Adapter 42.0000 44.00 Snom VoIP Accessories http://www.voipon.co.uk/snom-wireless-headset-adapter-p-488.html http://www.voipon.co.uk/images/snom_wifi_sml.jpg new Availability: In stock Wireless headsets are increasingly popular in professional business environments such as call centres - where the extra freedom can make an enormous difference to performing everyday tasks. The snom Headset Adapter , for the control of wireless headsets is the bridge between a professional VoIP phone system and professional wireless headsets. The snom Headset Adapter has an EHS interface which conforms to vendor specific standards as well as the DHSG standard (which enables the electronic reception of calls on the headset itself). When the telephone receives a call, the original ringing tone is signalled in the headset and the call can be answered and terminated on the headset. The snom Headset Adapter is designed specifically for the snom 320, 360, and 370 voIP phones.   Main Features /Specification DHSG Standard No additional power supply required Easy to connect Complete freedom of movement Signaling of ringtone Call acceptance on headset Call termination on headset Perfect integration of firmware Vendor specific EHS protocols The following headsets support electronic hook switch and are recommended by snom: GN Netcom: GN 9350, GN 9120, DHSG, GN 6210 Plantronics: CS 60, CS 65, CS 70 489 Sangoma A101X PCI Express PRI ISDN Card 287.0000 366.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a101x-pci-express-pri-isdn-card-p-489.html http://www.voipon.co.uk/images/sangoma_a101_pci_express_small.gif new Availability: In Stock Introducing the industry's first affordable single port T1/E1/J1 card with available Telco-grade hardware echo cancellation. It's designed for optimum voice support for smaller systems. The A101 is part of Sangoma's family of Advanced Flexible Telecommunications hardware product line it uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world. The A101 supports up to 2.048 Mbps of full duplex data through-put or up to 30 voice calls over a single T1, E1, or J1 line. With Sangoma cards, you can always take advantage of hardware and software improvements, as soon as they become available. The A101, like all cards in Sangoma's AFT family, is field upgradeable with crash-proof firmware. Choose the Sangoma A101D and A101D-X cards with Octasic's DSP hardware and certified algorithms to achieve carrier-grade echo cancellation and Voice Quality Enhancement functions on your open source or even proprietary telephone system. Technical specifications One T1/E1 port with optimum PCI Express or PCI interface for high performance voice and data applications. Line decoding: HDB3, AMI, B8ZS. Framing: CRC-4, Non CRC4, SF, D4. Also compatible with Japan's J1. Support for AsteriskTM, YateTM, and FreeSwitchTM PBX/IVR Projects, as well as other open source and proprietary PBX/Switch/IVR/VoIP gateway applications. All of Sangoma's AFT products use the same base PCI interface card and the same professionally engineered firmware on the same family of Field Upgradeable Gate Arrays. Fully compatible with all commercially available motherboards proper PCI-standard interrupt sharing without manual tuning. A101-X and A101D-X PCI Express: 1 Lane PCI Express bus. Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack mount servers. Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V. Temperature range: 0-50 °C. Autosense compatibility with 5 V and 3.3 V PCI busses. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. Use raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® routing stack is completely independent of TDM voice application for total system reliability. WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25. Optional DSP daughterboard on the A101D G.168-2002 echo cancellation in hardware. 1014 taps/128ms tail per channel on all 256 channels. DTMF decoding and tone recognition. Voice quality enhancement: Octasic music protection, acoustic echo control, and adaptive noise reduction. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/2000/XP. FreeBSD, Open BSD, NetBSD, Solaris.   Voice applications AsteriskTM, YateTM, Open PBX/IVR, FreeSwitchTM, TrixBoxTM, as well as proprietary applications. Line protocols FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022, Class A, CIPSR 24, AFIC-2016, IEC 60950. Technical certifications in Russia, Malaysia, and Australia. Higher level protocols IP/PX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (D.T. and TOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs T1/E1 Status alarm RED: Telco Red Alarm Condition. OOF: Out of Frame. LOS: Receive Loss of Signal. AIS: Alarm Indication Signal. RAI: Remote Alarm Indication (Yellow Alarm). Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Technical certifications in Russia, Malaysia and Australia Production quality ISO 9002 Warranty Five years parts and labor. PLUS 30-day "no questions asked" return policy. 490 Sangoma A101DX PCI Express PRI ISDN Card 446.2000 569.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a101dx-pci-express-pri-isdn-card-p-490.html http://www.voipon.co.uk/images/sangoma_a101_pci_express_small.gif new Availability: In Stock Introducing the industry's first affordable single port T1/E1/J1 card with available Telco-grade hardware echo cancellation. It's designed for optimum voice support for smaller systems. The A101 is part of Sangoma's family of Advanced Flexible Telecommunications hardware product line it uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world. The A101 supports up to 2.048 Mbps of full duplex data through-put or up to 30 voice calls over a single T1, E1, or J1 line. With Sangoma cards, you can always take advantage of hardware and software improvements, as soon as they become available. The A101, like all cards in Sangoma's AFT family, is field upgradeable with crash-proof firmware. Choose the Sangoma A101D and A101D-X cards with Octasic's DSP hardware and certified algorithms to achieve carrier-grade echo cancellation and Voice Quality Enhancement functions on your open source or even proprietary telephone system. Technical specifications One T1/E1 port with optimum PCI Express or PCI interface for high performance voice and data applications. Line decoding: HDB3, AMI, B8ZS. Framing: CRC-4, Non CRC4, SF, D4. Also compatible with Japan's J1. Support for AsteriskTM, YateTM, and FreeSwitchTM PBX/IVR Projects, as well as other open source and proprietary PBX/Switch/IVR/VoIP gateway applications. All of Sangoma's AFT products use the same base PCI interface card and the same professionally engineered firmware on the same family of Field Upgradeable Gate Arrays. Fully compatible with all commercially available motherboards proper PCI-standard interrupt sharing without manual tuning. A101-X and A101D-X PCI Express: 1 Lane PCI Express bus. Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack mount servers. Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V. Temperature range: 0-50 °C. Autosense compatibility with 5 V and 3.3 V PCI busses. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. Use raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® routing stack is completely independent of TDM voice application for total system reliability. WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25. Optional DSP daughterboard on the A101D G.168-2002 echo cancellation in hardware. 1014 taps/128ms tail per channel on all 256 channels. DTMF decoding and tone recognition. Voice quality enhancement: Octasic music protection, acoustic echo control, and adaptive noise reduction. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/2000/XP. FreeBSD, Open BSD, NetBSD, Solaris.   Voice applications AsteriskTM, YateTM, Open PBX/IVR, FreeSwitchTM, TrixBoxTM, as well as proprietary applications. Line protocols FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022, Class A, CIPSR 24, AFIC-2016, IEC 60950. Technical certifications in Russia, Malaysia, and Australia. Higher level protocols IP/PX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (D.T. and TOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs T1/E1 Status alarm RED: Telco Red Alarm Condition. OOF: Out of Frame. LOS: Receive Loss of Signal. AIS: Alarm Indication Signal. RAI: Remote Alarm Indication (Yellow Alarm). Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Technical certifications in Russia, Malaysia and Australia Production quality ISO 9002 Warranty Five years parts and labor. PLUS 30-day "no questions asked" return policy. 491 Sangoma A101D PCI PRI ISDN Card 398.3400 508.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-a101d-pci-pri-isdn-card-p-491.html http://www.voipon.co.uk/images/sangoma_a101_pcix_small.gif new Availability: In Stock Introducing the industry's first affordable single port T1/E1/J1 card with available Telco-grade hardware echo cancellation. It's designed for optimum voice support for smaller systems. The A101 is part of Sangoma's family of Advanced Flexible Telecommunications hardware product line it uses the same high-performance PCI or PCI Express interface that is providing superior performance in critical systems all over the world. The A101 supports up to 2.048 Mbps of full duplex data through-put or up to 30 voice calls over a single T1, E1, or J1 line. With Sangoma cards, you can always take advantage of hardware and software improvements, as soon as they become available. The A101, like all cards in Sangoma's AFT family, is field upgradeable with crash-proof firmware. Choose the Sangoma A101D and A101D-X cards with Octasic's DSP hardware and certified algorithms to achieve carrier-grade echo cancellation and Voice Quality Enhancement functions on your open source or even proprietary telephone system. Technical specifications One T1/E1 port with optimum PCI Express or PCI interface for high performance voice and data applications. Line decoding: HDB3, AMI, B8ZS. Framing: CRC-4, Non CRC4, SF, D4. Also compatible with Japan's J1. Support for AsteriskTM, YateTM, and FreeSwitchTM PBX/IVR Projects, as well as other open source and proprietary PBX/Switch/IVR/VoIP gateway applications. All of Sangoma's AFT products use the same base PCI interface card and the same professionally engineered firmware on the same family of Field Upgradeable Gate Arrays. Fully compatible with all commercially available motherboards proper PCI-standard interrupt sharing without manual tuning. A101-X and A101D-X PCI Express: 1 Lane PCI Express bus. Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Includes high quality, tested RJ45 cables and short 2U mounting clips for installation in 2U rack mount servers. Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V. Temperature range: 0-50 °C. Autosense compatibility with 5 V and 3.3 V PCI busses. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Supports Robbed Bit Channel Associated Signaling (CAS) and ISDN PRI. T1/E1 and fractional T1/E1, multiple channel HDLC per line for mixed data/TDM voice applications. Optimized per channel DMA streams and hardware-level HDLC handling unload the host CPU. Use raw bitstream interfaces to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® routing stack is completely independent of TDM voice application for total system reliability. WANPIPE® supports certified, field-tested, and reliable Frame Relay, PPP, HDLC, and X.25. Optional DSP daughterboard on the A101D G.168-2002 echo cancellation in hardware. 1014 taps/128ms tail per channel on all 256 channels. DTMF decoding and tone recognition. Voice quality enhancement: Octasic music protection, acoustic echo control, and adaptive noise reduction. Operating systems Linux (all versions, releases and distributions from 1.0 up). Windows NT/2000/XP. FreeBSD, Open BSD, NetBSD, Solaris.   Voice applications AsteriskTM, YateTM, Open PBX/IVR, FreeSwitchTM, TrixBoxTM, as well as proprietary applications. Line protocols FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022, Class A, CIPSR 24, AFIC-2016, IEC 60950. Technical certifications in Russia, Malaysia, and Australia. Higher level protocols IP/PX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (D.T. and TOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs T1/E1 Status alarm RED: Telco Red Alarm Condition. OOF: Out of Frame. LOS: Receive Loss of Signal. AIS: Alarm Indication Signal. RAI: Remote Alarm Indication (Yellow Alarm). Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950. Technical certifications in Russia, Malaysia and Australia Production quality ISO 9002 Warranty Five years parts and labor. PLUS 30-day "no questions asked" return policy. 492 IC-Talk Media Manager Soho - MM16 664 664.00 IC Talk IC Talk IP PBX http://www.voipon.co.uk/ictalk-media-manager-soho-mm16-p-492.html http://www.voipon.co.uk/images/ic_talk_mm16.jpg new Availability: In Stock 16 User Solid State IP PBX The MM16-Soho Range is tailored for smaller office / home office environments where space is limited and noise can be an issue. With these criteria in mind we have created the Soho series, it a a small footprint (190mm x 195mm x 83mm) wall mountable system which is solid state (no moving parts) and hence is silent in operation. Despite the small size of the device do not underestimate the power of the system as these are capable of running up to 16 Users (32 User upgrade available 1st quarter of 2007) and a whole wealth of features normally only found on enterprise class systems. Including: 8 Hours of Recording time for voicemails and greetings 6 Months of Call Detail Logging (export to CSV available) 2 x Multiway Conference Rooms (pin code protected) 2 x Queues 30 x Ring Groups Unlimited SIP trunks and loads more... Packs all the punch of a larger business phone system, without the cost! Hardware Specification Power: 230V AC 50Hz. External Power Adaptor max. consumption 80W (average running consumption 12-25W). Interfaces: Telephony Slot (varies dependant upon model), 10/100Mb/s Ethernet RJ45 for LAN connection. USB2.0 for Flash Backup Pen. Protocols: TCP/IP v4 (v6 ready), SIP (Session Initiation Protocol), H.323, IAX2, HTTP1.1 (for administration). Compression Codecs: ADPCM, G.711 (A-Law & &#956;-Law), G.726, G.729 (optional extra), GSM, iLBC, Linear, LPC-10 & Speex. Feature Overview:- Multiple inbound phone lines (from multiple technologies including Analogue, ISDN2 and IP Trunks) IP Phone extensions Native SIP based and also legacy analogue (using a gateway) Call Hold, Transfer (blind or attended), Pickup, Park, Forwarding (on busy, no answer or unconditional) Multiple call queues with position and estimated hold time announcements Multiple call groups for extensions and external phone numbers for call distribution Voicemail per extension accessible by phone, web or email Multi-level automated attendants for routing of calls more efficiently (i.e. press 1 for etc...) Multiple music on-hold tracks and categories using MP3's Call detail reporting Easy web interface for administration Web based Operators console Integrated fax reception (sent to email) Up to 8 hours of recording for messages and announcements Advanced call routing using flexible call rules (route based on number called and person calling) Media Link connect multiple managers together across multiple sites providing one large integrated phone system across multiple sites/locations Remote teleworker support with dynamic agent log-in Conference Server for larger multi-way conferences Interactive extension directory lookup service Support for e164 directory number lookups, so if a number has an Internet connection it will send the call direct for free. Versions Available IP Only Analogue - 2x2 (2 x FXO & 2 x FXS) or 4 x FXO ports Single ISDN2e - Basic Rate Euro ISDN (Euro ISDN (300 102) signalling, TE mode operation) Quad ISDN2e - Basic Rate Euro ISDN (Euro ISDN (300 102) signalling, TE & NT mode operation) Single ISDN30 - Primary Rate Euro ISDN (PRI-NET & PRI-CPE, Euro ISDN, NI1&2, 4ESS (AT&T), 5ESS (Lucent) & DMS100) 493 Patton SmartNode 4552 2 port ISDN BRI Gateway 182.0000 199.00 Patton Inalp Patton ISDN BRI Gateways http://www.voipon.co.uk/patton-smartnode-4552-2-port-isdn-bri-gateway-p-493.html http://www.voipon.co.uk/images/patton_smartnode_sn4552.jpg new Availability: In Stock SmartNode 2 BRI VoIP GW-Router, 1x10/100baseTX WAN, integrated 4-port 10/100bTX LAN switch, 2 voice channels, H.323 or SIP, External UI Power. The SmartNode 4552 enables integration of ISDN network users into their local phone service, a remote PBX and the data/VPN network through a single tightly integrated access device. Connecting to the local ISDN Phone as well as the local PSTN port, Patton's SessionRouter® technology links any standard ISDN telephone or PBX to VoIP while still connecting to legacy PSTN services. Built-in Lifeline support ensures phone connection to the PSTN in the event of failure. Gateway functions use standard CODECs such as G.723 and G.729 as well as industry standard SIP, H.323 and MGCP/IUA to ensure seamless connection and compatibility for all voice services. Broadband network connectivity integrates with any fixed IP, DHCP or PPPoE service. An integrated 10/100 Ethernet LAN switch, with advanced routing features such as NAT, Firewall/ACL, DynDNS as well as optional IPSec VPN, fulfills the requirements of demanding network users. Quality of Service (QoS) features complete the offering with advanced voice prioritization and traffic management. Patton's patent-pending DownStreamQoS® ensures voice without interruptions even over best-effort Internet connections. Features & Benefits 2-channel VoIP Media Gateway and router with dual ISDN BRI/S0 ports with NT and TE support. SessionRouter® with enhanced circuit-switched call routing allows user programmable call handling based on hunt groups, caller/called ID, and time of day. Pass local PSTN call through without packet conversion. SIP, H.323, and MGCP All SmartNodes support industry standard call-control signalling with standards-based compatibility Fast-Ethernet WAN and integrated 4-port 10/100 LAN switch with auto MDI-X. Access router with NAT, Firewall, PPPoE, DHCP, and DynDNS Quality of Service Advanced adaptive traffic management and shaping for maximum voice quality. Voice and data prioritization and DownStreamQoS® Built-in Web based management, SNMP, Command Line Interface and Auto-Provisioning for automated configuration distribution and software upgrades 494 Patton SmartNode 4634 3 port ISDN BRI Gateway 343.0000 380.00 Patton Inalp Patton ISDN BRI Gateways http://www.voipon.co.uk/patton-smartnode-4634-3-port-isdn-bri-gateway-p-494.html http://www.voipon.co.uk/images/patton_smartnode_sn4634.jpg new Availability: In Stock Smartnode 3 BRI VoIP IAD - 4 VoIP Call, Passthrough Relay; H.323 and SIP, Internal UI Power. The SmartNode 4630 series are the multi-port ISDN BRI models of the proven market-leading SmartNode VoIP product family. The available 3 and 5 BRI/So port configurations fit the requirements of small and medium enterprises looking for a cost-efficient way to network PBX systems on multiple sites or connect to a public Internet telephony service. The extra BRI port solves many VoIP network integration problems encountered in real-world installations. The port can synchronize the gateway and provide error-free ISDN data and fax transmissions, and it can be used as a fallback or local-breakout port for optimized call-routing and risk-free operation. With the life-line relay, the port even enables integration of an ISDN emergency terminal powered from the public ISDN. Like every SmartNode, the 4630 Series models are state-of-the-art VoIP gateways that also provide complete access routing and IP security features. Use the SmartNode as CPE or access router on broadband access, and you can benefit from industry leading Quality of Service (QoS) features ensuring a voice quality unmatched by any IP-phone or gateway on the market. The SmartNode 4630 is the solution for service providers and network integrators looking for a VoIP product that matches up to ISDN standards in terms of features and quality. SmartNode products provide seamless network integration, continuous trouble-free operation and cost effective deployment to protect your investments for the future. Features & Benefits 3/5 Ports Quality ISDN VoIP - 3 or 5 ISDN BRI So ports, 4 or 8 low-bandwidth voice or T.38 fax calls. Advanced adaptive traffic management and shaping for maximum voice quality. Voice prioritization and DownStreamQoS Full Telephony Features - SessionRouter allows flexible call routing and numbering plan adaptations, CLIP/CLIR, hold, transfer, and much more. Management & Provisioning - Web-based management, SNMP, command line interface, & auto-provisioning for automated configuration & SW upgrades. Complete Access Routing - Two 10/100 Ethernet ports with auto MDI-X. Access router with NAT, Firewall, PPPoE, DHCP, DynDNS & VPN with IPSec* Full VoIP protocol support - SIPv2, H.323v4, MGCP/IUA, ISDN, DSS1, QSIG*, T.38, fax and modem bypass, DTMF relay. Outstanding Interoperability - Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors, and Asterisk IP-PBX 495 Patton SmartNode 4638 5 port ISDN BRI Gateway 425.0000 459.00 Patton Inalp Patton ISDN BRI Gateways http://www.voipon.co.uk/patton-smartnode-4638-5-port-isdn-bri-gateway-p-495.html http://www.voipon.co.uk/images/patton_smartnode_sn4634.jpg new Availability: In Stock Smartnode 5 BRI VoIP IAD - 8 VoIP Call; Passthrough Relay; H.323 and SIP, Internal UI Power. The SmartNode 4630 series are the multi-port ISDN BRI models of the proven market-leading SmartNode VoIP product family. The available 3 and 5 BRI/So port configurations fit the requirements of small and medium enterprises looking for a cost-efficient way to network PBX systems on multiple sites or connect to a public Internet telephony service. The extra BRI port solves many VoIP network integration problems encountered in real-world installations. The port can synchronize the gateway and provide error-free ISDN data and fax transmissions, and it can be used as a fallback or local-breakout port for optimized call-routing and risk-free operation. With the life-line relay, the port even enables integration of an ISDN emergency terminal powered from the public ISDN. Like every SmartNode, the 4630 Series models are state-of-the-art VoIP gateways that also provide complete access routing and IP security features. Use the SmartNode as CPE or access router on broadband access, and you can benefit from industry leading Quality of Service (QoS) features ensuring a voice quality unmatched by any IP-phone or gateway on the market. The SmartNode 4630 is the solution for service providers and network integrators looking for a VoIP product that matches up to ISDN standards in terms of features and quality. SmartNode products provide seamless network integration, continuous trouble-free operation and cost effective deployment to protect your investments for the future. Features & Benefits 3/5 Ports Quality ISDN VoIP - 3 or 5 ISDN BRI So ports, 4 or 8 low-bandwidth voice or T.38 fax calls. Advanced adaptive traffic management and shaping for maximum voice quality. Voice prioritization and DownStreamQoS Full Telephony Features - SessionRouter allows flexible call routing and numbering plan adaptations, CLIP/CLIR, hold, transfer, and much more. Management & Provisioning - Web-based management, SNMP, command line interface, & auto-provisioning for automated configuration & SW upgrades. Complete Access Routing - Two 10/100 Ethernet ports with auto MDI-X. Access router with NAT, Firewall, PPPoE, DHCP, DynDNS & VPN with IPSec* Full VoIP protocol support - SIPv2, H.323v4, MGCP/IUA, ISDN, DSS1, QSIG*, T.38, fax and modem bypass, DTMF relay. Outstanding Interoperability - Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors, and Asterisk IP-PBX 496 Patton SmartNode 4112/JO/EUI 2 port FXO Analogue Gateway 233.0000 245.00 Patton Inalp Patton FXO Gateways http://www.voipon.co.uk/patton-smartnode-4112joeui-2-port-fxo-analogue-gateway-p-496.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode Dual FXO VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 497 Patton SmartNode 4112/JS/EUI 2 port FXS Analogue Gateway 266.0000 323.00 Patton Inalp Patton FXS Gateways http://www.voipon.co.uk/patton-smartnode-4112jseui-2-port-fxs-analogue-gateway-p-497.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode Dual FXS VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 498 Patton SmartNode SN4114/2JS2JO 2 x FXS, 2 x FXO Analogue Gateway 321.0000 342.00 Patton Inalp Patton FXO FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn41142js2jo-2-x-fxs-2-x-fxo-analogue-gateway-p-498.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 2 FXS & 2 FXO VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 499 Patton SmartNode SN4114/JO/EUI 4 port FXO Analogue Gateway 284.0000 299.00 Patton Inalp Patton FXO Gateways http://www.voipon.co.uk/patton-smartnode-sn4114joeui-4-port-fxo-analogue-gateway-p-499.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 4 FXO VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 500 Patton SmartNode SN4114/JS/EUI 4 port FXS Analogue Gateway 284.0000 299.00 Patton Inalp Patton FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn4114jseui-4-port-fxs-analogue-gateway-p-500.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 4 FXS VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 501 Patton SmartNode SN4116/4JS2JO 4 x FXS, 2 x FXO Analogue Gateway 498.0000 604.00 Patton Inalp Patton FXO FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn41164js2jo-4-x-fxs-2-x-fxo-analogue-gateway-p-501.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 4 FXS & 2 FXO VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 502 Patton SmartNode SN4116/JS/EUI 6 port FXS Analogue Gateway 450.0000 547.00 Patton Inalp Patton FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn4116jseui-6-port-fxs-analogue-gateway-p-502.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 6 FXS VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 503 Patton SmartNode SN4118/4JS4JO 4 x FXS, 4 x FXO Analogue Gateway 568.0000 690.00 Patton Inalp Patton FXO FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn41184js4jo-4-x-fxs-4-x-fxo-analogue-gateway-p-503.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 4 FXS & 4 FXO VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 504 Patton SmartNode SN4118/JS/EUI 8 port FXS Analogue Gateway 515.0000 625.00 Patton Inalp Patton FXS Gateways http://www.voipon.co.uk/patton-smartnode-sn4118jseui-8-port-fxs-analogue-gateway-p-504.html http://www.voipon.co.uk/images/patton_smartnode_sn4110.jpg new Availability: In Stock Smartnode 8 FXS VoIP Gateway, 1x10/100baseT, H.323 and SIP, External UI Power. The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services. The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network. Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice. Features & Benefits 10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port 14 LEDs for System, Ethernet, and Call status Universal 100-240VAC (contact us for 48VDC Power) Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time 2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc. 2 or 4 FXO ports connect to a PBX or the PSTN. Programmable call routing switches between any FXS, FXO, H.323 and SIP interface. Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.729ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality fax over IP. H.323v4 and SIP Signaling are both supported at the same time in one firmware. Maximum interoperability and deployable into any enterprise or carrier softswitch network, or standalone. IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network. Full ToIP® Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID and other call parameters 505 Patton SmartNode SN4960/1E15V/UI PRI 15 VoIP Channels Gateway 1123.0000 1364.40 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49601e15vui-pri-15-voip-channels-gateway-p-505.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 1 T1/E1/PRI VoIP IAD, 2x GigEthernet, 15 VoIP Channels, upgradeable to 30, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS&trade; and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 506 Patton SmartNode SN4960/1E30V/UI 1 x PRI 30 VoIP Channel Gateway 1774.0000 2156.00 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49601e30vui-1-x-pri-30-voip-channel-gateway-p-506.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 1 T1/E1/PRI VoIP IAD, 2x GigEthernet, 30 VoIP Channels; non-upgradeable, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS® and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 507 Patton SmartNode SN4960/4E30V/UI 4 x PRI 30 VoIP Channel Gateway 1774.0000 2516.00 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49604e30vui-4-x-pri-30-voip-channel-gateway-p-507.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 4 T1/E1/PRI VoIP IAD, 2x GigEthernet, 30 VoIP channels; field upgradeable to a max of 60 channels, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS® and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 508 Patton SmartNode SN4960/4E60V/UI 4 x PRI 60 VoIP channel Gateway 3077.0000 3740.00 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49604e60vui-4-x-pri-60-voip-channel-gateway-p-508.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 4 T1/E1/PRI VoIP IAD, 2x GigEthernet, 60 VoIP channels; non-upgradeable, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS® and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 509 Patton SmartNode SN4960/4E96V/UI 4 x PRI 96 VoIP channel Gateway 4143.0000 5036.00 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49604e96vui-4-x-pri-96-voip-channel-gateway-p-509.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 4 T1/E1/PRI VoIP IAD, 2x GigEthernet, 96 VoIP channels; field upgradeable to a max of 120 channels, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS® and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 510 Patton SmartNode SN4960/4E120V/UI 4 PRI 120 VoIP channel Gateway 4736.0000 5756.00 Patton Inalp Patton ISDN PRI Gateways http://www.voipon.co.uk/patton-smartnode-sn49604e120vui-4-pri-120-voip-channel-gateway-p-510.html http://www.voipon.co.uk/images/patton_smartnode_sn4960.jpg new Availability: In Stock SmartNode Hi-Cap 4 T1/E1/PRI VoIP IAD, 2x GigEthernet, 120 VoIP channels; non upgradeable, Internal UI Power. Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. The dual gigabit Ethernet ports connects to the network for the highest throughput with its integrated QoS router. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for a truly integrated voice and data access Like every SmartNode, the SN4960 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features. The SmartNode 4960 is ready for SIP TLS and SRTP through software upgrades. Exclusive DownStreamQoS® and Voice-over-VPN features give the clear advantage of uninterrupted and secure voice communication for any call today. Features & Benefits High Density--Up to 120 simultaneous voice or T.38 fax calls with one to four T1/E1/PRI ports and dual Gigabit Ethernet ports. Use any CODEC or fax on any port, any time. Universal SIP and T.38 support--Softswitch certified signaling support between all T1 RBS CAS, ISDN PRI, Q.SIG, SIP, H.323 and MGCP/IUA protocols. Secure Toll-Quality VoIP--DownStreamQoS and Voice-over-VPN with adaptive traffic management and shaping for maximum voice quality and secure voice communication. Transparent Telephony Features--Handles complex number manipulation and mapping scenarios for most seamless integration with existing infrastructure, CLIP, CLIR, hold, transfer and much more. Management & Provisioning--Web-based management, SNMP, command line interface. Automated provisioning for easy large-scale deployments. 514 Pirelli Discus DP-M30 Dual Mode GSM & Wi-Fi VoIP Phone 295 295.00 Pirelli Pirelli SIP/GSM Telephone http://www.voipon.co.uk/pirelli-discus-dpm30-dual-mode-gsm-wifi-voip-phone-p-514.html http://www.voipon.co.uk/images/pirelli_discus_dp_m30.jpg new Availability: This product has been discontinued by Pirelli The Pirelli Discus DP-M30 is a stylish Multimedia Phone combining Tri-band GSM and VoWLAN telephony based on the SIP standard in a single device. The DP-M30 Pirelli DiscusTM Dual Mode Phone is a stylish clamshell handset combining tri-band GSM and Voice over WLAN (VoWLAN) telephony based on the SIP standard in a single device. In addition to the commonly used data services such as SMS, MMS, E-mail and WAP, the DP-M30 offers rich multimedia features that comprise Internet Browsing, an MP3 music player and the Java applications. Thanks to the product's high throughput WiFi interface, both the voice and the multimedia applications can be enjoyed at a higher speed and a lower cost than standard mobile phones. The DP-M30 leverages on Pirelli's expertise in ToWLAN (Telephony over WLAN) with dual mode phones already pioneered by the DP-L10 model. Through Pirelli's specific optimization of the hardware and firmware characteristics, the DP-M30 reaches an optimal trade-off between cost and key service features such as battery duration, WiFi coverage and voice Quality of Service (QoS). The innovative handset design adds to the DP-M30 a unique esthetical appeal and convenient usability through facilities such as the three hotkeys for MP3 Play on the handset's front cover. The DP-M30 is designed and qualified to operate in a multi-vendor, standards-based environment. Moreover, the DP-M30 can seamlessly integrate within Pirelli's technology bundle for quadruple play services, which comprises: Pirelli's Multiplay Access Gateways, serving as multiservice routers with advanced MIMO WLAN capabilities specifically enhanced for VoIP and video-streaming applications over WiFi. Pirelli's AWR WLAN extender, a self-configuring Wi-Fi repeater supporting extended indoor coverage and seamless WLAN roaming Pirelli's DuMan Management System, an application allowing Over The Air (OTA) remote provisioning and firmware upgrade to streamline service delivery and customer support tasks for VoWLAN services based on dual mode handsets Pirelli Discus DP-M30 Key Features Dual Mode Handset (Wi-Fi/GSM) GSM tri-band (900/1800/1900MHz) Wi-Fi 802.11b/g SIP SMS, MMS, WAP, E-mail, Internet Browser, Java, SyncML, Bluetooth OTAP and Remote Management The Pirelli Discus DP-M30 is a dual-mode GSM/ SIP Wi-Fi handset, combining tri-band capabilities with the support of SIP VoIP via an integrated WLAN 802.11g interface. The Pirelli Discus DP-M30 works like a standard cell phone - any GSM SIM card will work and cell phone features such as SMS, MMS, e-mail, and Internet browsing are easily enabled. When at home or in office, the Pirelli Discus DP-M30 automatically switches to the Wi-Fi enabled fixed connection; making the same mobile services available indoors at a higher speed and a lower cost to you and your customers. The Pirelli Discus DP-M30 is developed on industry-standard protocols such as 3GPP, 802.11 and SIP guaranteeing interoperability. It also features Pirelli's enhanced software that makes dual-mode technology a viable commercial solution for you. The Pirelli software also supports seamless roaming, zero touch registration and remote manageability. The Pirelli Discus DP-M30 is designed and qualified to operate in a multi-vendor, standards-based environment. Pirelli's SIP Wi-Fi service oriented portfolio includes: Pirelli's AWR WLAN extender, a self-configuring Wi-Fi repeater supporting extended indoor coverage and seamless WLAN roaming Pirelli's Multiplay Access Gateways, delivering dual mode phone QoS and service provisioning Pirelli's PMP Management Platform, a TR-069 remote management system which supports the provisioning, diagnostics and software upgrades for the entire Fixed Mobile Convergence suite (DualPhone, AWR WLAN Extender and Multiplay Access Gateways) 516 Snom HS-MM3 Headset 31.5000 35.00 Snom VoIP Accessories http://www.voipon.co.uk/snom-hsmm3-headset-p-516.html http://www.voipon.co.uk/images/snom_hs_mm2_headset.jpg new Availability: In stock Snom Headsets have been updated with a new look and improved ergonomics and usability. With the snom headsets, snom provides a complete product line for use in a variety of Internet telephony environments: SMEs (small and medium-sized enterprises), home offices, and call centers. The snom HS-MM3 headset is the perfect complement to the snom 300 phone - whereas the HS-MM2 headset is designed for the Snom 320 , 360 and snom 370 VoIP phones. The new snom HS-MM2 and HS-MM3 monaural headsets provide users with more headset stability, ease of use, comfort and hands-free convenience in everyday communication. The snom monaural headset has one ear pad which can be worn on either ear. With the noise-canceling microphone, the snom headset keeps the user's voice crystal clear. The snom Headsets are a safe investment for your business, giving users working in telephone-intensive roles more flexibility, freedom and comfort. The use of headsets optimises the operating process and increases overall productivity. 517 Siemens S45 SIP DECT Handset 44.4600 46.00 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-s45-sip-dect-handset-p-517.html http://www.voipon.co.uk/images/siemens_s450ip_handset.jpg new Availability: Discontinued Alternative Siemens phones can be found here The Siemens Gigaset S45 DECT IP phone is a dual mode phone system which allows DECT phones to make VoIP call over an Internet connection, or land line calls over a fixed line connection (such as a BT phone line). One VoIP call and one fixed line call can be made at the same time with multiple handsets. Requires a Siemens Gigaset S450IP. 518 Xorcom Astribank-8 - 8 FXS Channel Bank 242.0000 272.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank8-8-fxs-channel-bank-p-518.html http://www.voipon.co.uk/images/xorcom_astribank_8_8_fxs.jpg new Availability: In Stock Xorcom Astribank-8 Channel Bank; 8 FXS Analog Channels Astribank-8 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 8 FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa. For more information about uses for the I/O ports read our white papers). Astribank-8 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-8 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 and future models of Astribank. Astribank-8 features: 8FXS ports or 8FXO ports 2 relay output ports for peripheral devices (alarm, gate, etc.) 4 input ports for peripheral devices USB-2 direct connection to Asterisk server; no PRI card required Full support of Asterisk conference capability High-quality analog telephony "Plug-&-Play" installation using Xorcom Rapid® Caller ID, message waiting indicator, individual ring patterns Supports international impedance and all analog telephone models Indicator lights USB "Plug and Play" operation Automatic configuration and setup with Xorcom Rapid distribution Technical Specifications: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current Caller ID support     519 Xorcom Astribank-8 - 8 FXO Channel Bank 280.0000 317.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank8-8-fxo-channel-bank-p-519.html http://www.voipon.co.uk/images/xorcom_astribank_8_8_fxo.jpg new Availability: In Stock Xorcom Astribank-8 Channel Bank (19" chassis); 8 FXO Analog Channels Astribank-8 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 8 FXO ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa. For more information about uses for the I/O ports read our white papers). Astribank-8 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-8 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 and future models of Astribank. Astribank-16 features: 8FXS ports or 8FXO ports 2 relay output ports for peripheral devices (alarm, gate, etc.) 4 input ports for peripheral devices USB-2 direct connection to Asterisk server; no PRI card required Full support of Asterisk conference capability High-quality analog telephony "Plug-&-Play" installation using Xorcom Rapid® Caller ID, message waiting indicator, individual ring patterns Supports international impedance and all analog telephone models Indicator lights USB "Plug and Play" operation Automatic configuration and setup with Xorcom Rapid distribution Technical Specifications: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current Caller ID support 520 Xorcom Astribank-16 - 16 FXS Channel Bank 483.0000 544.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank16-16-fxs-channel-bank-p-520.html http://www.voipon.co.uk/images/xorcom_astribank_16_16_fxs.jpg new Availability: In Stock Xorcom Astribank-16 Channel Bank; 16 FXS Analog Channels Astribank-16 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 16 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-16 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-16 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 and future models of Astribank. Astribank-16 features: 16FXS ports or 8FXS/8FXO ports or 16FXO ports Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current Caller ID support 521 Xorcom Astribank-16 - 8 FXS + 8 FXO Channel Bank 518.0000 583.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank16-8-fxs-8-fxo-channel-bank-p-521.html http://www.voipon.co.uk/images/xorcom_astribank_16_8_fxs_8_fxo.jpg new Availability: In Stock Xorcom Astribank-16 Channel Bank; 8 FXS and 8 FXO Analog Channels Astribank-16 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 16 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-16 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-16 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 and future models of Astribank. Astribank-16 features: 16FXS ports or 8FXS/8FXO ports or 16FXO ports Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current Caller ID support 522 Xorcom Astribank-16 - 16 FXO Channel Bank 576.0000 649.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank16-16-fxo-channel-bank-p-522.html http://www.voipon.co.uk/images/xorcom_astribank_16_16_fxo.jpg new Availability: In Stock Xorcom Astribank-16 Channel Bank (19? chassis); 16 FXO Analog Channels Astribank-16 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 16 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-16 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-16 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 and future models of Astribank. Astribank-16 features: 16FXS ports or 8FXS/8FXO ports or 16FXO ports Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current Caller ID support 523 Xorcom Astribank-24 - 24 FXS Channel Bank 710.0000 799.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank24-24-fxs-channel-bank-p-523.html http://www.voipon.co.uk/images/xorcom_astribank_24_24_fxs.jpg new Availability: In Stock Xorcom Astribank-24 Channel Bank; 24 FXS Analog Channels Astribank-24 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 24 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-24 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-24 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-24 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction * Caller ID detection * Pulse dialing support * 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: * 5 REN ringing generator * Programmable AC impedance * Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line * Constant current 524 Xorcom Astribank-24 - 16 FXS + 8 FXO Channel Bank 784.0000 883.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank24-16-fxs-8-fxo-channel-bank-p-524.html http://www.voipon.co.uk/images/xorcom_astribank_24_16_fxs_8_fxo.jpg new Availability: In Stock Xorcom Astribank-24 Channel Bank; 16 FXS + 8 FXO Analog Channels Astribank-24 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 24 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-24 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-24 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-24 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current 525 Xorcom Astribank-24 - 8 FXS + 16 FXO Channel Bank 813.0000 916.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank24-8-fxs-16-fxo-channel-bank-p-525.html http://www.voipon.co.uk/images/xorcom_astribank_24_8_fxs_16_fxo.jpg new Availability: In Stock Xorcom Astribank-24 Channel Bank; 8 FXS + 16 FXO Analog Channels Astribank-24 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 24 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-24 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-24 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current 526 Xorcom Astribank-24 - 24 FXO Channel Bank 843.0000 950.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank24-24-fxo-channel-bank-p-526.html http://www.voipon.co.uk/images/xorcom_astribank_24_24_fxs_24_fxo.jpg new Availability: In Stock Xorcom Astribank-24 Channel Bank; 24 FXO Analog Channels Astribank-24 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 24 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-24 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-24 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-24 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current 527 Xorcom Astribank-32 - 8 FXS + 24 FXO Channel Bank 1059.0000 1194.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank32-8-fxs-24-fxo-channel-bank-p-527.html http://www.voipon.co.uk/images/xorcom_astribank_32_8_fxs_24_fxo.jpg new Availability: In Stock Xorcom Astribank-32 Channel Bank; 8 FXS + 24 FXO Analog Channels Astribank-32 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 32 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-32 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current 528 Xorcom Astribank-32 - 16 FXS + 16 FXO Channel Bank 1007.0000 1134.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank32-16-fxs-16-fxo-channel-bank-p-528.html http://www.voipon.co.uk/images/xorcom_astribank_32_16_fxs_16_fxo.jpg new Availability: In Stock Xorcom Astribank-32 Channel Bank; 16 FXS + 16 FXO Analog Channels Astribank-32 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 32 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-32 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current 529 Xorcom Astribank-32 - 24 FXS + 8 FXO Channel Bank 961.0000 1082.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank32-24-fxs-8-fxo-channel-bank-p-529.html http://www.voipon.co.uk/images/xorcom_astribank_32_24_fxs_8_fxo.jpg new Availability: In Stock Xorcom Astribank-32 Channel Bank; 24 FXS + 8 FXO Analog Channels Astribank-32 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 32 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially ?Plug & Play?, unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-32 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications ? FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications ? FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) ? 3 REN for 4,000 feet line Constant current 530 Xorcom Astribank-32 - 32 FXS Channel bank 948.0000 1027.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank32-32-fxs-channel-bank-p-530.html http://www.voipon.co.uk/images/xorcom_astribank_32_32_fxs.jpg new Availability: In Stock Xorcom Astribank-32 Channel Bank; 32 FXS Analog Channels Astribank-32 is the easy and professional way to connect Asterisk severs to the world of traditional telephony. It is a smart channel bank for an Asterisk PBX that provides 32 FXO/FXS ports, 4 input ports and 2 output ports (I/O ports are used for operating external devices via the telephone and vice versa). Astribank-32 connects to the Asterisk server via a normal USB-2 device and does not require shutting down the machine and opening panels. Astribank-32 has an architecture that was designed for maximum ease in integration. When used with the Xorcom Rapid distribution or Xorcom TS-1 it is practially "Plug & Play", unlike any other solution on the market. Xorcom's driver for Astribank is now officially part of Zaptel. Digium announced the release of Zaptel 1.2.4 for Asterisk in mid-February. Among the changes and improvements, they listed the new driver for Astribank-8 (the first member of the Astribank family) and future models of Astribank. Astribank-32 features: Flexible port assignment: 32 FXS ports, or 16 FXS + 16 FXO ports, or 24 FXS + 8 FXO ports, or 8 FXS + 24 FXO ports Partial assemblies, known as Astribank-24 and Astribank-16, are also available in a 19" enclosure. Built-in echo suppression (FXS and FXO) Native Zaptel device (drivers included in Asterisk distribution) 2 output ports (relays) enable external units activation from the dial plan 4 input ports enable automatic dialing triggered by external events Compatible with international line impedances 100% solid state design, no moving parts Robust heavy-duty design Technical Specifications - FXO Module: * 80 dB dynamic range for transmitting and receiving 3 micro-Ampere on-hook current Programmable digital hybrid for near-end echo reduction Caller ID detection Pulse dialing support 12 Volts operation (110-240 Volts external power supply included) Technical Specifications - FXS Module: 5 REN ringing generator Programmable AC impedance Up to 6,000 feet line length (1 REN) - 3 REN for 4,000 feet line Constant current 531 Xorcom Astribank-BRI - 2 Port Channel Bank 222.0000 249.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-2-port-channel-bank-p-531.html http://www.voipon.co.uk/images/xorcom_astribank_bri_2_port.jpg new Availability: In Stock Xorcom Astribank-BRI Channel Bank; 2 Basic Rate Interface ports Astribank-BRI brings the ease and flexibility of the Astribank USB Channel Bank for Asterisk to the world of ISDN-BRI. Just like other Astribank family members, Astribank-BRI connects to Asterisk systems via a USB 2.0 port. The unit comes in 2, 4 and 8 port configurations (4, 8 and 16 voice channels, respectively). The BRI module may also be integrated into the Astribank-32 (19'' rack-mountable chassis) units, to provide a single unit featuring both BRI and analog ports. Astribank-BRI features: 2, 4 or 8 ports (4, 8 or 16 voice channels respectively) External USB 2.0 connection to Asterisk server Each port configurable to NT/TE by user 100% solid state device, no moving parts Robust, heavy-duty design 532 Xorcom Astribank-BRI - 4 Port Channel Bank 270.0000 304.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-4-port-channel-bank-p-532.html http://www.voipon.co.uk/images/xorcom_astribank_bri_4_port.jpg new Availability: In Stock Xorcom Astribank-BRI Channel Bank; 4 Basic Rate Interface ports Astribank-BRI brings the ease and flexibility of the Astribank USB Channel Bank for Asterisk to the world of ISDN-BRI. Just like other Astribank family members, Astribank-BRI connects to Asterisk systems via a USB 2.0 port. The unit comes in 2, 4 and 8 port configurations (4, 8 and 16 voice channels, respectively). The BRI module may also be integrated into the Astribank-32 (19'' rack-mountable chassis) units, to provide a single unit featuring both BRI and analog ports. Astribank-BRI features: 2, 4 or 8 ports (4, 8 or 16 voice channels respectively) External USB 2.0 connection to Asterisk server Each port configurable to NT/TE by user 100% solid state device, no moving parts Robust, heavy-duty design 533 Xorcom Astribank-BRI - 8 Port Channel Bank 370.0000 417.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-8-port-channel-bank-p-533.html http://www.voipon.co.uk/images/xorcom_astribank_bri_8_port.jpg new Availability: In Stock Xorcom Astribank-BRI Channel Bank; 8 Basic Rate Interface ports Astribank-BRI brings the ease and flexibility of the Astribank USB Channel Bank for Asterisk to the world of ISDN-BRI. Just like other Astribank family members, Astribank-BRI connects to Asterisk systems via a USB 2.0 port. The unit comes in 2, 4 and 8 port configurations (4, 8 and 16 voice channels, respectively). The BRI module may also be integrated into the Astribank-32 (19'' rack-mountable chassis) units, to provide a single unit featuring both BRI and analog ports. Astribank-BRI features: 2, 4 or 8 ports (4, 8 or 16 voice channels respectively) External USB 2.0 connection to Asterisk server Each port configurable to NT/TE by user 100% solid state device, no moving parts Robust, heavy-duty design 534 Patton M-ATA-1/E Micro Analog Telephone Adapter;1 x FXS RJ11; 1 73.3 73.30 Patton Inalp Patton Analog Adaptor http://www.voipon.co.uk/patton-mata1e-micro-analog-telephone-adapter1-x-fxs-rj11-1-p-534.html http://www.voipon.co.uk/images/patton__micro_adapter.jpg new Availability: In Stock The SmartLink Micro Analog Telephone Adapter provides connectivity for analog phones and faxes to a home, home office or corporate LAN. Connecting to any analog phone, fax or PBX, the SmartLink product is ay cost effective solution for small offices and telecommuters to access Internet-based telephone services and corporate intranet systems across established LAN and Internet connections like xDSL and cable modems. The M-ATA provides one Ethernet (RJ-45) port and one FXS (RJ-11) analog phone port for quick and easy interconnection to the local LAN. LEDs show at-a-glance the status of the system, LAN, WAN, and phone ports. A full suite of IP features (DHCP, NAT/PAT) are available to maximize universal connectivity. VLAN tagging and prioritization enables voice traffic to be handled before data traffic insuring higher quality voice calls. Support for PPPoE tunneling simplifies extending corporate intranet services to telecommuters. The user friendly web interface offers two levels of configuration; level one covers basic subscriber specific parameters, level two offers advanced settings for the transport network. Configuration and firmware can be downloaded from a centralized TFTP or HTTP server. The M-ATA is SIP standard compliant. Analog phones attached to the SmartLink can use advanced calling features such as call forwarding, caller ID, 3-way calling, call holding, call retrieval and call transfer. Features & Benefits Ultra-Miniature--Smallest full-function analog telephone adapter available today! Supports Over 20 Voice Calling Features--Call waiting, call conference, caller ID, hotline, distinctive ring and more! NAT, DHCP, PPPoE--Provides maximum connectivity across firewalls and transport networks. SIP Signaling--Deploy into any multimedia, interactive, or softswitch network with the leading call and session signaling protocol. Toll Quality CODECs & T.38 Fax--Use standard G.711 or G.726 CODECs for toll-quality voice, or G.723 or G.729 for low-bandwidth applications. Centralized Management--HTTP/SNMP manageable from any location. 535 Polycom Power Supply for Soundpoint IP320 and IP330 8.4500 8.45 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-power-supply-for-soundpoint-ip320-and-ip330-p-535.html http://www.voipon.co.uk/images/polycom_psu-sml.jpg new Availability: In Stock Polycom PSU Universal Power Supply for Polycom Soundpoint IP320, IP330 VoIP phones. 536 Pika Low Density Analog Board 539 539.00 Pika Technologies Pika Analog Cards http://www.voipon.co.uk/pika-low-density-analog-board-p-536.html http://www.voipon.co.uk/images/pika_low_density_analog_board.jpg new Availability: In Stock PIKA for Asterisk Low Density Analog Boards with DSP based echo cancellation plug-in to computer systems to connect your applications to both TDM and IP networks. They are a 4-port, half-length boards with a PCI interface that supports the new universal 3.3 volt power bus. With 20 proven years experience in the board market, PIKA has evolved its TDM interface cards to fully integrate into an Asterisk Open Source PBX application and provides the most reliable choice for your Asterisk application. Features & Benefits Single slot PCI card (PCI - x compatible) Software configurable as 4 loop start interfaces or 3 loop start plus one headset DSP based echo cancellation for superior quality 2 ports fax included at no additional cost Installation is as simple as downloading and installing the Pika channel driver for Asterisk and plugging in the boards! Pika provides no charge technical support to help you get your Asterisk application up and running Technical Specifications Analog line circuits - 4 loop start Host interface bus type - PCI (PCI - x compatible) DSPs: On-board DSPs - 1 (see note) Chip Type - Motorola 56303 Memory - 128k Clock Speed - 100MHz Instruction Speed - 100 MIPs Supported OS - Red Hat Enterprise, 4 Fedora Core 4, SuSE 10, AstLinux Host Interface Loop Start Trunk Interfaces (CO Interface Circuit) - On-hook audio reception DC resistance - North America: 360-140 ohms over 15-120 mA typical, Euro version: 470-154 ohms over 14-130 mA typical Network Interface - RJ14 connectors / RJ22 for headset/handset Loop range - 0-2000 ohms AC impedance - 600 ohms ( North America ) or complex (Euro version) Supervision - loop current drop, battery reversal, ringing Signaling - off-hook, flash, DTMF Loop current range - North America: 15-120 mA Euro version: 14-130 mA Compliance and Capabilities - FCC part 15 and FCC Part 68, Industry Canada CS-03, CSA C22.2 no. 950 NRTL/C, TBR21, EU 55022: 1998 Class B, EU 55024:1998, EU 60950:1992, 2002 / 95 / EC RoHS 6 DSP - Motorola 56303 DSP, Software reset on per DSP basis Media Capabilities - DTMF, tone, speech detection DTMF, tone generation Fax Country Approvals - North America, European Union Power Requirements - 355 mA Operating Temperature: 0 °C to +60 °C Storage Temperature: -20 °C to +85 °C Humidity, Non-condensing - 5% to 95% Mean Time Between Failure (MTBF) - North America variant 47 years EURO variant 45 years Warranty Pika provides 3 year warranty on all boards Product Support Pika provides no charge support on all products. Visit http://www.pikatechnologies.com/contact/index.htm ? 537 Pika T1/E1 Span License (30 ports) 101 101.00 Pika Technologies Pika PRI Cards http://www.voipon.co.uk/pika-t1e1-span-license-30-ports-p-537.html http://www.voipon.co.uk/images/pika_pri_pci_card.jpg new Availability: In Stock Unparalleled Flexibility Every T1/E1 gateway card ships from Pika with hardware capacity for up to 4 spans. At initial order,1, 2, 3 or 4 T1/E1 spans are enabled. However, should your Asterisk system grow over time, you can expand the port capacity by installing a cost effective span upgrade license (shipped to you by email). How simple can it get! 538 Pika Fax expansion port - Per port, for 3rd and 4th port 51 51.00 Pika Technologies Pika Analog Cards http://www.voipon.co.uk/pika-fax-expansion-port-per-port-for-3rd-and-4th-port-p-538.html http://www.voipon.co.uk/images/pika_low_density_analog_board.jpg new Availability: In Stock Add a 3rd or 4th fax expansion port. 539 Sangoma S5141 Dual Serial Card 352 352.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-s5141-dual-serial-card-p-539.html http://www.voipon.co.uk/images/sangoma_s5141_dual_serial.jpg new Availability:  Obsolete Please note the Sangoma A142 Data Card replaces the Sangoma S5141 card Technical Specifications Primary serial V.35/X.21/RS232 port to 4Mbps. Secondary serial V.35/X.21/RS232 port to 512kbps. Power: 550mA at +5v, 60Ma at +-12v. PCI 32 bit (5v and 64 bit (3.3v) compatible. Temperature range: 0 - 45C. All set-up and configuration is in software or by machine BIOS. Dimensions: 144mm x 99 mm. Features added to support Datascope applications All modem control lines are monitored. Detection of the presence or absence of TX and RX clock signals, and measurement of the clock rates. Either monitoring only or simulation (transmit and receive). Monitoring or simulation of ATM or HDLC at line speeds above 2Mbps, BSC at line speeds to 128kbps, Asynch to 256kbps, and raw unformatted bit streams to 2Mbps. Time stamps with a resolution of 100 microseconds or better to allow accurate sequencing of events. Each character can be individually time stamped. Serial interfaces RS232, V.35, X.21, RS422 EIA530 supported on Primary and Secondary ports. Clocking: Internally generated or external at line speeds to 2Mbps. NRZ, NRZi, FM0, FM1, Manchester encoding. Both ports are RS485 capable, supporting multipoint lines. Line protocols ATM, Frame Relay, X.25, HDLC, PPP, SS7, BSC Point-to-Point, BSC 3270, SDLC, Transparent bit-stream. Operating systems Windows® 2000, Windows® XP, Windows® 9x, Windows® ME, Linux (all versions, releases and distributions from 1.0 up), FreeBSD, Open BSD, NetBSD. Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty Three years parts and labour. Certification FCC Part 15 Class A, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 540 Sangoma S5142 Quad Serial Card 502 502.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-s5142-quad-serial-card-p-540.html http://www.voipon.co.uk/images/sangoma_s5142_quad_serial.jpg new Availability: Obsolete Please note the Sangoma A144 Data Card replaces the Sangoma S5142 card Technical Specifications 2 x Primary serial V.35/X.21/RS232 port to 4Mbps. 2 x Secondary serial V.35/X.21/RS232 port to 512kbps. All four ports supported by one DB-78 connector Card takes only 1 PCI slot to support 4 ports. Power: 550mA at +5v, 60Ma at +-12v. PCI 32 bit (5v and 64 bit (3.3v) compatible. Temperature range: 0 - 45C. All set-up and configuration is in software or by machine BIOS. Dimensions: 147mm x 99 mm. Features added to support Datascope applications All modem control lines are monitored. Detection of the presence or absence of TX and RX clock signals, and measurement of the clock rates. Either monitoring only or simulation (transmit and receive). Monitoring or simulation of ATM or HDLC at line speeds above 2Mbps, BSC at line speeds to 128kbps, Asynch to 256kbps, and raw unformatted bit streams to 2Mbps. Time stamps with a resolution of 100 microseconds or better to allow accurate sequencing of events. Each character can be individually time stamped. Serial interfaces RS232, V.35, X.21, RS422 EIA530 supported on Primary and Secondary ports. Clocking: Internally generated or external at line speeds to 2Mbps. NRZ, NRZi, FM0, FM1, Manchester encoding. Both ports are RS485 capable, supporting multipoint lines. Line protocols ATM, Frame Relay, X.25, HDLC, PPP, SS7, BSC Point-to-Point, BSC 3270, SDLC, Transparent bit-stream. Operating systems Windows® 2000, Windows® XP, Windows® 9x, Windows® ME, Linux (all versions, releases and distributions from 1.0 up), FreeBSD, Open BSD, NetBSD. Higher level protocols IP/IPX over Frame Relay/ PPP/ HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty Three years parts and labour. Certification FCC Part 15 Class A, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 541 Sangoma S518 ADSL Card 75.0000 79.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-s518-adsl-card-p-541.html http://www.voipon.co.uk/images/sangoma_s518_adsl.jpg new Availability: In stock Technical Specifications Designed and built to server standards of performance and reliability. PCI 2.2 bus interface, 64 bit (3.3v) and 32 bit (5v) compliant. Self-configuring and line optimizing. Power: 520mA at 5v. MTBF: > 100,000 hours. Temperature range: 0 to 45C. All set-up and configuration is in software or by machine BIOS. Dimensions: 140mm x 90 mm. Will connect properly to any current Central Office DSLAM on a POTS line in North America , providing at least the same performance as the Telco recommended hardware. ADSL Standards ITU G.992.1 (G.DMT) ITU G.992.2 (G.Lite) ITU G.992 Annex A, Annex C ANSI T1.413 Issue 2. Line protocols PPP over ATMPPP over Ethernet, Ethernet over ATM, and IP over ATM. Operating systems Windows® 2000, Windows® XP, Windows® 9x, Windows® ME, Linux (all versions, releases and distributions from 1.0 up). Higher level protocols IP/IPX over Frame Relay/ PPP/HDLC/ X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Warranty Three years parts and labour. Certification FCC Part 15 Class A, FCC Part 68, CE. Diagnostic Tools WANPIPEMON, SNMP, System logs. Production quality ISO 9002 542 Sangoma A500 BRI Card PCI 146.0000 190.00 Sangoma Sangoma BRI Cards http://www.voipon.co.uk/sangoma-a500-bri-card-pci-p-542.html http://www.voipon.co.uk/images/sangoma_a500_bri.jpg new Availability: In stock Sangoma built its business by designing hardware that simply works, the first time. We have taken the time to ensure our BRI solution delivers. The Sangoma A500 S/T BRI Interface Card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional Octasic® Telco-grade, hardware echo cancellation. A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable. Finally, a BRI card that upholds Sangoma's high standards of quality in engineering and untiring product support. Architecture The A500 consists of a Remora® BRI daughterboard mounted on the AFT PCI card. The Remora® BRI card has three sockets, each of which can accept an S/T BRI module. One S/T BRI module has two S/T four wire interfaces, which support TE or NT modes of operation. Changing modes requires no jumpers simply invert the module. Up to three additional Remora® daughterboards can be mounted in empty slot positions beside the A500 assembly. These are connected to the A500 by a special backplane bus connector. Technical Specifications From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers simply invert the colour-coded module. Supports Asterisk®, Yate®, FreeSwitch®, CallWeaver®, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single synchronous PCI and PCI Express interface for all 24 BRI interfaces. Six ports per RemoraTM card. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5 V and 3.3 V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3 V or 5 V. Temperature range: 0 - 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested, and reliable Frame Relay, PPP, HDLC and X.25. 543 Sangoma E1 / T1 Line Tapping System 150 150.00 Sangoma Sangoma PRI Cards http://www.voipon.co.uk/sangoma-e1-t1-line-tapping-system-p-543.html http://www.voipon.co.uk/images/sangoma_t1_e1_tapping_system.jpg new Availability: In Stock Line monitoring using the Sangoma AFT system As requirements for security, customer service and legal responsibilities have increased, it has become important to be able to monitor and record telephone conversations. Sangoma?s T1/E1 tapping solution is a high performance, robust and inexpensive toolkit that gives you complete access to the T1/E1 voice and signaling interfaces. Based on the popular A102, A104 and A108 AFT cards, the system supports the tapping of one, two and four T1/E1 lines per card respectively. A convenient Tap Connection Adapter is available, which allows the data transport line to be hard wired with strain relief. The tapping connections are then simply connected to the RJ45 connectors provided. Both normal impedance and high impedance (high sensitivity) modes are supported. Architecture The Sangoma Line Tapping Solution is comprised physically of an A102 (dual port) A104 (quad port) or A108 (octal port) T1/E1 card, standard cabling, and a PN 633 Tap Connection Adapter. The lines to be tapped are hard wired into the Tap Connection Adapter. Straight cables are connected between the Tap Connection Adapter and a pair of ports on the T1/E1 card, allowing both sides of the line to be monitored simultaneously. The system can be used in both standard impedance and high impedance modes, high impedance being used to ensure that no accidental short of the monitoring lines will affect the flow of T1/E1 traffic. Technical Specifications Supports tapping from one to four T1/E1 ports per card. Includes low-level API under Windows and Linux, which supports access to the G.711 data streams directly, as well as access to D channel HDLC traffic for interpretation of the PRI control packets. Both PRI and RBS-type signaling are supported. Line impedance: 100/120 ohms standard T1/E1 or 1200 ohms high impedance. Optional board-based DSP for DTMF detection and tone recognition. Highly-optimized driver allows tapping of up to 16 spans on a single PC with minimal system load. Connection to Tap Connection Adapter uses a standard straight RJ45 cable. Tapping solution available in PCI (5 V), PCI-X (3.3 V), and PCI-Express form factors. Card dimensions: 2U height form factor: 290 mm x 55 mm for use in a 2U chassis. Short 2U compatible mounting clips available for installation in 2U rackmount servers. 32-bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Compatible with all commercially available motherboards with full IRQ sharing with other PCI devices. Intelligent hardware. Downloadable Field Programmable Gate Array programming with multiple operating modes. Field upgradable so that new features related to voice and/or data can be added when they become available. Power: 800 mA peak, operational 300 mA max at +3.3 V or 5 V. Temperature range: 0 ? 50 ?C. Operating Systems Linux (all versions, releases and distributions from 1.0 up) Windows NT/2000/XP Diagnostic Tools WANPIPEMON, SNMP, System logs Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CISPR 24, AFIC-S016, IEC 60950. Technical certifications in Russia, Malaysia, Australian A-Tick Production quality ISO 9002 Warranty Five years parts and labour. 544 Pirelli Smartphone DP SW20 229 229.00 Pirelli Pirelli SIP/GSM Telephone http://www.voipon.co.uk/pirelli-smartphone-dp-sw20-p-544.html http://www.voipon.co.uk/images/pirelli_dual_smartphone_dp_sw20.jpg new Availability: In stock Discus Windows Dual Mode Smartphone DP-SW20 Pirelli's DP-SW20 is a compact Smartphone combining quad-band GSM and VoWLAN telephony based on the SIP standard in a single device. The DP-SW20 Pirelli Discus Dual Mode Phone is a compact smart-phone combining quad-band GSM and Voice over WLAN (VoWLAN) telephony based on the SIP standard in a single device. Based on the powerful Windows Mobile operating system, the DP-SW20 offers the convenience of a whole range of applications such as SMS, MMS, Outlook, e-mail, Tasks and Calendar management, Internet Ecplorer Browsing, MSN meddagin, Voice Recorder and a number of ulilities offered by the Windows environment. When at home, in the office, or under a hotspot coverage, the DP-SW20 allows to leverage on a Wi-Fi connection, making voice and data services availablle at a higher speed and at a lower cost. The DP-SW20 is designed and qualified to operate in a multi-vendor, standards-based environment. It also features Pirelli's software enhancements on Quality of Service, manageability and advanced Fixed Mobile Converged servicesm such as Voice Call Continuity. Moreover, the DP-SW20 can seamlessly integrate within Pirelli's technology bundle for quadruple play services, which comprises: Pirelli's Multiplay Access Gateways, serving as multiservice routers with advanced MIMO WLAN capabilities specifically enhanced for VoIP and video-streaming applications over WiFi. Pirelli's AWR WLAN extender, a self-configuring Wi-Fi repeater supporeting extended indoor coverage and seamless WLAN roaming. Pirelli's DuMan Management System, an application allowing Over The Air (OTA) remote provisioning and firmware upgrade to streamline service delivery and customer support tasks for VoWLAN services based on dual mode handsets. Summary of Features Dual Mode Phone (Wi-Fi/GSM) Windows Mobile GSM Quad-band (850/900/1800/1900MHz) SIP VoIP Internet Browser, SMS, MMS, WAP, E-mail, Java OTAP and Remote Management Feature Description Oprating System: Windows Mobile Cellular Protocols / Bands: GSM Quad/Band 850/900/1800/1900MHz GPRS Class B, multi-slot class 10 WLAN Features: Wi-Fi 802.11b/g, 2.4GHz WEP, WPA-PSK TKIP, WPA2-PSK AES EAP-TLS/SIM/PEAP Fast AP Roaming WME Qos VPN VoIP Codec: G.711, G.729 a/b Form Factor: Bar Type (107x45x16 mm) LCD Displays: 1 Main display 1.9" QCIF-TFT, 240x320p, 262K colour depth Camera: 1.3M pixel VGA - Camcorder Applications: SMS, MMS 1.2, WAP 2.0, JavaVM, Active Sync, Internet Explorer, MSN, E-mail (Outlook email, POP/SMTP, push email) Windows Media Player: (MP3, WMV, MPEG4) - QCIF Voice Recorder Ring-Tones Formats: Midi, Wav MP3 AAC AAC+ Connectors/Slots: Charging and data transfer Earphone Jack: for wired stereo handset Micro SD Connector Games: 2 games (Excluded Java Game) Language: English / Italian Configurability and Provisioning: OMA OTA Regulatory Conformance: CE Mark European Regulatory R&TTE directive: - EN 301 489-1/-7 (GSM EMC) - EN 301 489-17 (WLAN EMC) - EN 301 511 (GSM RF) - EN 300 328 (WLAN 802.11b/g RF) - EN 60950 (Safety) - EN 50360/50361 (SAR) Sales Package: - 1 x Dualphone DP-SW20 - 1 x Headset with earphone / microphone - 1 x Li-Ion Battery - 1 x travel charger with mini-USB plug - 1 x mini-USB cable Optional: - 1 x CD with PC utility - 1 x User manual 545 Digium Wildcard TE220 PCI Express ISDN PRI Card 399.0000 446.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te220-pci-express-isdn-pri-card-p-545.html http://www.voipon.co.uk/images/digium_te220_pci_card.jpg new Availability: In Stock Digium TE220 Two Port T1/E1 PCI Express Asterisk Interface Digium's TE220 PCI Express card provides termination of up to 60 channels of voice or data across two E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-port or per-card basis, the TE220 allows E1 and T1 circuits to be mixed with full channel synchronisation. Supporting PCIe x1, the TE220 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without considerations for voltage selection or lane size. The TE220 can be combined with Digium's VPMOCT064 Octasic DSP-based echo cancellation module. The VPMOCT064 provides the G.168 algorithm which is considered a benchmark for echo cancellation and performs 128ms (1024 taps) of echo cancellation across all 60 channels in E1 mode or all 48 channels in T1/J1 modes. Bundled with the VPMOCT064, the product SKU is TE220B. Digium has designed the TE220 to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk. The TE220 supports industry standard telephony protocols including North American and European Primary Rate signalling as well as standard Robbed Bit, Channel Associated Signalling in addition to standard PPP, HDLC, and Frame Relay data modes. 546 Digium Wildcard TE220B PCI Express ISDN PRI Card with Echo Cancellation 608.0000 676.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te220b-pci-express-isdn-pri-card-with-echo-cancellation-p-546.html http://www.voipon.co.uk/images/digium_te220_pci_card.jpg new Availability: In Stock 2 Port T1/E1 PCI Express Asterisk Card - with Echo Cancellation The Digium TE220 PCI Express card provides termination of up to 60 channels of voice or data across two E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-card or per-port basis, the Digium TE220 allows E1 and T1 circuits to be mixed with full channel synchronisation. Supporting PCIe x1, it can be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 - without considerations for voltage selection / lane size. The TE220 can be combined with the Digium VPMOCT064 Octasic DSP-based echo cancellation module. The VPMOCT064 provides the G.168 algorithm, noted as a benchmark for echo cancellation, and performs 128ms (1024 taps) of echo cancellation across all 60 channels in E1 mode or all 48 channels in T1/J1 modes. When combined with the VPMOCT064, the product is known as TE220B. Digium has designed the TE220 to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk. The TE220 supports industry standard telephony protocols including North American and European Primary Rate signalling as well as standard Robbed Bit, Channel Associated Signalling in addition to standard PPP, HDLC, and Frame Relay data modes. 547 Digium Wildcard TE420 PCI Express ISDN PRI Card 656.0000 729.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te420-pci-express-isdn-pri-card-p-547.html http://www.voipon.co.uk/images/digium_te420_pci_express_ca.jpg new Availability: In Stock Digiums TE420 4 Port T1/E1 PCI Express Asterisk Interface   The TE420 PCIe card provides termination of up to 120 channels of voice/data across four E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-card/per-port basis, the TE420 allows E1 and T1 circuits to be mixed with full channel synchronisation. The Digium TE420 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without worrying about voltage selection or lane size. The TE420 can be combined with Digium's VPMOCT128 Octasic DSP-based echo-cancellation module. The VPMOCT128 provides the benchmark G.168 algorithm and performs 128ms (1024 taps) of echo cancellation across all 96 channels in T1/J1 modes or all 128 channels in E1 mode. Bundled with the VPMOCT128, the product is known as TE420B. The Digium TE420 has been designed to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk. The TE420 supports industry standard telephony protocols including North American and European Primary Rate signalling as well as standard Robbed Bit, Channel Associated Signalling in addition to standard PPP, HDLC, and Frame Relay data modes. 548 Digium Wildcard TE420B PCI Express ISDN PRI Card with Echo Cancellation 999.0000 1180.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te420b-pci-express-isdn-pri-card-with-echo-cancellation-p-548.html http://www.voipon.co.uk/images/digium_te420_pci_express_ca.jpg new Availability: In Stock Four Port T1/E1 PCI Express Asterisk Interface with Echo Cancellation   Digium's TE420 PCI Express card provides termination of up to 120 channels of voice or data across four E1, T1, or J1 interfaces in a PCIe x1 form factor. Selectable on a per-port/per-card basis, the TE420 allows E1 and T1 circuits to be mixed with full channel synchronisation. Supporting PCIe x1, the TE420 may be used in any available PCIe 1.0 compliant slot - 1x, x4, x8, and x16 without worries about voltage selection or lane size. The TE420 can be combined with the Digium VPMOCT128 Octasic DSP-based echo cancellation module (not shown here). The VPMOCT128 provides the benchmark G.168 algorithm for echo cancellation and performs 128ms (1024 taps) of echo cancellation across all 128 channels in E1 mode or all 96 channels in T1/J1 modes. Combined with the VPMOCT128, the product is known as TE420B. T he TE420 has been designed by Digium to be fully compatible and integrated with Asterisk Business Edition, AsteriskNOW, and open source Asterisk. The TE420 supports industry standard telephony protocols including North American and European Primary Rate signalling as well as standard Robbed Bit, Channel Associated Signalling in addition to standard PPP, HDLC, and Frame Relay data modes. 549 Sangoma A500 BRI Card PCI Express 171.0000 225.00 Sangoma Sangoma BRI Cards http://www.voipon.co.uk/sangoma-a500-bri-card-pci-express-p-549.html http://www.voipon.co.uk/images/sangoma_a500_bri.jpg new Availability: In stock Sangoma built its business by designing hardware that simply works, the first time. We have taken the time to ensure our BRI solution delivers. The Sangoma A500 S/T BRI Interface Card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional Octasic© Telco-grade, hardware echo cancellation. A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable. Finally, a BRI card that upholds Sangoma&#8242;s high standards of quality in engineering and untiring product support. Architecture The A500 consists of a Remora© BRI daughterboard mounted on the AFT PCI card. The Remora© BRI card has three sockets, each of which can accept an S/T BRI module. One S/T BRI module has two S/T four wire interfaces, which support TE or NT modes of operation. Changing modes requires no jumpers - simply invert the module. Up to three additional Remora© daughterboards can be mounted in empty slot positions beside the A500 assembly. These are connected to the A500 by a special backplane bus connector. Technical Specifications From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers - simply invert the colour-coded module. Supports Asterisk®, Yate©, FreeSwitch©, CallWeaver©, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single synchronous PCI and PCI Express interface for all 24 BRI interfaces. Six ports per RemoraTM card. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5 V and 3.3 V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3 V or 5 V. Temperature range: 0 - 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested, and reliable Frame Relay, PPP, HDLC and X.25. 550 Sangoma A500 BRI Slave Card 95.0000 125.00 Sangoma Sangoma BRI Cards http://www.voipon.co.uk/sangoma-a500-bri-slave-card-p-550.html http://www.voipon.co.uk/images/sangoma_a500_slave_card.jpg new Availability: In stock A500 BRI ISDN PCI / PCI Express Card Sangoma built its business by designing hardware that simply works, the first time. We have taken the time to ensure our BRI solution delivers. The Sangoma A500 S/T BRI Interface Card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional Octasic© Telco-grade, hardware echo cancellation. A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable. Finally, a BRI card that upholds Sangoma&#8242;s high standards of quality in engineering and untiring product support. Architecture The A500 consists of a Remora© BRI daughterboard mounted on the AFT PCI card. The Remora© BRI card has three sockets, each of which can accept an S/T BRI module. One S/T BRI module has two S/T four wire interfaces, which support TE or NT modes of operation. Changing modes requires no jumpers - simply invert the module. Up to three additional Remora© daughterboards can be mounted in empty slot positions beside the A500 assembly. These are connected to the A500 by a special backplane bus connector. Technical Specifications From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers - simply invert the colour-coded module. Supports Asterisk®, Yate©, FreeSwitch©, CallWeaver©, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single synchronous PCI and PCI Express interface for all 24 BRI interfaces. Six ports per RemoraTM card. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5 V and 3.3 V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3 V or 5 V. Temperature range: 0 - 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested, and reliable Frame Relay, PPP, HDLC and X.25. 551 Sangoma A500 BRI Module 67.0000 90.00 Sangoma Sangoma BRI Cards http://www.voipon.co.uk/sangoma-a500-bri-module-p-551.html http://www.voipon.co.uk/images/sangoma_a500_bri_module.jpg new Availability: In stock A500 BRI ISDN PCI / PCI Express Card Sangoma built its business by designing hardware that simply works, the first time. We have taken the time to ensure our BRI solution delivers. The Sangoma A500 S/T BRI Interface Card delivers superior audio quality and scalability. Expand from two to twenty-four ports of BRI with optional Octasic© Telco-grade, hardware echo cancellation. A single PCI or PCI Express slot hosts the connection for up to 24 ports and ensures common synchronous clocking for all channels with no signaling issues. The card is 100% software configurable. Finally, a BRI card that upholds Sangoma&#8242;s high standards of quality in engineering and untiring product support. Architecture The A500 consists of a Remora© BRI daughterboard mounted on the AFT PCI card. The Remora© BRI card has three sockets, each of which can accept an S/T BRI module. One S/T BRI module has two S/T four wire interfaces, which support TE or NT modes of operation. Changing modes requires no jumpers - simply invert the module. Up to three additional Remora© daughterboards can be mounted in empty slot positions beside the A500 assembly. These are connected to the A500 by a special backplane bus connector. Technical Specifications From 2 to 24 ports are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers - simply invert the colour-coded module. Supports Asterisk®, Yate©, FreeSwitch©, CallWeaver©, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single synchronous PCI and PCI Express interface for all 24 BRI interfaces. Six ports per RemoraTM card. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Autosense compatibility with 5 V and 3.3 V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3 V or 5 V. Temperature range: 0 - 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested, and reliable Frame Relay, PPP, HDLC and X.25. 552 Polycom SoundPoint IP601 Expansion Module 140.0000 167.00 Polycom VoIP Accessories http://www.voipon.co.uk/polycom-soundpoint-ip601-expansion-module-p-552.html http://www.voipon.co.uk/images/polycom_ip601-exp_sml.jpg new Availability: In Stock Polycom Expansion Module - Turn the IP 601/650 into a fully featured Attendant Console Turn Your SoundPoint IP 601 or 650 Into a high-performance attendant console Augments the user interface of the SoundPoint IP 601 or 650 Simplifies management and monitoring of a high volume of simultaneous calls Together with a host phone, creates a high-performance, productivity-enhancing solution for telephone attendants Provides an easy transition from traditional Key Systems to the features and functionality of Voice over IP   Features & Benefits 160x320 pixel graphical grayscale LCD display 14 illuminated keys configurable as a line appearance or a speed dial with busy lamp field (BLF) Hot swappable - can be added to or removed from an idle host phone at any time Plug-and-play - requires no set-up as power and signaling are provided by the host phone No extra cables or power supplies are required User-friendly call visualization similar to that on the SoundPoint? IP 601 and 650       553 Camrivox Flexor 500 IP Telephone 57.0000 64.00 Camrivox Camrivox IP Telephones http://www.voipon.co.uk/camrivox-flexor-500-ip-telephone-p-553.html http://www.voipon.co.uk/images/camrivox_flexor_500.jpg new Availability: In stock   Flexor 500 brings new levels of functionality to IP Phones for the SoHo and SME markets. This next generation IP Phone connecting to IP-PBX or hosted services uniquely includes class leading Computer Telephony Integration (CTI). The Flexor 500 includes the features and capabilities found on current IP Phones, and adds unique innovations combined with a desirable and elegant product design. The Flexor 500 is part of a range of VoIP products designed, developed, and manufactured by Camrivox. The Flexor 500 is a multi-line Voice over IP telephone that seamlessly integrates with IP PBXs, Soft switches and Hosted VoIP services. Dedicated keys - provide direct access to most audio and call control functions and an intuitive user interface using a large backlit graphical LCD display make this an easy to use product. Superb sound quality - when using the handset, hands free speakerphone or headset. Class leading Computer Telephony Integration (CTI) - providing the maximum productivity and efficiency enhancements to business users. The CTI functions include on screen call "pop-ups", click to dial from any PC application and call control from the PC. Features Large size backlit graphical LCD display Hands free high quality speaker phone Headset Interface Dual 10/100M Ethernet interface Multi-line, multi-account support Address book and Outlook Synchronization Computer Telephony Integration (CTI) Operator Branding option Web browser configuration Camrivox zero touch provisioning compatible SIP compatible (RFC3261) Compatible with IP-PBX and Hosted VoIP services Quality of Service (QoS) Interfaces Power: 5V DC (use only supplied version) LAN Interface: RJ45, Ethernet (10/100) PC Interface: RJ45, Ethernet (10/100) 128 x 64 pixel LCD graphical display Handset: RJ14 Standard handset connector Speakerphone: full duplex Headset : 2.5mm Port Voice Algorithms and Protocols SIP, SIPS, RTP, SDP UDP, TCP and TLS supported VAD - Voice Activity Detect CNG - Comfort Noise Generation Jitter Buffer - Adaptive Packet Loss Compensator International Call Progress Tone generator DTMF: In-band and Out-of-band (RFC2833) Caller ID support (Name and Number) Call Waiting Indicator tone Message Waiting Indicator tone G.711, G.726, G.729, consult factory for others Networking DHCP client, DNS client ICMP, TCP/UDP/IP STUN SNTP HTTP/HTTPS AES RIP and RIPv2 Quality of Service (QoS) 802.1Q Type of Service (ToS) Pacing and prioritization of voice data VLAN (802.1pq) Management Configuration, diagnostic and firmware-upgrade locally or by operator Voice diagnostics Remote provisioning Product Physical Data Measurements: 170 x 180 x 90 mm Weight: 350 grams Environmental 5°C - 40°C operating temperature 5% -90% non-condensing humidity Power Supply (external) Input 100-240 VAC 50/60Hz; 0.5A Output 5.0 VDC; 1.5A 554 Hitachi Wireless IP5000 USB Cable 23.0000 24.00 Hitachi Hitachi IP Telephone http://www.voipon.co.uk/hitachi-wireless-ip5000-usb-cable-p-554.html http://www.voipon.co.uk/images/hitachi_ip5000_usb_cable.jpg new Availability: In Stock Connect to your PC's USB interface using an optional USB cable to recharge the IP5000's battery, upgrade the IP-5000 or configure and provision through Hitachi's new USB Software Manager feature. 555 Alona Female Spanish Asterisk Voice Prompt 68.35 68.35 Asterisk Voice Prompts http://www.voipon.co.uk/alona-female-spanish-asterisk-voice-prompt-p-555.html http://www.voipon.co.uk/images/alona.jpg new Availability: In Stock Female Spanish Asterisk voice prompts professionally sound recorded by Alona, a native Spanish speaking voice artist. "You too, can benefit from our professionally recorded voices in your Asterisk PBX. Give your customers a great impression from the very first call they make." Native speaking Spanish voice artist with a great voice. This product can directly replace the standard voice prompts of Asterisk. Completely compatible with Trixbox, Freepbx, Amp and Asterisk@Home Record your own prompts at any time and have them recorded by the same voice artist. Top quality recordings at 44.1khz in a professional sound proofed studio. Free updated for a year. First impressions count. When a customer calls what do you want the first impression to be? Do you want to sound professional?, do you want to sound confident?, do you want to have the right accents? It's time to let Alona handle your calls. Not only do you get a great voice but you get all the necessary voice prompts to handle each and every feature of asterisk. This included voicemail, call queues, call transfers, call parking, numbers, digits, alphabets and error messages. But you get more than just Asterisk support. With the ever expanding products on the market that utilise the features of Asterisk and extend them even further we have extended the range of recordings to include voice prompts specific to trixbox, FreePbx, Amp and Asterisk@home. Custom voice recordings can be ordered online The Spanish recording can be extended with your own custom recordings and delivered within a few days depending on how busy Alona is. And just to top it of Alona is a top voice artist. To make things even better Alona is a top of her professional voice artist so your guaranteed to get the best recordings with the right intonation, pacing and energy for all of your work. Top of the range Protools equipped recording studios Our three studios are all acoustically designed to give the highest quality recordings with a totally silent environment. We then edit the voice recordings down and remove any pops or clicks from the final work. Delivered in any Audio format you require. The final voice prompts are transcoded into whatever codec you need like Alaw, Ulaw, gsm or Wav. We transcode straight from the original 44.1khz recordings to ensure you get the best quality out of each codec. Free updates for a year The voice prompts are continuing to evolve as new features and functionality is added to Asterisk. When this occurs we record new prompts and release a new version. Since you get free updates for a year you don&trade;t have to worry about this continual change, simply install the new product and your off and running. Listen to some samples: (in .wav 706kbps 44khz) agent-user.wav call-fwd-cancelled.wav conf-getconfno.wav dir-instr.wav priv-instruct.wav priv-trying.wav vm-tocallback.wav 556 Brad Male American Asterisk Voice Prompt 39.82 39.82 Asterisk Voice Prompts http://www.voipon.co.uk/brad-male-american-asterisk-voice-prompt-p-556.html http://www.voipon.co.uk/images/brad.jpg new Availability: In Stock Brad - Male American English Asterisk voice prompts professionally sound recorded by native speaking voice artist. "You too, can benefit from our professionally recorded American English voices in your Asterisk PBX. Give your customers a great impression from the very first call they make." Native American English voice artist with a professional business voice. Direct replacement for the standard Asterisk voice prompts. Extra prompts included for compatibility with Trixbox, amp, FreePbx and others. Unique, specific, recordings for your ivr menu and call queues can be seamlessly recorded by the same artists. Recorded at 44.1khz in our professional recording studio. Free upgrades for one year First impressions count. When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Brad to handle your calls? All the standard voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Follow me, Call transfers, Call Parking, Voice Mail, Transfers, Error messages, Numbers (digits), letters and Phonetics. But it's More than just Asterisk. Now we have compatible voices even if your not rolling your own. We have extra recordings specifically designed for each of the features unique to FreePbx, Trixbox and Amp. Custom voice prompts can be ordered online Custom voice prompts can be ordered online and are available for download shortly after the order has been processed. Usually within a couple of days depending on the demands on the voice artists. Seemless recordings created by the same voice artist Brad records the customizations ensuring a seamless telephone experience for your yourselves and your customers, while we ensure that the voice artists delivery is identical to the full Asterisk pack. We only work with "top of their profession" voice artists that are native speakers in their chosen language. This ensures that the completed voice prompts are very high quality and the voice artist is professional, fast and highly skilled. Professionally recorded in our top of the range Protools studio. All the recordings are conducted in our professional top of the line Protools equipped studios which are acoustically designed by professional acoustic architects for accurate audio monitoring ensuring every voice prompt sounds the same on the phone as it does in the studio. The Voice prompts are recorded in 44.1 khz but get processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. If you need higher quality voice prompts we can provide them in any audio format you require. We ensure that the voice prompts are kept updated by releasing new versions every time an Asterisk release requires new or modified prompts. This ensures that you can use any new or improved Asterisk features as soon as they become available. Listen to some samples: (in .wav 706kbps 44khz) agent-user.wav call-fwd-cancelled.wav conf-getconfno.wav dir-instr.wav priv-instruct.wav priv-trying.wav queue-holdtime.wav 557 Marilda Female German Asterisk Voice Prompt 68.32 68.32 Asterisk Voice Prompts http://www.voipon.co.uk/marilda-female-german-asterisk-voice-prompt-p-557.html http://www.voipon.co.uk/images/marilda.jpg new Availability: In Stock Marilda - German Asterisk voice prompts professionally sound recorded by native speaking voice artist. "You too, can benefit from our professionally recorded German voices in your Asterisk PBX. Give your customers a great impression from the very first call they make. Native German voice artist with a professional business voice. Direct replacement for the standard Asterisk voice prompts. Extra prompts included for compatibility with Trixbox, amp, FreePbx and others. Unique, specific, recordings for your ivr menu and call queues can be seamlessly recorded by the same artists. Recorded at 44.1khz in our professional recording studio. Free upgrades for one year First impressions count. When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Marilda to handle your calls? All the standard voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Follow me, Call transfers, Call Parking, Voice Mail, Transfers, Error messages, Numbers (digits), letters and Phonetics. But it's More than just Asterisk. Now we have compatible voices even if your not rolling your own. We have extra recordings specifically designed for each of the features unique to FreePbx, Trixbox and Amp. Custom voice prompts can be ordered online Custom voice prompts can be ordered online and are available for download shortly after the order has been processed. Usually within a couple of days depending on the demands on the voice artists. Seemless recordings created by the same voice artist Marilda records the customizations ensuring a seamless telephone experience for your yourselves and your customers, while we ensure that the voice artists delivery is identical to the full Asterisk pack. We only work with "top of their profession" voice artists that are native speakers in their chosen language. This ensures that the completed voice prompts are very high quality and the voice artist is professional, fast and highly skilled. Professionally recorded in our top of the range Protools studio. All the recordings are conducted in our professional top of the line Protools equipped studios which are acoustically designed by professional acoustic architects for accurate audio monitoring ensuring every voice prompt sounds the same on the phone as it does in the studio. The Voice prompts are recorded in 44.1 khz but get processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. If you need higher quality voice prompts we can provide them in any audio format you require. We ensure that the voice prompts are kept updated by releasing new versions every time an Asterisk release requires new or modified prompts. This ensures that you can use any new or improved Asterisk features as soon as they become available. Listen to some samples: (in .wav 706kbps 44khz) agent-user.wav call-fwd-cancelled.wav conf-getconfno.wav dir-intro.wav priv-instruct.wav priv-trying.wav vm-starmain.wav 558 Silvia Female Portuguese Asterisk Voice Prompt 68.32 68.32 Asterisk Voice Prompts http://www.voipon.co.uk/silvia-female-portuguese-asterisk-voice-prompt-p-558.html http://www.voipon.co.uk/images/silvia.jpg new Availability: In Stock Silvia - Portuguese Asterisk voice prompts professionally sound recorded by native speaking voice artist. "You too, can benefit from our professionally recorded Portuguese voices in your Asterisk PBX. Give your customers a great impression from the very first call they make." Native Portuguese voice artist with a professional business voice. Direct replacement for the standard Asterisk voice prompts. Extra prompts included for compatibility with Trixbox, amp, FreePbx and others. Unique, specific, recordings for your ivr menu and call queues can be seamlessly recorded by the same artists. Recorded at 44.1khz in our professional recording studio. Free upgrades for one year First impressions count. When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Silvia to handle your calls? All the standard voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Follow me, Call transfers, Call Parking, Voice Mail, Transfers, Error messages, Numbers (digits), letters and Phonetics. But it's More than just Asterisk. Now we have compatible voices even if your not rolling your own. We have extra recordings specifically designed for each of the features unique to FreePbx, Trixbox and Amp. Custom voice prompts can be ordered online Custom voice prompts can be ordered online and are available for download shortly after the order has been processed. Usually within a couple of days depending on the demands on the voice artists. Seemless recordings created by the same voice artist Silvia records the customizations ensuring a seamless telephone experience for your yourselves and your customers, while we ensure that the voice artists delivery is identical to the full Asterisk pack. We only work with "top of their profession" voice artists that are native speakers in their chosen language. This ensures that the completed voice prompts are very high quality and the voice artist is professional, fast and highly skilled. Professionally recorded in our top of the range Protools studio. All the recordings are conducted in our professional top of the line Protools equipped studios which are acoustically designed by professional acoustic architects for accurate audio monitoring ensuring every voice prompt sounds the same on the phone as it does in the studio. Listen to some samples: (in .wav 706kbps 44khz) agent-user.wav call-fwd-cancelled.wav conf-getconfno.wav dir-instr.wav priv-instruct.wav priv-trying.wav vm-tempgreeting.wav 559 Tansel Female Turkish Asterisk Voice Prompt 68.32 68.32 Asterisk Voice Prompts http://www.voipon.co.uk/tansel-female-turkish-asterisk-voice-prompt-p-559.html http://www.voipon.co.uk/images/tansel.jpg new Availability: In Stock Tansel - Turkish Asterisk voice prompts professionally sound recorded by native speaking voice artist. "You too, can benefit from our professionally recorded Turkish voices in your Asterisk PBX. Give your customers a great impression from the very first call they make." Native Turkish voice artist with a professional business voice. Direct replacement for the standard Asterisk voice prompts. Extra prompts included for compatibility with Trixbox, amp, FreePbx and others. Unique, specific, recordings for your ivr menu and call queues can be seamlessly recorded by the same artists. Recorded at 44.1khz in our professional recording studio. Free upgrades for one year First impressions count. When a customer calls, what voice would you like them to hear? If your customers are mainly British, why give them the impression you are American? Why not employ Tansel to handle your calls? All the standard voice prompts are included to allow you to use the full functionality of the Asterisk PBX right out of the box. This includes Voice Menus, Call Queues, Follow me, Call transfers, Call Parking, Voice Mail, Transfers, Error messages, Numbers (digits), letters and Phonetics. But it's More than just Asterisk. Now we have compatible voices even if your not rolling your own. We have extra recordings specifically designed for each of the features unique to FreePbx, Trixbox and Amp. Custom voice prompts can be ordered online Custom voice prompts can be ordered online and are available for download shortly after the order has been processed. Usually within a couple of days depending on the demands on the voice artists. Seemless recordings created by the same voice artist Tansel records the customizations ensuring a seamless telephone experience for your yourselves and your customers, while we ensure that the voice artists delivery is identical to the full Asterisk pack. We only work with "top of their profession" voice artists that are native speakers in their chosen language. This ensures that the completed voice prompts are very high quality and the voice artist is professional, fast and highly skilled. Professionally recorded in our top of the range Protools studio. All the recordings are conducted in our professional top of the line Protools equipped studios which are acoustically designed by professional acoustic architects for accurate audio monitoring ensuring every voice prompt sounds the same on the phone as it does in the studio. The Voice prompts are recorded in 44.1 khz but get processed into the gsm audio format in 8.000 kHz, Mono 1 kb/sec. If you need higher quality voice prompts we can provide them in any audio format you require. We ensure that the voice prompts are kept updated by releasing new versions every time an Asterisk release requires new or modified prompts. This ensures that you can use any new or improved Asterisk features as soon as they become available. Listen to some samples: (in .wav 706kbps 44khz) agent-user.wav call-fwd-cancelled.wav conf-getconfno.wav dir-instr.wav priv-instruct.wav priv-trying.wav vm-tempgreetactive.wav 560 AVM Fritz!Box Fon WLAN 7170 115.0000 118.00 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-wlan-7170-p-560.html http://www.voipon.co.uk/images/avm_fritzbox_fon_wlan_7170.jpg new Availability: In stock PBX, router, and ADSL modem in a single device FRITZ!Box Fon WLAN 7170 and FRITZ!Box Fon 5124 combine a PBX for Internet and landline telephony, router, and ADSL modem in a single device. Telephone conversations over the Internet or regular landlines are possible even when the computer is off. Existing analogue telephones can be connected to the FRITZ!Box. In addition, the FRITZ!Box Fon WLAN 7170 includes a connector for ISDN telephones and ISDN telephone systems. With four LAN ports, the FRITZ!Box can connect computers as well as network-capable devices like game consoles to the Internet. The ADSL modem supports ADSL2+ with transfer rates of up to 16 Mbit/s. The integrated WLAN router in the FRITZ!Box Fon WLAN 7170 offers wireless Internet access as well. The factory configured, custom encryption guarantees a secure WLAN connection. Convenient, simple Internet telephony with FRITZ!Box Frau mit FRITZ!BoxYou can use the FRITZ!Box to place calls over the Internet or your landline. Your computer does not even have to be on to place calls over the Internet. Existing analogue telephones can be connected to the FRITZ!Box. The FRITZ!Box Fon 5124 includes two analogue connectors for telephones and fax machines. In addition, the FRITZ!Box Fon WLAN 7170 includes three analogue connectors as well as a connector for ISDN telephones and ISDN telephone systems. The integrated bandwidth management guarantees optimal data and voice transmission for excellent voice quality. The FRITZ!Box telephone system provides a wide range of features like toggling, call waiting, call forwarding, and three-way conference calls. Comprehensive, useful functions and intuitive operation The new FRITZ!Box models also feature our award-winning, user-friendly, simple operation. Numerous functions, like child lock, which allows you determine how long and at what times your children can use the Internet, or eco mode, which helps reduce the power consumption of the FRITZ!Box, offer real value-added features. AVM regularly expands the FRITZ!Boxes with practical new functions through firmware updates which are available to users free of charge. Share network printers and hard drives The USB host port of the FRITZ!Box enables a wide range of network applications. Memory sticks, printers, or hard drives can be connected to the FRITZ!Box and shared by all computers on the network. The computers connected to the FRITZ!Box are also networked together. The user does not need to configure any network settings because the IP addresses are automatically assigned, if desired. Secure, fast wireless Internet access The FRITZ!Box Fon WLAN 7170 offers wireless Internet access. Encryption is factory preconfigured to guarantee a secure WLAN connection as soon as you start using the FRITZ!Box. Each unit has its own unique, custom password that appears on the bottom of the device. In addition, the latest WLAN security standards (WPA and WPA2) offer maximum security. Apart from the common 802.11g and b wireless standards, the FRITZ!Box, in conjunction with the FRITZ!WLAN USB Stick, supports the 802.11g++ standard as well. This increases the net data transfer rate by approximately one third. The WLAN can be switched off automatically using sleep mode or manually with a switch on the back of the device. With the FRITZ!WLAN USB Stick and the Stick & Surf technology developed by AVM, establishing a secure WLAN connection is even simpler; the Stick installs automatically under Windows XP Service Pack 2 or higher, and computers can be connected to the FRITZ!Box without any additional configuration. 561 AVM Fritz!Box Fon 5124 93 93.00 AVM AVM FRITZ!Box http://www.voipon.co.uk/avm-fritzbox-fon-5124-p-561.html http://www.voipon.co.uk/images/avm_fritzbox_fon.jpg new Availability: In stock PBX, router, and ADSL modem in a single device FRITZ!Box Fon WLAN 7170 and FRITZ!Box Fon 5124 combine a PBX for Internet and landline telephony, router, and ADSL modem in a single device. Telephone conversations over the Internet or regular landlines are possible even when the computer is off. Existing analogue telephones can be connected to the FRITZ!Box. In addition, the FRITZ!Box Fon WLAN 7170 includes a connector for ISDN telephones and ISDN telephone systems. With four LAN ports, the FRITZ!Box can connect computers as well as network-capable devices like game consoles to the Internet. The ADSL modem supports ADSL2+ with transfer rates of up to 16 Mbit/s. The integrated WLAN router in the FRITZ!Box Fon WLAN 7170 offers wireless Internet access as well. The factory configured, custom encryption guarantees a secure WLAN connection. Convenient, simple Internet telephony with FRITZ!Box Frau mit FRITZ!BoxYou can use the FRITZ!Box to place calls over the Internet or your landline. Your computer does not even have to be on to place calls over the Internet. Existing analogue telephones can be connected to the FRITZ!Box. The FRITZ!Box Fon 5124 includes two analogue connectors for telephones and fax machines. In addition, the FRITZ!Box Fon WLAN 7170 includes three analogue connectors as well as a connector for ISDN telephones and ISDN telephone systems. The integrated bandwidth management guarantees optimal data and voice transmission for excellent voice quality. The FRITZ!Box telephone system provides a wide range of features like toggling, call waiting, call forwarding, and three-way conference calls. Comprehensive, useful functions and intuitive operation The new FRITZ!Box models also feature our award-winning, user-friendly, simple operation. Numerous functions, like child lock, which allows you determine how long and at what times your children can use the Internet, or eco mode, which helps reduce the power consumption of the FRITZ!Box, offer real value-added features. AVM regularly expands the FRITZ!Boxes with practical new functions through firmware updates which are available to users free of charge. Share network printers and hard drives The USB host port of the FRITZ!Box enables a wide range of network applications. Memory sticks, printers, or hard drives can be connected to the FRITZ!Box and shared by all computers on the network. The computers connected to the FRITZ!Box are also networked together. The user does not need to configure any network settings because the IP addresses are automatically assigned, if desired. Secure, fast wireless Internet access The FRITZ!Box Fon WLAN 7170 offers wireless Internet access. Encryption is factory preconfigured to guarantee a secure WLAN connection as soon as you start using the FRITZ!Box. Each unit has its own unique, custom password that appears on the bottom of the device. In addition, the latest WLAN security standards (WPA and WPA2) offer maximum security. Apart from the common 802.11g and b wireless standards, the FRITZ!Box, in conjunction with the FRITZ!WLAN USB Stick, supports the 802.11g++ standard as well. This increases the net data transfer rate by approximately one third. The WLAN can be switched off automatically using sleep mode or manually with a switch on the back of the device. With the FRITZ!WLAN USB Stick and the Stick & Surf technology developed by AVM, establishing a secure WLAN connection is even simpler; the Stick installs automatically under Windows XP Service Pack 2 or higher, and computers can be connected to the FRITZ!Box without any additional configuration. 562 Digium TDM11B - Asterisk Retail Package 124.95 124.95 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm11b-asterisk-retail-package-p-562.html http://www.voipon.co.uk/images/asterisk_retail_package.jpg new Availability: In Stock The Wildcard TDM400P is a half-length PCI 2.2 card for connecting up to 4 analog telephones in a PC. The Card supports a combination of up to 4 FXS and/or FXO modules for a total of 4 lines. The TDM11B comes installed with 1 FXO module for connecting to 1 analog telephone (POTS) line and 1 FXS module for connecting to 1 analog telephone. Using Digium\'s Asterisk® PBX software and standard PC hardware a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX/Voicemail platform. The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. By using S110M and X100M modules with the TDM400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules. Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules. The Retail Package Includes: One (1) TDM400P One (1) FXS Module (green) One (1) FXO Module (red) Hardware Installation manual RJ11 cables (one for each active port) Digium® | Asterisk reversible screwdriver Digium | Asterisk mouse pad Suggested Hardware Requirements: 300MHz Pentium or PowerPC based system (or better) One Available PCI slot - Must have PCI 2.2 Linux 2.4 Series Kernel 563 Digium TDM22B Asterisk Retail Package 223.95 223.95 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm22b-asterisk-retail-package-p-563.html http://www.voipon.co.uk/images/asterisk_retail_package.jpg new Availability: In Stock The Wildcard TDM400P is a half-length PCI 2.2 card for connecting up to 4 analog telephones in a PC. The Card supports a combination of up to 4 FXS and/or FXO modules for a total of 4 lines. The TDM22B comes installed with 2 FXO modules for connecting to 2 analog telephone (POTS) lines and 2 FXS modules for connecting to 2 analog telephones. Using Digium's Asterisk® PBX software and standard PC hardware a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX/Voicemail platform. The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. By using S110M and X100M modules with the TDM400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules. Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules. The Retail Package Includes: One (1) TDM400P Two (2) FXS Modules (green) Two (2) FXO Modules (red) Hardware Installation manual RJ11 cables (one for each active port) Digium | Asterisk reversible screwdriver Digium | Asterisk mouse pad Suggested Hardware Requirements: 300MHz Pentium or PowerPC based system (or better) One Available PCI slot - Must have PCI 2.2 Linux 2.4 Series Kernel   564 Digium TDM04B - Asterisk Retail Package 247.95 247.95 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm04b-asterisk-retail-package-p-564.html http://www.voipon.co.uk/images/asterisk_retail_package.jpg new Availability: In Stock The Wildcard TDM400P is a half-length PCI 2.2 card for connecting up to 4 analog telephones in a PC. The Card supports a combination of up to 4 FXS and/or FXO modules for a total of 4 lines. The TDM04B comes installed with 4 FXO modules for connecting to 4 analog telephone (POTS) lines. Using Digium's Asterisk® PBX software and standard PC hardware a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX/Voicemail platform. The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. By using S110M and X100M modules with the TDM400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules. Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules. The Retail Package Includes: One (1) TDM400P Four (4) FXO Modules (red) Hardware Installation manual RJ11 cables (one for each active port) Digium | Asterisk reversible screwdriver Digium | Asterisk mouse pad Suggested Hardware Requirements: 300MHz Pentium or PowerPC based system (or better) One Available PCI slot - Must have PCI 2.2 Linux 2.4 Series Kernel 565 Digium TDM40B - Asterisk Retail Package 219 219.00 Digium Digium TDM400P http://www.voipon.co.uk/digium-tdm40b-asterisk-retail-package-p-565.html http://www.voipon.co.uk/images/asterisk_retail_package.jpg new Availability: In Stock The Wildcard TDM400P is a half-length PCI 2.2 card for connecting up to 4 analog telephones in a PC. The Card supports a combination of up to 4 FXS and/or FXO modules for a total of 4 lines. The TDM22B comes installed with 2 FXO modules for connecting to 2 analog telephone (POTS) lines and 2 FXS modules for connecting to 2 analog telephones. Using Digium's Asterisk? PBX software and standard PC hardware a user can create a SOHO (Small Office Home Office) telephony environment which includes all the sophisticated features of a high end PBX/Voicemail platform. The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. By using S110M and X100M modules with the TDM400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules. Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules. The Retail Package Includes: One (1) TDM400P Four (4) FXS Modules (green) Zero (0) FXO Modules (red) Hardware Installation manual RJ11 cables (one for each active port) Digium | Asterisk reversible screwdriver Digium | Asterisk mouse pad Suggested Hardware Requirements: 300MHz Pentium or PowerPC based system (or better) One Available PCI slot - Must have PCI 2.2 Linux 2.4 Series Kernel 566 Xorcom Astribank-8 - 6 FXS 2 FXO 291.0000 328.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribank8-6-fxs-2-fxo-p-566.html http://www.voipon.co.uk/images/xorcom_astribank_6fxs_2fxo.jpg new Availability: In Stock Astribank 6FXS, 2FXO The latest member of the Astribank family is designed to provide a professional, simple and easy-to-implement solution for Asterisk installations that require a minimal number of analog trunks and extensions. With 6 FXS ports to drive up to 6 analog telephone sets or fax machines and 2 FXO ports to accommodate 2 PSTN analog lines, the unit is an ideal match for small installations, demonstrations and training purposes. 4 input ports and 2 output ports are available for auxiliary appliances (such as door locks and alarm systems). This unit is offered to Asterisk boot-camps under preferred "not for resale" terms. With Xorcom's Live CD you can set up a pre-configured, fully operational Asterisk IP-PBX with working telephony interfaces (for analog telephone sets and trunks) in just 3 minutes! 568 Rhino R24FXX-EC 24 Port Analog PCI Card - base board w/EC 176.0000 199.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-r24fxxec-24-port-analog-pci-card-base-board-wec-p-568.html http://www.voipon.co.uk/images/r24fxx.jpg new Availability: In Stock 24 Port FXS/FXO with Echo Cancellation PCI Plug-In Card THIS CARD HAS THE ABILITY TO ADD DUAL FXO AND DUAL FXS CARDS WHICH ARE SOLD SEPERATELY! Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R24FXS-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R4FXO requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. 24 Port FXX PCI Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Infineon PEB3268 DualSLIC chip Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R24FXS loads the PCI bus to no more than the load of a T1 card. One female RJ11 connector at card bracket, with Velco strap Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring 5-year limited warranty DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection SLIC Features On-chip uLaw or aLaw CODEC Integrated ringing generator, 65 Vrms capable +12V power derived from ATX power connector -48V DC on hook voltage, 25mA maximum loop current, loop start feed USA AC and DC impedance characteristics 500msec end-of-call battery interruption, programmable to 3 seconds MWI neon bulb capable On-hook data transmission Mechanical Data Size: 4.0? tall, 11.00? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 569 Rhino R24FXS-EC 24 channel, fixed FXS card w/EC 730 730.00 Rhino Rhino Non-Modular Analog Cards http://www.voipon.co.uk/rhino-r24fxsec-24-channel-fixed-fxs-card-wec-p-569.html http://www.voipon.co.uk/images/r24fxs.jpg new Availability: This product is now obsolete 24 Port FXS with Echo Cancellation PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R24FXS-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R24FXO-EC requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. R24FXS with EC 24 Port FXS PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Infineon PEB3268 DualSLIC chip Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R24FXS loads the PCI bus to no more than the load of a T1 card. One female RJ11 connector at card bracket, with Velco strap Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring Benefits Lowest cost 24 port FXS card Fully Zaptel and Asterisk compliant Echo Cancellation standard Plug multiple cards into one PC without PCI worries No plug-in modules SLIC Features On-chip uLaw or aLaw CODEC Integrated ringing generator, 65 Vrms capable +12V power derived from ATX power connector -48V DC on hook voltage, 25mA maximum loop current, loop start feed USA AC and DC impedance characteristics 500msec end-of-call battery interruption, programmable to 3 seconds MWI neon bulb capable On-hook data transmission Unlimited support Echo Cancellation Specs On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Mechanical Data Size: 4.00? tall, 11.20? wide Form Factor: Single PCI slot Shipping Weight: Less than one pound with all included components 570 Rhino R24FXO-EC 24 channel, fixed FXO card w/EC 681.0000 805.00 Rhino Rhino Non-Modular Analog Cards http://www.voipon.co.uk/rhino-r24fxoec-24-channel-fixed-fxo-card-wec-p-570.html http://www.voipon.co.uk/images/r24fxo.jpg new Availability: In Stock 24 Port FXO with Echo Cancellation PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R24FXO-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R24FXO-EC requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. R24FXO with EC 24 Port FXO PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R24FXO loads the PCI bus to no more than the load of a T1 card. One female RJ11 connector at card bracket, with Velco strap Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring Benefits Lowest cost 24 port FXO card Fully Zaptel and Asterisk compliant Echo Cancellation standard Plug multiple cards into one PC without PCI worries No plug-in modules Unlimited support DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection Echo Cancellation Specs On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Mechanical Data Size: 4.00? tall, 11.20? wide Form Factor: Single PCI slot Shipping Weight: Less than one pound with all included components 571 Rhino R8FXX-EC Octal Analog PCI Card - base board w/EC 150.0000 170.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-r8fxxec-octal-analog-pci-card-base-board-wec-p-571.html http://www.voipon.co.uk/images/r8fxx.jpg new Availability: In Stock Rhino R8FXX-EC Analog PCI Base Card with Built-In Echo Cancellation THIS CARD HAS THE ABILITY TO ADD DUAL FXO AND DUAL FXS CARDS WHICH ARE SOLD SEPERATELY! The Rhino R8FXX-EC is the first analog card for Asterisk that eliminates "bit-banging" the PCI bus. Other manufacturers have been known to have problems with bit-banging. The R8FXX-EC has built in echo cancellation so there is no need to purchase a separate module. The Rhino R8FXX PCI analog card features a control element to remove all "bit banging" on the PCI bus, something that other manufacturer's have yet to solve! This Rhino feature significantly reduces PCI and CPU overhead. Rhino R8FXX-EC Analog PCI Base Card with Built-In Echo Cancellation Rhino's onboard echo-cancellation provides excellent voice quality and fast convergence rate. There is no divergence due to doubletalk and cancels multiple independent tails. The Tone Disabler disables echo canceller during voiceband modem and FAX connections. The echo cancellation module is a G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino R8FXX-EC Features & Functionality Hosts up to four, dual channel FXS or FXO modular cards, using four RJ11 jacks with two lines per jack (inside and outside pairs) FXS channels utilize the Infineon PEB3268 ringing SLIC single chip FXO channels utilize the Silicon Labs Si3050 and Si3019 DAA chipset Channel status easily seen via eight on-board LEDs User upgradeable, modules snap on and off easily, yet are rigorously held in place both electrically and mechanically Rhino R8FXX-EC Echo-Cancellation Specifications On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. 572 Rhino R4FXO-EC Quad FXO PCI Card w/EC 204.0000 239.00 Rhino Rhino Non-Modular Analog Cards http://www.voipon.co.uk/rhino-r4fxoec-quad-fxo-pci-card-wec-p-572.html http://www.voipon.co.uk/images/r4fxo.jpg new Availability: In Stock Quad FXO (R4FXO) PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R4FXO-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R4FXO requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. R4FXO with EC Quad FXO PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R4FXO loads the PCI bus to no more than the load of a T1 card. Four RJ11 jacks at card bracket Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring 5-year limited warranty Benefits Lowest cost four port FXO card Fully Zaptel and Asterisk compliant Echo Cancellation standard Plug multiple cards into one PC without PCI worries No plug-in modules Unlimited support DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection Echo Cancellation Specs On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Mechanical Data Size: 3.90? tall, 5.50? wide Form Factor: Single PCI slot Shipping Weight: Less than one pound with all included components 573 Rhino Dual FXO Analog Module 55.0000 65.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-dual-fxo-analog-module-p-573.html http://www.voipon.co.uk/images/rhino_dual_fxo_module.jpg new Availability: In stock The Rhino Dual FXO module plugs into the R8FXX and R24FXX base cards and provides a universal telephone line interface that is compatible with 70 country line conditions. It supports two analog POTs incoming lines and is the ideal solution for expanding the number of lines you are using to connect to the PSTN. This module, as well as all the other Rhino products, is rock solid. Our warranty proves it: we offer the best one on the market, the Rhino 5-year limited warranty. And don?t forget that by purchasing this product you are entitled to the only lifetime support on the market. This means that whatever problem you might be experiencing with your Rhino product our technical team is here to help you in no time. In the rare case of a damaged module, you can replace it with a good one without having to throw away the entire board. This will save you money and valuable time, because the card can continue to host your voice applications on the other modules, temporarily losing just two channels instead of all channels. 574 Rhino Dual FXS Analog Module 48.0000 56.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-dual-fxs-analog-module-p-574.html http://www.voipon.co.uk/images/rhino_dual_fxs_module.jpg new Availability: In stock The Rhino Dual FXS module plugs into the R8FXX and R24FXX base cards and provides two telephone line extensions with integrated on-board battery and ringing. It offers the Infineon 3268 single chip solution for fully functional extension generation. This module, as well as all the other Rhino products, is rock solid. Our warranty proves it: we offer the best one on the market, the Rhino 5-year limited warranty. And don?t forget that by purchasing this product you are entitled to the only lifetime support on the market. This means that whatever problem you might be experiencing with your Rhino product our technical team is here to help you in no time. In the rare case of a damaged module, you can replace it with a good one without having to throw away the entire board. This will save you money and valuable time, because the card can continue to host your voice applications on the other modules, temporarily losing just two channels instead of all channels. 575 Irish VoIP Numbers 32.0000 34.00 Incoming Numbers http://www.voipon.co.uk/irish-voip-numbers-p-575.html http://www.voipon.co.uk/images/irish_voip_numbers.jpg new Availability: 24 Hours   VoIPon now provides Irish VoIP numbers for use with your IP phone or PBX. You can choose a number from various Irish regions below. You pay only a yearly rental fee. There are no additional charges. Charges to the caller are based on their network provider and do not differ from standard charges for calling Irish numbers. Irish Numbers   VoIPon can now provide you with an Irish VoIP Number, which can be configured to point to your VoIP phone or diverted to any destination worldwide, enabling callers to contact you from normal landlines, mobiles and payphones. 578 Hitachi IP5000 Protective Casing 22.0000 25.00 Hitachi Hitachi IP Telephone http://www.voipon.co.uk/hitachi-ip5000-protective-casing-p-578.html http://www.voipon.co.uk/images/hitachi_ip500_rubber_case.jpg new Availability: In Stock The IP5000 Protective Casing and belt clip is ruggedized for Warehouse and Factory use. 579 Trixbox Pro Standard Edition (SE) 0 0.00 Trixbox Trixbox Pro http://www.voipon.co.uk/trixbox-pro-standard-edition-se-p-579.html http://www.voipon.co.uk/images/trixbox_standard_edition_se.jpg new Availability: In Stock Recommended Deployment: < 10 employees Trixbox Standard Edition Download: Click Here trixbox Pro SE is our free hybrid-hosted telephony solution based on Asterisk® and improved for reliability and scalability (up to 500 users per site2). trixbox Pro SE provides enterprise-class phone system features for FREE! Besides the usual fare of Asterisk features, trixbox Pro SE inclues: an easy user interface, web-based voicemail, powerful exportable reporting, click-to-call, integrated HUDlite, mouse-driven operator panel, Outlook integration, real-time resource graphs, system alerts, auto-card configuring, seamless VoIP trunking, simplified co-branding, and more. Hybrid-hosted The Best of Both Worlds! trixbox® Pro is built on a unique patent-pending architecture that has the reliability of premise, but the flexibility of hosted, therefore "hybrid-hosted". Like a hosted solution, you get free VoIP calling, easy telecommuters, anywhere management, and monitoring. But, like a premise solution, you also get rock-solid PSTN connectivity, advanced premise-style features (hosted usually lags years behind), and complete call privacy. Secure Local Data Because trixbox Pro starts at the premise, it continues to receive and make calls if your Internet connection goes down. Your data is also 100% private because your voice traffic, voicemails, and recordings are only stored at your premise. Our hybrid-hosted architecture communicates with your trixbox Pro server via a VPN connection which means that you don't need to open any firewall ports! Hybrid-Hosted Benefits Multi-office Management via Admin Portal Web-based User Portal System Monitoring Moves, Adds, Changes, and Deletes (MACDs) Management Automatic Software Updates Zero-Configuration Teleworker Functionality Report Generation Web-based Management Portal Our trixbox Data Center maintains a highly available cluster of Web servers that allow all administrative and user interfaces to be accessible from anywhere at anytime through any Web browser. In contrast, a locally hosted management portal would require that you allow external Web access to your system, which may compromise the security of your internal network. [ control panel screenshots ] System Monitoring You can sleep knowing that we continually monitor the health of your local trixbox Pro server for: RAID drive failure notification PBX software errors Critical services failures Disk space overcapacity When a critical error is detected, our monitoring system tries first to repair and restart the service. When this is not possible our system will automatically generate a support ticket to trixbox's 24-hour support center for remedial action. [ support plan info ] Moves, Adds, Changes, and Deletes Management Every move, add, change, or deletion (MACD) made through the Web-based Control Panel first updates the trixbox Pro Data Center and is then pushed to the local server. Maintaining a real-time image of each local system configuration allows your system to be rebuilt and recovered rapidly if necessary. Automatic Software Updates Our hybrid-hosted architecture allows for the consistent, real-time display of the status of software versions installed at each site. Software updates are transparently sent to the local server during low-load times. Core upgrades are initiated by you and non-critical items show up automatically. This means we can add features and improve scale while you sleep! Zero-Configuration Teleworker - patent-pending! Because of the hybrid-hosted architecture, telecommuters and remote workers can be supported with zero-configuration required. The trixbox Data Center gathers the public IP address of the local trixbox Pro Server so that it can be used by remote IP phones to enable a simple registration process. Combined with DMZ/Host Server firewall settings, this virtually eliminates problems that usually occur in supporting IP phones for remote telecommuters, while protecting the security of your organization's internal systems. Report Generation trixbox Pro has an extensive call data reports (CDR) system where call reports are generated at our Data Center (via the hosted interface). This allows your local premise server to focus on call processing and avoid the CPU spikes often associacated with running large local database queries. Of course, all of your original call details are kept intact at your premise (in case you want to use a third-party reporting engine). TrixBox Pro Comparison Table Features SE w/HUDlite EE w/HUD Pro CCE w/HUD Pro   FREE!             Auto-Attendant (IVR) Outlook Integration Unlimited Extensions** CRM Integration Voicemail Voicemail-to-Email Hot Desk Music-on-Hold Scheduler Custom CTI (AGI) Analog & IP Phones Ring-All (Blast Group) Call Forwarding Name Directory DIDs VoIP-Ready 1 acct 2 accts unlim. PSTN Fallback (patent-pending) Telecommuters Branch Office Support Web-based Control Panel Powerful Reporting Re-Brandable Interface co full full Conference Bridge - Extension Groups - Routing by DIDs - Paging / Zone Paging - Intercom / Zone Intercom - Voicemail Groups - Advanced Call Forwarding - Call Return - Call Out - Report Exporting (.csv) - Custom Caller IDs - IVR Authentication - SMS/Pager Voicemail Notify - Upload Voice Prompts - Alerts & Notifications - Trunks Status Pages - Real-Time System Graphs - Historical System Graphs - Unlimited Call Queues** - - Full Featured A.C.D. - - Skills-based Routing - - Graphical Queue Reports - - Agent Call Recording - - Agent Variable Log-off - - Agents on Cell Phones - - Agents Shared across Sites - - Real-Time Queue Stats - -         HUD       Operator Panel (w/ BLF) *** Call Parking Area Drag & Drop Call Call Control Color-Coded Call Status Drag & Drop to Voicemail Extension Sorting Enterprise Instant Messaging Outlook Integration Presence Management Click-to-Call Mobile Phones Click-to-Email Desktop Alerts Interactive Desktop Alerts - On-the-Fly Recording - Group & User Permissions - Extension Grouping - Extension Search - Extension Search - QuickMenu - Shortcuts (Hotkeys) - Queue Status - - Agent Login/Logout - - Call Barging (active) - - Call Monitoring (passive) - - Web Access to Recordings - - Advanced CRM Integration - - ** Unlimited configuration, hardware platform may impose operational constraints *** The Operator Panel feature requires i) a phone with BLF capability and ii> a phone that is natively supported by Asterisk. 580 Trixbox Pro Enterprise Edition (EE) Lifetime License 125 125.00 Trixbox Trixbox Pro http://www.voipon.co.uk/trixbox-pro-enterprise-edition-ee-lifetime-license-p-580.html http://www.voipon.co.uk/images/trixbox_enterprise_edition_ee.jpg new Availability: In Stock Recommended Deployment: 10-500 employees A step up from trixbox Pro SE, trixbox Pro EE contains all of the features of Standard Edition, plus conference bridges, multiple auto-attendants, paging, intercom (group + individual), group permissions, full reseller re-branding capabilities, and more. trixbox Pro EE also comes standard with HUD Pro, which adds presence management, drag and drop call control, private enterprise chat, interactive desktop alerts, and more. Hybrid-hosted The Best of Both Worlds! trixbox® Pro is built on a unique patent-pending architecture that has the reliability of premise, but the flexibility of hosted, therefore "hybrid-hosted". Like a hosted solution, you get free VoIP calling, easy telecommuters, anywhere management, and monitoring. But, like a premise solution, you also get rock-solid PSTN connectivity, advanced premise-style features (hosted usually lags years behind), and complete call privacy. Secure Local Data Because trixbox Pro starts at the premise, it continues to receive and make calls if your Internet connection goes down. Your data is also 100% private because your voice traffic, voicemails, and recordings are only stored at your premise. Our hybrid-hosted architecture communicates with your trixbox Pro server via a VPN connection which means that you don't need to open any firewall ports! Hybrid-Hosted Benefits Multi-office Management via Admin Portal Web-based User Portal System Monitoring Moves, Adds, Changes, and Deletes (MACDs) Management Automatic Software Updates Zero-Configuration Teleworker Functionality Report Generation Web-based Management Portal Our trixbox Data Center maintains a highly available cluster of Web servers that allow all administrative and user interfaces to be accessible from anywhere at anytime through any Web browser. In contrast, a locally hosted management portal would require that you allow external Web access to your system, which may compromise the security of your internal network. [ control panel screenshots ] System Monitoring You can sleep knowing that we continually monitor the health of your local trixbox Pro server for: RAID drive failure notification PBX software errors Critical services failures Disk space overcapacity When a critical error is detected, our monitoring system tries first to repair and restart the service. When this is not possible our system will automatically generate a support ticket to trixbox's 24-hour support center for remedial action. [ support plan info ] Moves, Adds, Changes, and Deletes Management Every move, add, change, or deletion (MACD) made through the Web-based Control Panel first updates the trixbox Pro Data Center and is then pushed to the local server. Maintaining a real-time image of each local system configuration allows your system to be rebuilt and recovered rapidly if necessary. Automatic Software Updates Our hybrid-hosted architecture allows for the consistent, real-time display of the status of software versions installed at each site. Software updates are transparently sent to the local server during low-load times. Core upgrades are initiated by you and non-critical items show up automatically. This means we can add features and improve scale while you sleep! Zero-Configuration Teleworker - patent-pending! Because of the hybrid-hosted architecture, telecommuters and remote workers can be supported with zero-configuration required. The trixbox Data Center gathers the public IP address of the local trixbox Pro Server so that it can be used by remote IP phones to enable a simple registration process. Combined with DMZ/Host Server firewall settings, this virtually eliminates problems that usually occur in supporting IP phones for remote telecommuters, while protecting the security of your organization's internal systems. Report Generation trixbox Pro has an extensive call data reports (CDR) system where call reports are generated at our Data Center (via the hosted interface). This allows your local premise server to focus on call processing and avoid the CPU spikes often associacated with running large local database queries. Of course, all of your original call details are kept intact at your premise (in case you want to use a third-party reporting engine). TrixBox Pro Comparison Table Features SE w/HUDlite EE w/HUD Pro CCE w/HUD Pro   FREE!             Auto-Attendant (IVR) Outlook Integration Unlimited Extensions** CRM Integration Voicemail Voicemail-to-Email Hot Desk Music-on-Hold Scheduler Custom CTI (AGI) Analog & IP Phones Ring-All (Blast Group) Call Forwarding Name Directory DIDs VoIP-Ready 1 acct 2 accts unlim. PSTN Fallback (patent-pending) Telecommuters Branch Office Support Web-based Control Panel Powerful Reporting Re-Brandable Interface co full full Conference Bridge - Extension Groups - Routing by DIDs - Paging / Zone Paging - Intercom / Zone Intercom - Voicemail Groups - Advanced Call Forwarding - Call Return - Call Out - Report Exporting (.csv) - Custom Caller IDs - IVR Authentication - SMS/Pager Voicemail Notify - Upload Voice Prompts - Alerts & Notifications - Trunks Status Pages - Real-Time System Graphs - Historical System Graphs - Unlimited Call Queues** - - Full Featured A.C.D. - - Skills-based Routing - - Graphical Queue Reports - - Agent Call Recording - - Agent Variable Log-off - - Agents on Cell Phones - - Agents Shared across Sites - - Real-Time Queue Stats - -         HUD       Operator Panel (w/ BLF) *** Call Parking Area Drag & Drop Call Call Control Color-Coded Call Status Drag & Drop to Voicemail Extension Sorting Enterprise Instant Messaging Outlook Integration Presence Management Click-to-Call Mobile Phones Click-to-Email Desktop Alerts Interactive Desktop Alerts - On-the-Fly Recording - Group & User Permissions - Extension Grouping - Extension Search - Extension Search - QuickMenu - Shortcuts (Hotkeys) - Queue Status - - Agent Login/Logout - - Call Barging (active) - - Call Monitoring (passive) - - Web Access to Recordings - - Advanced CRM Integration - - ** Unlimited configuration, hardware platform may impose operational constraints *** The Operator Panel feature requires i) a phone with BLF capability and ii> a phone that is natively supported by Asterisk. 581 Trixbox Pro Call Center Edition (CCE) Lifetime License 250 250.00 Trixbox Trixbox Pro http://www.voipon.co.uk/trixbox-pro-call-center-edition-cce-lifetime-license-p-581.html http://www.voipon.co.uk/images/trixbox_call_center_edition_cce.jpg new Availability: In Stock Recommended Deployment: Call center w/ 2 - 200 agents trixbox Pro CCE builds on the powerful features of both Standard and Enterprise Edition by adding advanced call center capabilities at an unbelievably low price. Designed for companies with 2 - 200 agents, Call Center Edition adds robust ACD and IVR capabilities with unlimited queues3, skills-based routing, real-time queue statistics, graphical reports, web-based recording access, and more. trixbox Pro CCE comes with HUD Pro, which brings advanced agent capabilities such as on-the-fly recording, call barge, call monitor, CRM integration, and one-touch agent login. Hybrid-hosted The Best of Both Worlds! trixbox® Pro is built on a unique patent-pending architecture that has the reliability of premise, but the flexibility of hosted, therefore "hybrid-hosted". Like a hosted solution, you get free VoIP calling, easy telecommuters, anywhere management, and monitoring. But, like a premise solution, you also get rock-solid PSTN connectivity, advanced premise-style features (hosted usually lags years behind), and complete call privacy. Secure Local Data Because trixbox Pro starts at the premise, it continues to receive and make calls if your Internet connection goes down. Your data is also 100% private because your voice traffic, voicemails, and recordings are only stored at your premise. Our hybrid-hosted architecture communicates with your trixbox Pro server via a VPN connection which means that you don't need to open any firewall ports! Hybrid-Hosted Benefits Multi-office Management via Admin Portal Web-based User Portal System Monitoring Moves, Adds, Changes, and Deletes (MACDs) Management Automatic Software Updates Zero-Configuration Teleworker Functionality Report Generation Web-based Management Portal Our trixbox Data Center maintains a highly available cluster of Web servers that allow all administrative and user interfaces to be accessible from anywhere at anytime through any Web browser. In contrast, a locally hosted management portal would require that you allow external Web access to your system, which may compromise the security of your internal network. [ control panel screenshots ] System Monitoring You can sleep knowing that we continually monitor the health of your local trixbox Pro server for: RAID drive failure notification PBX software errors Critical services failures Disk space overcapacity When a critical error is detected, our monitoring system tries first to repair and restart the service. When this is not possible our system will automatically generate a support ticket to trixbox's 24-hour support center for remedial action. [ support plan info ] Moves, Adds, Changes, and Deletes Management Every move, add, change, or deletion (MACD) made through the Web-based Control Panel first updates the trixbox Pro Data Center and is then pushed to the local server. Maintaining a real-time image of each local system configuration allows your system to be rebuilt and recovered rapidly if necessary. Automatic Software Updates Our hybrid-hosted architecture allows for the consistent, real-time display of the status of software versions installed at each site. Software updates are transparently sent to the local server during low-load times. Core upgrades are initiated by you and non-critical items show up automatically. This means we can add features and improve scale while you sleep! Zero-Configuration Teleworker - patent-pending! Because of the hybrid-hosted architecture, telecommuters and remote workers can be supported with zero-configuration required. The trixbox Data Center gathers the public IP address of the local trixbox Pro Server so that it can be used by remote IP phones to enable a simple registration process. Combined with DMZ/Host Server firewall settings, this virtually eliminates problems that usually occur in supporting IP phones for remote telecommuters, while protecting the security of your organization's internal systems. Report Generation trixbox Pro has an extensive call data reports (CDR) system where call reports are generated at our Data Center (via the hosted interface). This allows your local premise server to focus on call processing and avoid the CPU spikes often associacated with running large local database queries. Of course, all of your original call details are kept intact at your premise (in case you want to use a third-party reporting engine). TrixBox Pro Comparison Table Features SE w/HUDlite EE w/HUD Pro CCE w/HUD Pro   FREE!             Auto-Attendant (IVR) Outlook Integration Unlimited Extensions** CRM Integration Voicemail Voicemail-to-Email Hot Desk Music-on-Hold Scheduler Custom CTI (AGI) Analog & IP Phones Ring-All (Blast Group) Call Forwarding Name Directory DIDs VoIP-Ready 1 acct 2 accts unlim. PSTN Fallback (patent-pending) Telecommuters Branch Office Support Web-based Control Panel Powerful Reporting Re-Brandable Interface co full full Conference Bridge - Extension Groups - Routing by DIDs - Paging / Zone Paging - Intercom / Zone Intercom - Voicemail Groups - Advanced Call Forwarding - Call Return - Call Out - Report Exporting (.csv) - Custom Caller IDs - IVR Authentication - SMS/Pager Voicemail Notify - Upload Voice Prompts - Alerts & Notifications - Trunks Status Pages - Real-Time System Graphs - Historical System Graphs - Unlimited Call Queues** - - Full Featured A.C.D. - - Skills-based Routing - - Graphical Queue Reports - - Agent Call Recording - - Agent Variable Log-off - - Agents on Cell Phones - - Agents Shared across Sites - - Real-Time Queue Stats - -         HUD       Operator Panel (w/ BLF) *** Call Parking Area Drag & Drop Call Call Control Color-Coded Call Status Drag & Drop to Voicemail Extension Sorting Enterprise Instant Messaging Outlook Integration Presence Management Click-to-Call Mobile Phones Click-to-Email Desktop Alerts Interactive Desktop Alerts - On-the-Fly Recording - Group & User Permissions - Extension Grouping - Extension Search - Extension Search - QuickMenu - Shortcuts (Hotkeys) - Queue Status - - Agent Login/Logout - - Call Barging (active) - - Call Monitoring (passive) - - Web Access to Recordings - - Advanced CRM Integration - - ** Unlimited configuration, hardware platform may impose operational constraints *** The Operator Panel feature requires i) a phone with BLF capability and ii> a phone that is natively supported by Asterisk. 582 Aastra SIP Dect 142 IP Telephone 141.0000 156.00 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/aastra-sip-dect-142-ip-telephone-p-582.html http://www.voipon.co.uk/images/aastra_sip_dect.jpg new Availability: In stock NB: Please note that the Aastra SIP Dect 142 requires either the RFP L32 or RFP L34 in order to operate DECToverIP: Cordless voice communication via IP-networks using SIP Mobility in IP-based networks The convergence of voice and data use in one common infrastructure and network have revolutionised the PBX-market. Transmission of voice is now dominated by the Internet. With OpenMobility IP the mobility disadvantage of VoIP-networks is a thing of the past. A Solution with all the Benefits of DECT DECT (Digital Enhanced Cordless Telecommunication) has proven itself to be the leading, globally used radio-supported technology. DECT radio networks are reliable and offer seamless roaming and handover. Calls remain uninterrupted and a high voice quality is preserved even when changing radio cell areas. Combined Advantages of VoIP and DECT To fully make use of advantages of the IP-network and the DECT-technology together, Aastra has developed DECT-Radio Fixed Parts (RFP) with IP-interfaces for integrating DECT into an IP-network. One OpenMobility Manager (OMM) controls the operation of all DECT-Radio Fixed Parts regardless of the size of the IP network. The OMM is installed in one of the RFPs. A separate server is not required. Administration is via browser interface. The OMM can be fitted with SIP protocol support in order to add DECT mobility to an Asterisk call manager or a SIP enabled PABX. The OMM can handle up to 256 RFPs'. The maximum number of handsets supported is 512. DECToverIP Highlights Outstanding voice quality & data transmission No restriction of connection distances, the range of the mobile network equals the size of the IP-infrastructure Combined usage of mobile voice and data in one network Unitary concept for installation, set up, maintenance and operation Option for easy and cost-efficient networking between sites and for growing existent networks Roaming between sites that are connected via WAN RFP 32 Access Points Features   Voice Mobility Synchronisation via air interface Connection via Ethernet 10/100 BaseT Authentication/Encryption of base and handset Operating states monitoring with 3 LEDs IPv4, Power over LAN according to IEEE 802.3af Platform for the OpenMobility Manager 8 simultaneous Voice and up to 12 Signaling Channels Endpoint for media streams Packet handling via RTP/RTCP DHCP Network Boot, SW-Download / Update Codec G.711, G.723, G.729AB Support of QoS via DiffServ and ToS-Flag Echo Cancellation Jitter compensation Voice Activity Detection with comfort noise Service and Installation Central configuration with the WEB - configurator of the OpenMobility Manager Central cluster administration Central system log book 3 LEDs for signalization of operation status Alternative wiring from below (Cable channel) or from above (suspended ceiling) 583 Aastra 51i IP Telephone 57.5000 65.00 Aastra Aastra IP Telephones http://www.voipon.co.uk/aastra-51i-ip-telephone-p-583.html http://www.voipon.co.uk/images/aastra_51i.jpg new Availability: In stock Exceptional Features & Value in an Entry Level IP Phone The Aastra 51i offers many features and great flexibility in an open-standards based, carrier-grade, basic level IP phone. With a sleek, elegant design and LCD display the 51i is fully interoperable with leading IP telephony platforms, providing advanced XML capability to access custom applications and support for high-quality SIP communications. Part of the 5i Series of Aastra IP phones, the 51i is ideally suited for light use, small / medium business applications such as lobbies or common areas, where single line high-quality SIP communication is required. Main Features XML Browser The Aastra 51i incorporates XML browser capabilities allowing access to customised services and applications. It is possible to create internal service applications using development guides available from Aastra. This feature offers unlimited potential to customize the 51i to meet specific business needs or vertical applications using the display keypad. Enhanced Call Management With extensive storage capacity for personal directory, redial list, and callers log, the Aastra 51i can improve efficiency by providing more call information with the push of a button. Great features such as shared call and bridged line appearances, call forward, call transfer and call waiting increase call flexibility and control. High Quality Audio All Aastra IP phones offer full-duplex speakerphone with superb voice clarity and delivery. With years of experience in telecommunications and telephone design, Aastra can ensure superior voice quality on every product we sell. Simplified Deployment & Implementation From initial deployment and configuration to future enhancements and upgrades, Aastra IP phones are designed to save your business time and money. Dual auto-sensing switched ethernet ports eliminate extra wiring and simplify installations. Integrated IEEE 802.3af Powerover- Ethernet allows easy deployment with centralised powering and backup. Easily created configuration files, using any text editor, can be used to configure phones individually or centrally. 584 Trixbox Pro Softphone Software 25 25.00 Trixbox Trixbox Pro http://www.voipon.co.uk/trixbox-pro-softphone-software-p-584.html http://www.voipon.co.uk/images/trixbox_softphone_license.jpg new Availability: In Stock Trixbox Pro Softphone Software License 585 Trixbox Pro Link (Linking Sites) Lifetime License 370 370.00 Trixbox Trixbox Pro http://www.voipon.co.uk/trixbox-pro-link-linking-sites-lifetime-license-p-585.html http://www.voipon.co.uk/images/trixbox_pro_link.jpg new Availability: In Stock Trixbox Pro Link (Linking Sites) Lifetime License Note: Please note price displayed is for Trixbox Pro Enterprise Edition. For Trixbox Pro Call Centre Edition please select in upgrades 586 Linksys SPA 8000 Analog Telephone Adaptor 185 185.00 Linksys Linksys Analog Adaptors http://www.voipon.co.uk/linksys-spa-8000-analog-telephone-adaptor-p-586.html http://www.voipon.co.uk/images/linksys_spa_8000_small.jpg new Availability: In stock SPA8000 Product Description The Linksys SPA8000 8-Port IP Telephony Gateway is a full featured Analog Terminal Adapter (ATA) for small business enterprises providing enhanced communication services via a broadband connection to the Internet. The SPA8000 features eight RJ-11 FXS ports to connect analog telephones to IP-based data networks and includes a single multi-port RJ-21 50-pin connector offering an alternative connection choice when deploying the telephony gateway in varied environments. The device also has one 10/100Base-T RJ-45 Ethernet interface to connect to either a router or multilayer switch. Solid in design, the SPA8000 is an affordable solution that is ideally suited for use in business and consumer VoIP service offerings including call centers and multi-dwelling environments. Customers can also protect and extend their investments by continuing to utilize their existing analog telephones and teleconferencing equipment. Installed by the end user and remotely provisioned, configured and maintained by the service provider, each SPA8000 converts voice traffic into data packets for transmission over an IP network and uses common standards for voice and data networking for reliable voice and fax operation. Toll Quality Voice and Carrier-Grade Feature Support The SPA8000 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via our advanced implementation of standard voice coding algorithms. The SPA8000 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems. Large-Scale Deployment and Management The SPA8000 offers all the key features and capabilities with which service providers can provide customized services to their subscribers. The SPA8000 can be remotely provisioned and supports dynamic, in-service software upgrades. A secure profile upload saves providers the time, expense and hassle of managing and pre-configuring or re-configuring customer premise equipment (CPE) for deployment. Ironclad Security Linksys understands that security for both end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The SPA8000 supports secure, encryption-based methods for communication, provisioning and servicing. SPA8000 Features Eight voice ports (RJ11) for analog phones or Fax machines Impedance Agnostics - 8 Configurable Settings Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Caller ID with Name/Number (Multi-national Variants) Caller ID Blocking Call Forwarding: No answer, Busy, All Do Not Disturb Call Transfer Three-way Conference Calling with Local Mixing Message Waiting Indication - Visual and Tone Based Call Return Call Back on Busy Call Blocking with Toll Restriction Delayed Disconnect Distinctive Ringing - Calling and Called Number Off-hook Warning Tone Selective/Anonymous Call Rejection Touch Tone Phone Keypad Configuration with Interactive Voice Response (IVR) Fax: G.711 Pass Through or Real Time Fax over IP via T.38* *T.38 support is dependent on fax machine and network/transport resilience Package Contents 1 - SPA8000 Phone Adapter Unit 1 - 12v Power Adapter 1 - RJ45 Ethernet Cable 1- Quick Installation Guide SPA8000 Specifications * Note: Many specifications are programmable within a defined range or list of options. Please see the SPA Administration Guide for details. The target configuration profile is uploaded to the SPA8000 at the time of provisioning. MAC Address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) ARP - Address Resolution Protocol DNS - A Record (RFC 1706), SRV Record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) DHCP Server - Dynamic Host Configuration Protocol (RFC 2131) PPoE Client - Point to Point Protocol over Ethernet (RFC 2516) ICMP - Internet Control Message Protocol (RFC792)TCP - Transmission Control Protocol (RFC793) UDP - User Datagram Protocol (RFC768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) DiffServ (RFC 2475), Type of Service - TOS (RFC 791/1349) VLAN Tagging - 802.1p SNTP - Simple Network Time Protocol (RFC 2030) Upload Data Rate Limiting - Static and Automatic QoS - Voice Packet Prioritization over Other Packet Types MAC Address Cloning Port Forwarding SIP channels support both UDP and TCP transport VPN Pass-Through with IPSec ESP, PPTP, and L2TP SIPv2: Session Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264) - SIP Proxy Redundancy - Dynamic via DNS SRV, A Records - Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks - NAT (incl. STUN) Secure (Encrypted) Calling via Pre-Standard Implementation of Secure RTP Codec Name Assignment - G.711 (a-law and &#956;-Law) - G.726 (16/24/32/40 kbps) - G.729 A - G.723.1 (6.3 kbps, 5.3 kbps) Dynamic PayloadAdjustable Audio Frames per Packet Fax Tone Detection Pass-Through Fax Pass-Though - Using G.711 DTMF: In-band & Out-of-band (RFC 2833) (SIP Info) Flexible Dial Plan Support with Interdigit Timers and IP Dialing Call Progress Tone Generation Jitter Buffer - Adaptive Frame Loss Concealment Full Duplex Audio Echo Cancellation (G.165/G.168) VAD - Voice Activity Detection with Silence Suppression Comfort Noise Generation (CNG) QoS (Ethernet port up-stream bandwidth control, Physical port, MAC address, Application) Attenuation / Gain Adjustments Flash Hook Timer MWI - Message Waiting Indicator Tones VMWI - Visual Message Waiting Indicator via FSK Polarity ControlHook Flash Event Signalling Caller ID Generation (Name & Number) - Bellcore, DTMF, ETSI Music on Hold Client Streaming Audio Server - up to 10 sessions System DDR SDRAM - 16M Bytes System Flash ROM - 4M Bytes Security Password Protected System Reset to Factory Default Password Protected Admin and User Access Authority Provisioning/Configuration/Authentication: - HTTPS with Factory Installed Client Certificate HTTP Digest - Encrypted Authentication via MD5 (RFC 1321) Authentication: EAP-TLS, EAP-TTLS, and EAP-PEAP SIP TLS (Transport Layer Security) Up to 256-bit AES Encryption Provisioning, Administration & Maintenance Web Browser Administration & Configuration via Integral Web Server Telephone Key Pad Configuration with Interactive Voice Prompts Automated Provisioning & Upgrade via HTTPS, HTTP, TFTP Asynchronous Notification of Upgrade Availability via NOTIFY Non-intrusive, In-Service Upgrades Report Generation & Event Logging Stats in BYE Message Syslog & Debug Server Records Per Line and Purpose Configurable Syslog and Debug Options Physical Interfaces 8 port standard FXS voice ports (RJ-11) RJ-21 (50-pin telco connector) multi-port voice connection 1 WAN 100baseT RJ-45 Ethernet Port (IEEE 802.3) Reset button Subscriber Line Interface Circuit (SLIC) Ring Voltage: 40-90 Vpk Configurable Ring Frequency: 20 Hz - 25 Hz Ring Waveform: Trapezoidal Maximum Ringer Load: 5 REN On-hook/off-hook Characteristics: - On-hook voltage (tip/ring): -46 ~ -56V - Off-hook current: 18-25 mA - Terminating Impedance: 600 ohm resistive or 270 ohm + 750ohm// 150nF complex impedance Frequency Response 300 - 3400Hz Return Loss (600 ohm, 300-3400Hz) Up to 20dB Insertion Loss (1 Vrms @1 kHz) 3dB - 4dB THD (350 mV peak @ 300 Hz) Up to 3% Idle Channel Noise -72 dB (typ.) Longitudinal Balance 55 dB (typ.) Off Hook Threshold (Line Seizure) Rdc On Hook Threshold (Line Release) Rdc >10000 ohm Rdc DC Supervisory Range Rdc > 450 ohm Regulatory Compliance FCC (Part 15 Class B), CE, ICES-003, C-Tick Certification, RoHS, UL Power Supply Switching Type (100-240v) Automatic DC Input Voltage: 12V DC at 3.0 A Max. Power Adapter: 100-240v - 50-60Hz AC Input Indicator Lights/LED Power, Ethernet, USB, Voice Status, Phone 1,2,3,4,5,6,7,8 Documentation Quick Installation, User, and Configuration Guides are downloaded from www.Linksys.com Administration Guide - Service Providers Only Provisioning Guide - Service Providers Only Environmental Dimensions 6.69 x 1.54 x 8.66&prime; (170 x 39 x 220 mm) W x H x D Weight 2.85 lbs (1.30 kg) Operating Temp. 32 to 113°F (0°C to 45°C) Storage Temp. -13 to 185°F (-20°C to 85°Operating Humidity 10% to 90% Non-Condensing, operating and non-operating Storage Humidity 10% to 90% Non-Condensing, operating and non-operating 587 Voismart V2GSM 2 Port Card 560.0000 625.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-2-port-card-p-587.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 588 Voismart V4GSM 4 Port Card 816.0000 960.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v4gsm-4-port-card-p-588.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 589 Voismart V4BRI 4 Port BRI Card 272.0000 335.00 VoiSmart Voismart Asterisk BRI Card http://www.voipon.co.uk/voismart-v4bri-4-port-bri-card-p-589.html http://www.voipon.co.uk/images/voismart_visdn.jpg new Availability: In Stock VoiSmart v4BRI ISDN board is a PCI bus ISDN adapter, based on Cologne Chip chipset, with 4 basic rate ports. Each port can be configured, separately, as a terminal (TE) or network (NT) port, it can be optionally given a 120 Ohm termination and can optionally supply power to the bus. VoiSmart vISDN is fully supported by vISDN (http://www.visdn.org/), the multiservice ISDN architecture sponsored by Espia, owner of VoiSmart registered trademark, which allows the full exploitation of the board possibilities and features. ISDN is 100% Asterisk compatible. ISDN protocol implementation has been certified by Telecom Italia Labs, according to TBR3 requirements and the strict compatibility constraints of Italian Public Network, as required by Telecom Italia. VoiSmart vISDN ISDN board has also two connectors for an external PCM bus to link other vISDN boards or E1 boards, or a hardware DSP, soon available by VoiSmart. Technical features: 2.2 PCI Bus (3.3V supply) Interfaces: 4 BRI ports, RJ-45 plugs Port configuration: NT/TE, jumper selectable 120 Ohm / High Impedance terminations, dipswitch selectable Power supply: enable/disable via jumper HDLC integrated for each channel One 2-color LED, on each port 590 Xorcom Astribank-BRI 8 FXS 4 BRI 483.0000 544.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-8-fxs-4-bri-p-590.html http://www.voipon.co.uk/images/xr0029_small.jpg new Availability: In Stock Built-in Features FXS Ports: 8 FXS BRI ISDN Ports: 4 Chassis (Default): Compact BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 591 Xorcom Astribank-BRI 16 FXS 4 BRI 685.0000 772.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-16-fxs-4-bri-p-591.html http://www.voipon.co.uk/images/xr0029.jpg new Availability: In Stock Built-in Features FXS Ports: 16 BRI ISDN Ports: 4 Chassis (Default): 19" 1U BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 592 Xorcom Astribank-BRI 24 FXS 4 BRI 885.0000 997.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-24-fxs-4-bri-p-592.html http://www.voipon.co.uk/images/xr0032_small.jpg new Availability: In Stock Built-in Features FXS Ports: 24 BRI ISDN Ports: 4 Chassis (Default): 19" 1U BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 593 Xorcom Astribank-BRI 8 FXS 2 BRI 437.0000 492.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-8-fxs-2-bri-p-593.html http://www.voipon.co.uk/images/xr0033_small.jpg new Availability: In Stock Built-in Features FXS Ports: 8 Ports BRI ISDN Ports: 2 Chassis (Default): Compact BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 594 Xorcom Astribank-BRI 8 FXS 8 BRI 587.0000 661.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-8-fxs-8-bri-p-594.html http://www.voipon.co.uk/images/xr0042_small.jpg new Availability: In Stock Built-in Features FXS Ports: 8 BRI ISDN Ports: 8 Chassis (Default): Compact BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 595 Xorcom Astribank-BRI 16 FXS 8 BRI 786.0000 886.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-16-fxs-8-bri-p-595.html http://www.voipon.co.uk/images/xr0043_small.jpg new Availability: In Stock Built-in Features FXS Ports: 16 BRI ISDN Ports: 8 Chassis (Default): 19" 1U BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 596 Xorcom Astribank-BRI 24 FXS 8 BRI 983.0000 1108.00 Xorcom Xorcom Channel Banks http://www.voipon.co.uk/xorcom-astribankbri-24-fxs-8-bri-p-596.html http://www.voipon.co.uk/images/xr0044_small.jpg new Availability: In Stock Built-in Features FXS Ports: 24 BRI ISDN Ports 8 Chassis (Default): 19" 1U BRI (Basic Rate Interface) ISDN is a digital telephony interface used to connect to the telephony provider, legacy PBX or ISDN telephone sets. There are two basic configurations for each BRI port - TE (Terminal) - connects to telephony provider and NT (Network) - connects to ISDN phones (requires power supply option) or legacy PBX (as PSTN ISDN trunks). Astribank BRI module has three models: XR00.. with 2 ports, XR00.. with 4 ports and XR00.. with 8 ports. BRI module may be available as part of Astribank full size (1U) units as well as in the compact units. One BRI module (with up to eight BRI ports) may be installed in an Astribank unit. Features: 2, 4 or 8 BRI ISDN ports Each port is user configurable - NT/TE Power supply for ISDN telephone sets (Optional) LED status indication for each port including NT/TE status Available in combined units with Analog ports (see XR0029 picture) 597 Vegastream Vega 5024 24 FXS + 2 FXO 1089.0000 1210.00 Vegastream Vegastream Vega 5000 http://www.voipon.co.uk/vegastream-vega-5024-24-fxs-2-fxo-p-597.html http://www.voipon.co.uk/images/5024.jpg new Availability: In Stock The Vega 5000 VoIP Gateway range is the Winner of the Internet Telephony 2005 Product of the Year Award and is designed to connect analog phone systems to IP networks. Available with 24 or 48 FXS ports, the Vega 5000 analog VoIP Gateway is designed to connect standard analog handsets typically to an Internet Telephony Service Provider (ITSP) or a corporate network. It also offers two additional network-facing analog ports for connection to a telephone line or analog PBX. The Vega 5000 VoIP Gateway range is the Winner of the Internet Telephony 2005 Product of the Year Award Features & Benefits Open, Non-Proprietary, Configurable Interfaces All VegaStream VoIP Gateways support SIP, H.323 and T.38 Fax, resulting in significant levels of interoperability with a wide range of existing telecommunications and VoIP equipment. The Vega 5000 is configurable for different country requirements including tones and line impediance. This ensures that IP connected networks with multinational site locations can be supported with confidence in virtually all instances. Lifeline PSTN Backup When powered, the Vega 5000 VoIP Gateway will route calls to or from the unit's two FXO ports. Under power failure conditions, the two FXO ports provide a hard-wired bypas feature to two FXS ports. This allows PSTN calls still to be made, giving resilience and confidence of business continuity, even under power failure conditions. Density With up to 48 FXS ports in a single 1U high enclosure which is only half a rack deep, VegaStream's range of VoIP Gateways provide exceptional density for deployment, conserving space within a communications room. 598 Vegastream Vega 5048 48 FXS 2 FXO 1950.0000 2105.00 Vegastream Vegastream Vega 5000 http://www.voipon.co.uk/vegastream-vega-5048-48-fxs-2-fxo-p-598.html http://www.voipon.co.uk/images/5048.jpg new Availability: In Stock The Vega 5000 VoIP Gateway range is the Winner of the Internet Telephony 2005 Product of the Year Award and is designed to connect analog phone systems to IP networks. Available with 24 or 48 FXS ports, the Vega 5000 analog VoIP Gateway is designed to connect standard analog handsets typically to an Internet Telephony Service Provider (ITSP) or a corporate network. It also offers two additional network-facing analog ports for connection to a telephone line or analog PBX. The Vega 5000 VoIP Gateway range is the Winner of the Internet Telephony 2005 Product of the Year Award Features & Benefits Open, Non-Proprietary, Configurable Interfaces All VegaStream VoIP Gateways support SIP, H.323 and T.38 Fax, resulting in significant levels of interoperability with a wide range of existing telecommunications and VoIP equipment. The Vega 5000 is configurable for different country requirements including tones and line impediance. This ensures that IP connected networks with multinational site locations can be supported with confidence in virtually all instances. Lifeline PSTN Backup When powered, the Vega 5000 VoIP Gateway will route calls to or from the unit's two FXO ports. Under power failure conditions, the two FXO ports provide a hard-wired bypas feature to two FXS ports. This allows PSTN calls still to be made, giving resilience and confidence of business continuity, even under power failure conditions. Density With up to 48 FXS ports in a single 1U high enclosure which is only half a rack deep, VegaStream's range of VoIP Gateways provide exceptional density for deployment, conserving space within a communications room. 601 Aastra DECToverIP RFP (Radio Fixed Parts) L32 459.0000 510.00 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/aastra-dectoverip-rfp-radio-fixed-parts-l32-p-601.html http://www.voipon.co.uk/images/aastra_dectoverip_l32.jpg new Availability: In stock Please Note: The RFP L32 is for use with the Aastra DECToverIP 142 Application - RFP L32: Indoors, PoE or power supply DECToverIP: Cordless voice communication via IP-networks using SIP New freedom: mobility in IP telephony networks The convergence of use for voice and data in one common infrastructure and network have revolutionized the PBX-market. Transmission of voice is now dominated by voIP. With OpenMobility IP the disadvantage of VoIP-networks for having restric ted mobility is a thing of the past. DECT benefits, in particular voice quality, are undiminished with Aastra wireless DECT (Digital Enhanced Cordless Telecommunication) has proven itself to be the leading, globally used radio-supported technology. DECT radio networks are reliable and provide seamless roaming and handover. Calls remain uninterrupted and an unmatched high voice quality is maintained even when changing radio cell areas. Combined benefits of VoIP and DECT To harness the unique advantages of the IP-network and the DECT-technology together, Aastra has developed DECT-Radio Fixed Parts (RFP) with IP-interfaces for integrating DECT into the IP-network. One OpenMobility Manager (OMM) controls the operation of all DECT-RFP irrespective of the size of the IP-network. The OMM is installed in one of the RFPs'. A separate server is not necessary. Administration is via a browser interface. The OMM can be fitted with SIP protocol support in order to add DECT mobility to an Asterisk-based call manager or a SIP enabled PABX. The OMM can handle up to 256 RFPs'. The maximum number of handsets supported is 512.   DECToverIP Benefits Outstanding voice quality & data transmission No restriction of connection distances, the range of the mobile network equals the size of the IP-infrastructure Combined use of mobile voice and data in one network Unitary concept for installation, set up, maintenance and service Option for easy and cost-efficient networking between sites and for growing existent networks Roaming between sites that are connected via WAN   RFP 32 Access Points Features   Voice Mobility Synchronisation via air interface Authentication/Encryption of base and handset Operating states monitoring with 3 LEDs Connection via Ethernet 10/100 BaseT 802.3af IPv4, Power over LAN according to IEEE Platform for the OpenMobility Manager 8 simultaneous Voice and up to 12 Signaling Channels Endpoint for media streams Network Boot, SW-Download / Update Packet handling via RTP/RTCP DHCP Codec G.711, G.723, G.729AB Support of QoS via DiffServ and ToS-Flag Jitter compensation Echo Cancellation Voice Activity Detection with comfort noise Service and Installation Central configuration with the WEB - configurator of the OpenMobility Manager Central system log book 3 LEDs for signalization of operation status Central cluster administration Alternative wiring from below (Cable channel) or from above (suspended ceiling) 603 Vegastream Vega 400 - 1 x E1 (30 Channels) 2480.0000 2731.00 Vegastream Vegastream Vega 400 http://www.voipon.co.uk/vegastream-vega-400-1-x-e1-30-channels-p-603.html http://www.voipon.co.uk/images/vegastream_400.jpg new Availability: In Stock The Vega 400 digital gateway connects digital telephony equipment to IP networks. All Vega 400 gateways have four E1 / T1 interfaces and can be purchased with various VoIP capacities up to 120 VoIP channels. The Vega 400 has been designed with future expansion in mind and consists of: Base Vega 400 unit with expansion slots Field-installable expansion modules to provide additional VoIP channel capacity A Vega 400 unit can therefore be installed with one VoIP channel capacity and then later additional capacity can be provisioned as and when it is needed. Each E1 / T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can, therefore, be connected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Open, Non-Proprietary Interfaces The Vega 400 gateway supports the following signaling schemes: ETSI ISDN NI1, NI2, AT&T 5ESS, DMS100 ISO QSIG Basic Call and QSIG feature transparency Channel Associated Signaling (CAS) All VegaStream gateways can support SIP, H.323 and T.38 fax. The Vega 400 gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment. Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.723.1 (5.3/6.4 kbps) G.729a (8 kbps) G.711 (a-law/µ-law) (64 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 120 VoIP channels Telephony Interface Primary Rate ISDN (User configurable NT/TE): 4 x E1 Euro-ISDN ISO QSIG QSIG Feature Transparency (H.323) CAS R2MFC 4 x T1 NI1/NI2 AT&T 5ESS DMS100 CAS (RBS) - E&M wink start - Loop start - Ground start ISO QSIG QSIG Feature Transparency (H.323) LAN Interface 2 x 10 BaseT / 100 BaseTX, full or half duplex Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Certification Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power ISDN: NT/TE and Link up > LAN: Speed, Activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Power 100..240 VAC, 47..63 Hz, 1..0.5 A -48V DC also available, 1.2A (Max) 604 Vegastream Vega 400 - 2 x E1 (60 Channels) 3805.0000 4187.00 Vegastream Vegastream Vega 400 http://www.voipon.co.uk/vegastream-vega-400-2-x-e1-60-channels-p-604.html http://www.voipon.co.uk/images/vegastream_400.jpg new Availability: In Stock The Vega 400 digital gateway connects digital telephony equipment to IP networks. All Vega 400 gateways have four E1 / T1 interfaces and can be purchased with various VoIP capacities up to 120 VoIP channels. The Vega 400 has been designed with future expansion in mind and consists of: Base Vega 400 unit with expansion slots Field-installable expansion modules to provide additional VoIP channel capacity A Vega 400 unit can therefore be installed with one VoIP channel capacity and then later additional capacity can be provisioned as and when it is needed. Each E1 / T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can, therefore, be connected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Open, Non-Proprietary Interfaces The Vega 400 gateway supports the following signaling schemes: ETSI ISDN NI1, NI2, AT&T 5ESS, DMS100 ISO QSIG Basic Call and QSIG feature transparency Channel Associated Signaling (CAS) All VegaStream gateways can support SIP, H.323 and T.38 fax. The Vega 400 gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment. Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.723.1 (5.3/6.4 kbps) G.729a (8 kbps) G.711 (a-law/µ-law) (64 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 120 VoIP channels Telephony Interface Primary Rate ISDN (User configurable NT/TE): 4 x E1 Euro-ISDN ISO QSIG QSIG Feature Transparency (H.323) CAS R2MFC 4 x T1 NI1/NI2 AT&T 5ESS DMS100 CAS (RBS) - E&M wink start - Loop start - Ground start ISO QSIG QSIG Feature Transparency (H.323) LAN Interface 2 x 10 BaseT / 100 BaseTX, full or half duplex Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Certification Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power ISDN: NT/TE and Link up > LAN: Speed, Activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Power 100..240 VAC, 47..63 Hz, 1..0.5 A -48V DC also available, 1.2A (Max) 605 Vegastream Vega 400 - 3 x E1 (90 Channels) 4789.0000 5267.00 Vegastream Vegastream Vega 400 http://www.voipon.co.uk/vegastream-vega-400-3-x-e1-90-channels-p-605.html http://www.voipon.co.uk/images/vegastream_400.jpg new Availability: In Stock The Vega 400 digital gateway connects digital telephony equipment to IP networks. All Vega 400 gateways have four E1 / T1 interfaces and can be purchased with various VoIP capacities up to 120 VoIP channels. The Vega 400 has been designed with future expansion in mind and consists of: Base Vega 400 unit with expansion slots Field-installable expansion modules to provide additional VoIP channel capacity A Vega 400 unit can therefore be installed with one VoIP channel capacity and then later additional capacity can be provisioned as and when it is needed. Each E1 / T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can, therefore, be connected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Open, Non-Proprietary Interfaces The Vega 400 gateway supports the following signaling schemes: ETSI ISDN NI1, NI2, AT&T 5ESS, DMS100 ISO QSIG Basic Call and QSIG feature transparency Channel Associated Signaling (CAS) All VegaStream gateways can support SIP, H.323 and T.38 fax. The Vega 400 gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment. Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.723.1 (5.3/6.4 kbps) G.729a (8 kbps) G.711 (a-law/µ-law) (64 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 120 VoIP channels Telephony Interface Primary Rate ISDN (User configurable NT/TE): 4 x E1 Euro-ISDN ISO QSIG QSIG Feature Transparency (H.323) CAS R2MFC 4 x T1 NI1/NI2 AT&T 5ESS DMS100 CAS (RBS) - E&M wink start - Loop start - Ground start ISO QSIG QSIG Feature Transparency (H.323) LAN Interface 2 x 10 BaseT / 100 BaseTX, full or half duplex Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Certification Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power ISDN: NT/TE and Link up > LAN: Speed, Activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Power 100..240 VAC, 47..63 Hz, 1..0.5 A -48V DC also available, 1.2A (Max) 606 Vegastream Vega 400 - 4 x E1 (120 Channels) 5131.0000 5643.00 Vegastream Vegastream Vega 400 http://www.voipon.co.uk/vegastream-vega-400-4-x-e1-120-channels-p-606.html http://www.voipon.co.uk/images/vegastream_400.jpg new Availability: In Stock The Vega 400 digital gateway connects digital telephony equipment to IP networks. All Vega 400 gateways have four E1 / T1 interfaces and can be purchased with various VoIP capacities up to 120 VoIP channels. The Vega 400 has been designed with future expansion in mind and consists of: Base Vega 400 unit with expansion slots Field-installable expansion modules to provide additional VoIP channel capacity A Vega 400 unit can therefore be installed with one VoIP channel capacity and then later additional capacity can be provisioned as and when it is needed. Each E1 / T1 interface can be independently configured as network side or terminal side. The Vega 400 gateway can, therefore, be connected to a PBX and the PSTN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Open, Non-Proprietary Interfaces The Vega 400 gateway supports the following signaling schemes: ETSI ISDN NI1, NI2, AT&T 5ESS, DMS100 ISO QSIG Basic Call and QSIG feature transparency Channel Associated Signaling (CAS) All VegaStream gateways can support SIP, H.323 and T.38 fax. The Vega 400 gateway has proven interoperability with a wide range of existing telecommunications and VoIP equipment. Interfaces VoIP Protocols SIP H.323 version 4 Audio codecs: G.723.1 (5.3/6.4 kbps) G.729a (8 kbps) G.711 (a-law/µ-law) (64 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 120 VoIP channels Telephony Interface Primary Rate ISDN (User configurable NT/TE): 4 x E1 Euro-ISDN ISO QSIG QSIG Feature Transparency (H.323) CAS R2MFC 4 x T1 NI1/NI2 AT&T 5ESS DMS100 CAS (RBS) - E&M wink start - Loop start - Ground start ISO QSIG QSIG Feature Transparency (H.323) LAN Interface 2 x 10 BaseT / 100 BaseTX, full or half duplex Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Certification Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power ISDN: NT/TE and Link up > LAN: Speed, Activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Power 100..240 VAC, 47..63 Hz, 1..0.5 A -48V DC also available, 1.2A (Max) 607 Aastra DectOverIP RFP (Radio Fixed Parts) L34 512.0000 569.00 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/aastra-dectoverip-rfp-radio-fixed-parts-l34-p-607.html http://www.voipon.co.uk/images/aastra_dectoverip_l34.jpg new Availability: In stock Application - RFP L34: Outdoors, external antennas, PoE only DECToverIP: Cordless voice communication via IP-networks using SIP New freedom: mobility in IP-based networks The convergence of use for voice and data in one common infrastructure and network have revolutionised the PBX-market. Transmission of voice is now dominated by voIP. With OpenMobility IP the disadvantage of VoIP-networks of restricted mobility is a thing of the past. DECT benefits, in particular the quality of voice transmission, are undiminished with Aastra RFP DECT (Digital Enhanced Cordless Telecommunication) has proven itself to be the world-leading radio-supported technology. DECT radio networks are reliable and provide seamless roaming and handover. Calls remain uninterrupted and an unmatched high voice quality is preserved even when changing radio cell areas. Combined advantages of VoIP and DECT To utilise the unique advantages of the IP-network and the DECT-technology together, Aastra has developed DECT-Radio Fixed Parts (RFP) with IP-interfaces for integrating DECT into the IP-network. One OpenMobility Manager (OMM) controls the operation of all DECT-Radio Fixed Parts irrespective of the size of the IP-network. The OMM is installed in one of the RFPs'. A separate server is not required. Administration is via browser interface. The OMM can be fitted with SIP protocol support to add DECT mobility to an Asterisk-based call manager or a SIP enabled PABX. The OMM can handle up to 256 RFPs'. The maximum number of handsets supported is 512. DECToverIP Highlights Exceptional voice quality and data transmission No restriction of connection distances, the range of the mobile network equals the size of the IP-infrastructure Combined usage of mobile voice and data in one network Unitary concept for installation, setup, maintenance and service Option for easy and cost-efficient networking between sites and for growing existent networks Roaming between sites that are connected via WAN   RFP 32 Access Points Features Voice Mobility Synchronisation via air interface Authentication/Encryption of base and handset Connection via Ethernet 10/100 BaseT Operating states monitoring with 3 LEDs IPv4, Power over LAN according to IEEE 802.3af Platform for the OpenMobility Manager 8 simultaneous Voice and up to 12 Signaling Channels Endpoint for media streams DHCP Packet handling via RTP/RTCP Network Boot, SW-Download / Update Codec G.711, G.723, G.729AB Support of QoS via DiffServ and ToS-Flag Jitter compensation Echo Cancellation Voice Activity Detection with comfort noise Service and Installation Central configuration with the WEB - configurator of the OpenMobility Manager Central cluster administration Central system log book 3 LEDs for signalization of operation status Alternative wiring from below (Cable channel) or from above (suspended ceiling) 608 Trixbox Appliance Base 695 695.00 Trixbox Trixbox Appliance http://www.voipon.co.uk/trixbox-appliance-base-p-608.html http://www.voipon.co.uk/images/trixbox_appliance_base.jpg new Availability: Late June 08 - Accepting Pre Orders Rhino Ceros PBX A wide range of Asterisk boxes is available from Rhino. Why not take a look at some of these great value high quality alternatives ? The trixbox Appliance entry point, this package is an all-in-one trixbox solution configured for VoIP telephony. Hardware specifications include an Intel P4 Processor, 512 MB RAM, 80GB Dual HD's with RAID, and dual NIC (onboard network adapter and additional NIC with 4-port switch). trixbox Appliance Base with Intel P4 Processor, 512 MB RAM, 80GB Dual HD's with RAID, and dual NIC Unlike other telephony appliances, the trixbox Appliance fits into any business. Literally! The trixbox Appliance features 5 different mounting options. From a 19" telephony relay rack mount or cabinet slide-mount, to wall mount and free-standing options, the trixbox Appliance is guaranteed to fit your business needs. Built for the small and medium business, the trixbox Appliance is not a "mini-appliance." In fact, the trixbox Appliance can scale up to 4 T1s or 48 analogue lines. The trixbox appliance supports Asterisk-compatible cards from virtually any telephony card manufacturer. Reliability is key for any business, whatever the size. With the dual power supply, the trixbox Appliance Enterprise Edition offers high reliability with the ability to run on multiple power sources; a feature you won't find on any other telephony appliance. In addition, the second hard drive offers data mirroring, ensuring that your data is always safe from a hardware failure. Featuring two network cards, the trixbox Appliance can be networked in a variety of configurations to meet your specific business needs. To add even more value, one of the network cards features an onboard 4- port switch. What's that loud noise you hear? It's not your trixbox Appliance! With its intriguing one-of-a-kind custom front panel and limited fan noise, the trixbox Appliance was designed to be seen, not heard. With a back-lit 4 line LCD included prominently on the front of your trixbox Appliance, you can check the status of the system, including the status of calls and queues, all in a single glance. trixbox Hardware Specifications 3U chassis, 19" rack mountable. Mounting options: o 19" telephony relay rack mount o 19" cabinet slide-mount o Wall mount o Free-standing floor o Free-standing tabletop 3Ghz P4 1MB Cache 512MB RAM (1 GB with Enterprise Edition) 80GB 7200 RPM SATA HD's (disk mirroring) Dual power supply (Enterprise Edition) 2 PCI slots (up to 6 Sangoma cards using Remora expansion cards; system supports 7 full-length bays) Dual NIC; one NIC has a 4-port onboard switch 4-line LCD display 12 month manufacturer's warranty 609 Trixbox Enterprise Appliance Base 1145 1145.00 Trixbox Trixbox Appliance http://www.voipon.co.uk/trixbox-enterprise-appliance-base-p-609.html http://www.voipon.co.uk/images/trixbox_appliance_base.jpg new Availability: Late June 08 - Accepting Pre Orders Rhino Ceros PBX A wide range of Asterisk boxes is available from Rhino. Why not take a look at some of these great value high quality alternatives ? Serious Business? trixbox Enterprise Appliances add more memory and redundant power. This package includes an Intel P4 Processor, 1GB RAM, 80GB Dual HDs with RAID, dual NIC (onboard network adapter and additional NIC with 4-port switch), and a fully redundant power supply; pre-configured for VoIP connectivity. Unlike other telephony appliances, the trixbox Appliance fits into any business. Literally! The trixbox Appliance features 5 different mounting options. From a 19? telephony relay rackmount or cabinet slide-mount, to wall mount and free-standing options, the trixbox Appliance is guaranteed to fit your business needs. Built for the small and medium business, the trixbox Appliance is not a ?mini-appliance.? In fact, the trixbox Appliance can scale up to 4 T1s or 48 analog lines. The trixbox appliance supports Asterisk-compatible cards from virtually any telephony card manufacturer. Reliability is key for any business, whatever the size. With the dual power supply, the trixbox Appliance Enterprise Edition offers high reliability with the ability to run on multiple power sources; a feature you won?t find on any other telephony appliance. In addition, the second hard drive offers data mirroring, ensuring that your data is always safe from a hardware failure. Featuring two network cards, the trixbox Appliance can be networked in a variety of configurations to meet your specific business needs. To add even more value, one of the network cards features an onboard 4- port switch. What?s that loud noise you hear? It?s not your trixbox Appliance! With its intriguing one-of-a-kind custom front panel and limited fan noise, the trixbox Appliance was designed to be seen, not heard. With a back-lit 4 line LCD included prominently on the front of your trixbox Appliance, you can check the status of the system, including the status of calls and queues, all in a single glance. trixbox Hardware Specifications 3U chassis, 19" rackmountable. Mounting options: 19" telephony relay rackmount 19" cabinet slide-mount Wall mount Free-standing floor Free-standing tabletop 3Ghz P4 1MB Cache 512MB RAM (1 Gb with Enterprise Edition) 80GB 7200 RPM SATA HDs (disk mirroring) Dual power supply (Enterprise Edition) 2 PCI slots (up to 6 Sangoma cards using Remora expansion cards; system supports 7 full-length bays) Dual NIC; one NIC has a 4-port onboard switch 4-line LCD display 12 month manufacturer's warranty 610 Vegastream Vega 50 6x4 - 8 x FXO 873.0000 970.00 Vegastream Vegastream Vega 50 6x4 http://www.voipon.co.uk/vegastream-vega-50-6x4-8-x-fxo-p-610.html http://www.voipon.co.uk/images/vega_50_6x4.gif new Availability: In Stock The Vega 50 6x4 gateway connects a range of legacy telephony equipment, including PBXs, ISDN telephones, the ISDN, analog phones and the PSTN to IP networks. The Vega 50 6x4 supports up to 24 analog ports, up to 16 basic rate ISDN channels on 8 interfaces or a mixture of port types and capacities. The Vega 50 6x4 is available factory configured in a number of different configurations, other configurations are available to special order. The following gateway is configured as follows : 4 FXO Each BRI interface can be independently configured as network side or terminal side. The Vega 50 6x4 gateway can, therefore, be connected to a PBX and the ISDN simultaneously. This configuration provides: No disruption to the configuration of existing equipment Flexibility and choice for call routing Analog interfaces on the Vega 50 6x4 can be configured as 600R, 900R or CTR-21 Open, Non-Proprietary Interfaces The Vega 50 6x4 BRI interfaces supports ETSI BRI. The Vega 50 6x4, analog interfaces support standard loop start signalling. The Vega 50 6x4 gateways have proven interoperability with a wide range of existing telecommunications and VoIP equipment. All VegaStream gateways can support SIP, H.323 and T.38 fax. The gateway can be configured for different country requirements, such as tones and line impedance. Service Provider Applications: Customer premises gateway PSTN gateway Enterprise Applications: Enterprise VoIP networking Lifeline PSTN Backup Vega 50 6x4 variants equipped with FXS ports are also fitted with two FXO ports. When powered the Vega can route calls to or from these two FXO ports. Under power failure conditions the two FXO ports provide a hard-wired bypass to two FXS ports allowing PSTN calls to be made even under this failure condition. Interfaces VoIP Protocols SIP H.323 version 4Audio codecs: - G.711 (a-law/µ-law) (64 kbps) - G.729a (8kbps) - G.723.1 (5.3/6.4 kbps) Fax Support - up to G3 fax, using T.38 Modem Support - up to V.90, using G.711 Up to 26 VoIP channels Features Identification Caller ID presentation Caller ID screening allows connections to be accepted only from selected call sources H.323 gatekeeper registration SIP Registration and Digest Authentication Operations, Maintenance and Billing HTTP web server Radius Accounting Remote firmware upgrade: Auto code upgrade Auto configuration upgrade SNMP TFTP/FTP support VT100 - RS232/Telnet Routing and Numbering Dial Planner - sophisticated call routing capabilities, standalone or gatekeeper/proxy integration Direct Dialing In (DDI) SIP registration to multiple proxies FXS ports support Call Waiting and Call transfer (blind and consultative) NAT traversal Call Quality Adaptive jitter removal Comfort noise generation Silence suppression 802.1p/Q VLAN tagging Differentiated Services (DiffServ) Type of Service (ToS) QoS statistics reporting Echo cancellation (G.168 up to 128ms) Hardware Environmental 0° .. 40°C 0% .. 90% humidity (non-condensing) Indicators LED: Power Vega ready Lifeline, FXS, FXO, BRI calls active BRI link up LAN: speed / activity Physical Dimensions 437mm (17.2") x 43mm (1.7") x 275mm (10.8") width/height/depth Industrial rackmount: 483mm (19"), 1U Weight: 6.5kgs Accessories 1 x RJ21 to punch down Connector 1 x RG21 to RJ45 cable (1.5m) Power 100..240 VAC, 47..63 Hz, 1..0.5 A Program Storage Code and configuration data are stored in FLASH and executed from RAM 611 Polycom VSX 6000 Video Conferencing Device 2385.0000 2649.00 Polycom Polycom IP Video Conferencing http://www.voipon.co.uk/polycom-vsx-6000-video-conferencing-device-p-611.html http://www.voipon.co.uk/images/polycom_vsx_6000.jpg new Availability: 2-3 days from order Features and Benefits Audio G.722.1 Annex C 14 kHz crystal clear, wideband audio, while only using 24-48 Kbps of bandwidth, depending on the data rate of the call Eliminates fatigue associated with straining to hear every word Near CD-quality sound for dynamic audio performance Superb clarity to hear the subtleties of every word Rated the best audio algorithm in the industry; outperforming MPEG4 in all tests Included with every VSX system For more detailed information, see www.polycom.com/siren14 Polycom StereoSurround 2 channels of 14 kHz crystal clear audio Superb clarity to hear the subtleties of every word Easily decipher multiple, simultaneous conversations Distinguish which side of the room people are talking from, just like in a real meeting SoundStation VTX 1000® Integration Use the SoundStation VTX 1000 as the microphone for the video system, as well as a standalone audio conferencing phone, and eliminates the VSX tabletop microphone array Enables POTS voice-only callers to be added to a video conference Maximize the value of audio and video equipment investments when used together Enhanced voice pickup range compared to other microphone solutions - you can additionally use the SoundStation VTX 1000 extension microphones for even greater pickup Video dial, mute, call hangup and call redial from the SoundStation VTX 1000 keypad Upgrade an audio call between SoundStation VTX 1000s when integrated with your VSX to video by simply pressing one button Supports StereoSurround VSX Microphone Array 360° audio pickup Padded feet filter out table-top noise Sensitive enough to pick up whispers, smart enough to eliminate unwanted background noise Versatile and attractive design for table top, wall or ceiling mount Only one VSX microphone array needed to support StereoSurround Gated enhanced stereo pickup Audio Error Concealment Reduced audio drop-out on noisy networks Smooth, consistent verbal communications without interruptions Corrects audio from any system seamlessly Activates automatically, only when needed * SoundStation VTX 1000 integration not available in all territories. Video H.264 Video Use less bandwidth to conduct video calls of equal quality when compared to calls using the H.263 standard Unparalleled video quality for lower line rate applications More bandwidth is available for other business Cost savings to any organization Supported at data rates from 64Kbps to 768Kbps Pro-Motional H.263 receive only Full-screen, TV-like video quality with fluid, precise motion handling Greater image detail with 50/60 fields per second Most natural video communications experience possible XGA Monitor Support Use a XGA display for main display or 2nd display Clearly see content in XGA resolution Flexibility in configuration, allowing users the display of their choice Supports up to 1024 x 768 resolution Support for 16:9 or 4:3 Format User selectable display format 16:9 aspect ratio with Dual Monitor Emulation is perfect for widescreen displays High resolution graphics displays for content presentation Option to zoom video to fit the 16:9 display Video Error Concealment Reduced video drop-out on busy IP networks and the Internet Smooth, continuous video without interruptions Maintain active face-to-face contact Accomplish meeting objectives without worrying about the video integrity Activates automatically, only when needed Content Sharing People+Content H.239 Polycom People+Content or standards based H.239 to insure interoperability Dual images allows the far end to see the content and the speaker at the same time Using Polycom's Visual Concert VSX, easily attach a laptop and display PC content (audio and video) during a video call Perfect for outside speakers/visitors to easily add content to a video conference People+Content IP Using a light utility on your PC, simply connect to your V500 and PC content is shown to all conference participants This feature can be used by anyone in the conference, even remote voice-only participants! Adjustable Bandwidth for Content Choose variable bandwidth allocated to People or Content Allows setting quality preference of a 90% / 10% split between People and Content or sharing bandwidth equally between People and Content (50% / 50%) User adjustable setting can be accessed both during a call and outside of a call User Interface Support for API Control Commands Custom control touch panel integration including AMX® or Crestron® is made easy with extensive support for API command set Integrator Reference Manual for VSX Series details over 200 commands for special applications Sample control templates are available for touch panel integration that provides basic functionality right out of the box Calendar & Call Scheduler Quick and easy access to a monthly calendar with current date and time posting Schedule individual or repeat meetings to dial automatically Create and save multipoint team meetings Customizable Home Page Display only the necessary buttons on the Home Page Unique design to meet your organizationâ€'s needs Simplified graphical interface for novice users Easy to understand icon driven menu navigation decreases learning curve Kiosk node allows you to create a look and feel that suits your application Leads to increased use of video Dual Monitor Emulation Use Dual Monitor Emulation for the most efficient use of a single display Makes effective use of 16:9 displays View near and far video windows at the same time Alternate layout views supporting near end, far end and content User Selectable Camera Icons Use icons that represent your industry Intuitive icons match the input device Custom name gives the icon clear meaning Selection of 5 libraries to choose from Decreases learning curve Display Configuration Choices Allows you to configure displays for your application Single or dual displays XGA or NTSC / PAL 4x3 or 16x9 formats Decide what shows on each display, near end, far end, content or VCR output Platform Versatility Integration with Microsoft Live Communications Server (LCS) via SIP Integrates directly with Microsoft collaboration infrastructure Registers and authenticates with Microsoft LCS 2005 VSX users can be added to Microsoft Communicator Presence information sent to LCS indicating video buddies' availability You can use your Buddy list to launch calls from the VSX 6000 user interface Multiple cameras supported Provides flexibility and choice Second camera can be easily added, and selected in the user interface Add document camera, VCR/DVD or any other second video source to bring another dimension to any video conference Secure FTP, Telnet and Web VSX systems are the most secure video conferencing solutions in the industry Now, all access to the VSX system can be secured including accessing the VSX system via web-browser as well as secure telnet and ftp access via SSH for secure system administration and application connectivity Advanced Encryption Standard (AES) Communications are confidential and secure Authentication by the National Institutes of Standards means it's credible Built-in, no extra hardware required H.460 NAT/Firewall Traversal Standards compliance for traversing NATs and Firewalls Session border controller (V2IU) is only needed at the central location and all VSX systems with H.460 can communicate through the single V2IU Uses H.460.18 for signaling traversal and call establishment and H.460.19 for media traversal IP (H.323, SIP or SCCP), Serial Connectivity H.323 and SIP are standard; choose the H.320 interface best for your application SCCP integration with Cisco Call Manager (in 128 MB systems) Choose the right option for your specific network connection Supports UPnP and NAT for automatic setup of conferences conducted through firewalls Wide range of IP QoS services to insure call quality and integrity Optional H.320 interface is a Quad BRI module Multiple Use RS-232 Port RS-232, DB9 port Transparent data pass-thru Peripheral options from Polycom Custom Products E-Mail dialing format when dialing across IP security boundaries Easy and intuitive E-mail video dialing Operates with Polycom's V2IU Firewall Traversal Appliance Deployable to wide range of customers, suppliers and partners Interoperability Part of Polycom Unified Collaborative Communications Remote management through Global Management System MGC Click & View offers a variety of layout templates for a multi-point call right from the systemâ€'s handheld remote Extended conferencing with Polycom PathNavigator Schedule, invite participants and manage conferences easily with Polycom Conference Suite Expand meeting capabilities by adding voice-only participants using the SoundStation VTX 1000 conference phone Use the Polycom SE200 for management, scheduling and gatekeeping Configurable MTU packet size Allows administrators to set the MTU packet size, based on needs of their network Optimizes packet size minimizing overhead and network congestion Default setting is 1260 Standards-based Solid, reliable platform Qualified by independent test labs Interoperates with other vendor's systems Easy to install even in a multi-vendor environment See technical specifications for complete list of standards FIPS 140-2 Certified Secure FTP, Telnet and Web VSX and V-series systems are the most secure video conferencing solutions in the industry Now, all access to the VSX system can be secured including accessing the VSX system via web-browser as well as secure telnet and ftp access via SSH for secure system administration and application connectivity Baseline Mode Set your system to baseline mode, H.261 and G.711, for maximum interoperability with legacy video conferencing systems Accessible via web, FTP or on-screen UI for easy access by the network administrator Can be turned on and off as needed 612 Polycom VSX 3000 Video Conferencing Device 2649 2649.00 Polycom Polycom IP Video Conferencing http://www.voipon.co.uk/polycom-vsx-3000-video-conferencing-device-p-612.html http://www.voipon.co.uk/images/polycom_vsx_3000.jpg new Availability: 2-3 days from order Polycom Video Conferencing Office Face-to-face calling solutions: From your office, from your home, or on the road. Polycom has the right video conferencing solutions to address the unique requirements of the office environment. VSX 3000: A full-featured, integrated video conferencing system for the executive office and remote offices. The Polycom VSX 3000 is a stylish and compact video conferencing system that provides excellent video and audio quality. Featuring an integrated LCD screen, the Polycom VSX 3000 is the perfect video conferencing solution for executive suites and remote offices. As a fully integrated video conferencing system, the Polycom VSX 3000 is easier to use in a shared environment, delivering ease of use with powerful video and audio performance. The Polycom VSX 3000 also serves as a PC display while not in a video call saving valuable desk space. Up to three other participants can be added with built-in multipoint conferencing bridge Easily share PC data during video conference with all participants LCD display that doubles as a PC monitor Offers exceptional video with H.264 video technology and near CD-quality audio with Polycom Siren 14 and Polycom StereoSurround. PC-based content can be added cable-free, by using the People+Content IP software option. Document camera or VCR/DVD can display pictures or movie files during a conference. Participants can see both the presenter and content simultaneously with Polycom Dual Monitor Emulation. Options: People+Content Share PC content cable free with the conference participants no need to connect your PC to the VSX system; shows people and data simultaneously MPPlus 4-way internal MCU including cascading and audio transcoding, mixed multipoint (IP and ISDN) Multiple network interfaces Purchase either IP only or IP/ISDN systems 613 Polycom V500 Video Conferencing Device 990.0000 1099.00 Polycom Polycom IP Video Conferencing http://www.voipon.co.uk/polycom-v500-video-conferencing-device-p-613.html http://www.voipon.co.uk/images/polycom_v500.jpg new Availability: 2-3 days from order Audio - Polycom V500 Features and Benefits G.722.1 Annex C 14 kHz crystal clear, wideband audio, while only using 24-48 Kbps of bandwidth, depending on the data rate of the call Eliminates fatigue associated with straining to hear every word Near CD-quality sound for dynamic audio performance Superb clarity to hear the subtleties of every word Rated the best audio algorithm in the industry; outperforming MPEG4 in all tests Included with every V-series and VSX system For more detailed information, see www.polycom.com/siren14 Audio Error Concealment Reduced audio drop-out on busy IP or ISDN networks Smooth, consistent verbal communications without interruptions Corrects audio from any system seamlessly Activates automatically, only when needed Video H.264 Video Use less bandwidth to conduct video calls of equal quality compared to calls using the H.263 standard Unparalleled video quality for lower line rate applications Remote office advantage since less bandwidth is needed Cost savings to any organization Supported at data rates from 64Kbps to 768Kbps Pro-Motional H.263 receive only Full-screen, TV-like video quality with fluid, precise motion handling Greater image detail with 50/60 fields per second Most natural video communications experience possible Video Error Concealment Reduced video drop-out on busy IP networks and the Internet Smooth, continuous video without interruptions Maintain active face-to-face contact Accomplish meeting objectives without worrying about the video integrity Activates automatically, only when needed; drops out when not needed Content Sharing People+Contentent H.239 Polycom People+Content or standards based H.239 to insure interoperability Dual images allows the far end to see the content and the speaker at the same time People+Content IP Using a light utility on your PC, simply connect to your V500 and PC content is shown to all conference participants This feature can be used by anyone in the conference, even remote voice-only participants! Adjustable Bandwidth for Content Choose variable bandwidth allocated to People or Content Allows setting quality preference of a 90% / 10% split between People and Content or sharing bandwidth equally between People and Content (50% / 50%) User adjustable setting can be accessed both during a call and outside of a call User Interface Calendar & Call Scheduler Quick and easy access to a monthly calendar with current date and time posting Schedule individual or repeat meetings to dial automatically Create and save multipoint team meetings Customizable Home Page Display only the necessary buttons on the Home Page Unique design to meet your organization's needs Simplified graphical interface for novice users Easy to understand icon driven menu navigation decreases learning curve Kiosk Mode allows you to create a look and feel that suits your application Leads to increased use of video Dual Monitor Emulation Use Dual Monitor Emulation for the most efficient use of a single display Makes optimum use of a 16:9 display View near and far video windows at the same time Alternate layout views supporting near end, far end and content User Selectable Camera Icons Use icons that represent your industry Intuitive icons match the input device Custom name gives the icon clear meaning Selection of 5 libraries to choose from Decreases learning curve Platform Versatility Integration with Microsoft Live Communications Server (LCS) via SIP Integrates directly with Microsoft collaboration infrastructur Registers and authenticates with Microsoft LCS 2005 VSX and V-series users can be added to Microsoft Communicator Presence information sent to LCS indicating video buddies' availability You can use your Buddy list to launch calls from the V500 user interface Simple Plug and Play Installation Integrated microphone and camera Sleek, set top format to use on top of TV or flat panel display Headphone jack for privacy Personalized graphical user interface makes it easy to use Install wizard guides available for easy installation Use TV grade displays (NTSC/PAL) Use TV speakers for external audio IP (H.323 or SIP) or ISDN/IP Models Choose the right configuration for your specific network connection Supports UPnP and NAT for automatic setup of conferences conducted through firewalls Wide range of IP QoS services to insure call quality and integrity Advanced Encryption Standard (AES) Communications are confidential and secure Authentication by the National Institutes of Standards means it's credible Built-in, no extra hardware required H.460 NAT/Firewall Traversal Standards compliance for traversing NATs and Firewalls Session border controller (V2IU) is only needed at the central location and all VSX and V-series systems with H.460 can communicate through the single V2IU Uses H.460.18 for signaling traversal and call establishment and H.460.19 for media traversal E-Mail dialing format when dialing across IP security boundaries Easy and intuitive E-mail video dialing Operates with Polycom's V2IU Firewall Traversal Appliance Deployable to wide range of customers, suppliers and partners Interoperability Part of Polycom Unified Collaborative Communications Remote management through Global Management System MGC Click & View offers a variety of layout templates for a multi-point call right from the system's handheld remote Extended conferencing with Polycom PathNavigator Schedule, invite participants and manage conferences easily with Polycom Conference Suite Use the Polycom SE200 for management, scheduling and gatekeeping Configurable MTU packet size Allows administrators to set the MTU packet size, based on needs of their network Optimizes packet size minimizing overhead and network congestion Default setting is 1260 Standards-based Solid, reliable platform Qualified by independent test labs Interoperates with other vendor's systems Easy to install even in a multi-vendor environment See technical specifications for full list of standards Baseline Mode Set your system to baseline mode, H.261 and G.711, for maximum interoperability with legacy video conferencing systems Accessible via web. FTP or on-screen UI for easy access by the network administrator Can be turned on and off as needed FIPS 140-2 Certified Provides ultimate security including Secure Telnet, Secure TLS in Web, Telnet, FTP, Secure Web (HTTPS), Security Mode Optimal for networks that demand FIPS security validation on devices 614 IC-Talk Media Manager MM50 1329.05 1329.05 IC Talk IC Talk IP PBX http://www.voipon.co.uk/ictalk-media-manager-mm50-p-614.html http://www.voipon.co.uk/images/ic-talk_media_manager_mm550.jpg new 50 User Solid State 1U high 19" rack mount IP PBX... The MM50 Range is tailored for medium office / large office and is the latest in the product line. The MM50 is a 1U high 19" rack mountable system which is solid state (no moving parts) and hence is silent in operation. The device is cool running and also shallow depth for convenience. Utilising the latest in solid state flash memory the MM50 can store up to 8 hours of voicemail recordings and 6 months of call details (CDR). Enterprise class features and up to 50 users make this small format system a real power house with unparalleled capabilities and functions to help make your business perform better, respond quicker and sound more professional. An extensive list of functions and features coupled with a custom programming interface mean that these systems can be highly integrated into your organisation whilst adhering to the industry standards ensuring you get the best choice of hardware available. Features include:- 50 x Extensions 50 x Ring Groups 8 Hours of Recording time for voicemails and greetings 6 Months of Call Detail Logging (export to CSV available) 4 x Call Queues 4 x Multiway Conference Rooms (pin code protected) Choice of PSTN Interface (IPOnly, 4 x ISDN2e or ISDN30e) Unlimited SIP trunks and loads more... See the full feature list on the bottom right... Packs all the punch of a larger business phone system, without the cost! Versions Available IP Only Quad ISDN2e - Basic Rate Euro ISDN (Euro ISDN (300 102) signalling, TE & NT mode operation) Single ISDN30 - Primary Rate Euro ISDN (PRI-NET & PRI-CPE, Euro ISDN, NI1&2, 4ESS (AT&T), 5ESS (Lucent) & DMS100) Hardware Specification Power: 230V AC 50Hz. External Power Adaptor max. consumption 80W (average running consumption 12-25W). Interfaces: Telephony Slot (varies dependant upon model), 10/100Mb/s Ethernet RJ45 for LAN connection. USB2.0 for Flash Backup Pen. Protocols: TCP/IP v4 (v6 ready), SIP (Session Initiation Protocol), H.323, IAX2, HTTP1.1 (for administration). Compression Codecs: ADPCM, G.711 (A-Law & &#956;-Law), G.726, G.729 (optional extra), GSM, iLBC, Linear, LPC-10 & Speex. 615 IC-Talk Media Manager MM150r 2279.05 2279.05 IC Talk IC Talk IP PBX http://www.voipon.co.uk/ictalk-media-manager-mm150r-p-615.html http://www.voipon.co.uk/images/ic-talk_media_manager_mm150r2.jpg new 50 User Solid State 1U high 19" rack mount IP PBX... The MM50 Range is tailored for medium office / large office and is the latest in the product line. The MM50 is a 1U high 19" rack mountable system which is solid state (no moving parts) and hence is silent in operation. The device is cool running and also shallow depth for convenience. Utilising the latest in solid state flash memory the MM50 can store up to 8 hours of voicemail recordings and 6 months of call details (CDR). Enterprise class features and up to 50 users make this small format system a real power house with unparalleled capabilities and functions to help make your business perform better, respond quicker and sound more professional. An extensive list of functions and features coupled with a custom programming interface mean that these systems can be highly integrated into your organisation whilst adhering to the industry standards ensuring you get the best choice of hardware available. Features include:- 50 x Extensions 50 x Ring Groups 8 Hours of Recording time for voicemails and greetings 6 Months of Call Detail Logging (export to CSV available) 4 x Call Queues 4 x Multiway Conference Rooms (pin code protected) Choice of PSTN Interface (IPOnly, 4 x ISDN2e or ISDN30e) Unlimited SIP trunks and loads more... See the full feature list on the bottom right... Packs all the punch of a larger business phone system, without the cost! Versions Available IP Only Quad ISDN2e - Basic Rate Euro ISDN (Euro ISDN (300 102) signalling, TE & NT mode operation) Single ISDN30 - Primary Rate Euro ISDN (PRI-NET & PRI-CPE, Euro ISDN, NI1&2, 4ESS (AT&T), 5ESS (Lucent) & DMS100) Hardware Specification Power: 230V AC 50Hz. External Power Adaptor max. consumption 80W (average running consumption 12-25W). Interfaces: Telephony Slot (varies dependant upon model), 10/100Mb/s Ethernet RJ45 for LAN connection. USB2.0 for Flash Backup Pen. Protocols: TCP/IP v4 (v6 ready), SIP (Session Initiation Protocol), H.323, IAX2, HTTP1.1 (for administration). Compression Codecs: ADPCM, G.711 (A-Law & &#956;-Law), G.726, G.729 (optional extra), GSM, iLBC, Linear, LPC-10 & Speex. 616 Redfone foneBRIDGE2-EC Quad T1/E1 Ethernet Bridge with EC 1303.0000 1371.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2ec-quad-t1e1-ethernet-bridge-with-ec-p-616.html http://www.voipon.co.uk/images/redfone_fonebridge2_e1_t1.jpg new Availability: In stock Asterisk? T1/E1 Redundancy, High Availability and now Carrier Class Echo Cancellation in an economical package. foneBRIDGE2-EC is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP/PBX systems. foneBRIDGE2-EC eliminates the need to install proprietary TDM hardware cards in approved compatible server configurations. Instead, foneBRIDGE2-EC terminates T1/E1 PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities. Engineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, foneBRIDGE2 provides low-latency delivery of your critical voice traffic. Failover & Asterisk HA Enabled - High-Availability through rapid failover capabilities ensures your critical telephony services are always on and available Flexible Configuration - Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment Solid State Embedded Appliance - No moving components - Low heat and power consumption - High MTBF (mean time before failure) Carrier Class Echo Cancellation - ITU G.168 Compliant - 128ms of echo tail on over 100 concurrent channels - Qualified as Toll Quality by AT&T KEY FEATURES Target Applications - T1/E1 PRI Trunk termination - Legacy PBX-to-Asterisk integration - High-Availability/Failover Asterisk clusters - Channel Bank connectivity - Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) - Blade Servers where PCI slots are not available Standards Standards Telephony - PRI Switch Compatibility- EuroISDN, AT&T 4ESS, DMS 100, Lucent 5E, NI1/ NI2; Network or CPE - Line Interface- Dual or Quad T1/E1 (RJ45), per port configurable - Line Encoding- AMI/B8ZS for T1, AMI/HDB3 for E1 - Super Frame (SF) and Extended Super Frame (ESF) - Robbed Bit Signaling (RBS/CAS) - Short-haul and Long Haul line build out (LBO) - Adaptive Equalizer for line attenuation conditioning Echo Cancellation - Utilizes on board high-performance digital signal processing with TI's 6000? DSP platform - 128ms (1024 taps) of echo tail - Comfort Noise Generator - Automatic tail search, fast convergence, and no divergence due to double talk - Tone disabler for clear fax and modem communication - Powerful and flexible FPGA based architecture provides for quick field upgradeability Ethernet - 10/100-BASE-TX Half/Full-duplex - 2 x Dedicated RJ45 Specifications - Electrical: DC 500mA Max @ 5V (2.5W) - Environmental: 0 to 50 deg C operating - Physical: Dimensions: 5.00" x 6.50" x 1.00" Weight: 1 pound Mounting: Flange Mount or Desktop 617 Redfone foneBRIDGE2-EC Dual T1/E1 Ethernet Bridge with EC 923.0000 971.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2ec-dual-t1e1-ethernet-bridge-with-ec-p-617.html http://www.voipon.co.uk/images/redfone_fonebridge2_e1_t1.jpg new Availability: In stock Asterisk? T1/E1 Redundancy, High Availability and now Carrier Class Echo Cancellation in an economical package. foneBRIDGE2-EC is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP/PBX systems. foneBRIDGE2-EC eliminates the need to install proprietary TDM hardware cards in approved compatible server configurations. Instead, foneBRIDGE2-EC terminates T1/E1 PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using native Asterisk TDMoE formats and utilities. Engineered around our unique high-speed SoC (System-on-Chip) TDMoE engine, fone-BRIDGE2 provides low-latency delivery of your critical voice traffic. Failover & Asterisk HA Enabled - High-Availability through rapid failover capabilities ensures your critical telephony services are always on and available Flexible Configuration - Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment Solid State Embedded Appliance - No moving components - Low heat and power consumption - High MTBF (mean time before failure) Carrier Class Echo Cancellation - ITU G.168 Compliant - 128ms of echo tail on over 100 concurrent channels - Qualified as Toll Quality by AT&T KEY FEATURES Target Applications - T1/E1 PRI Trunk termination - Legacy PBX-to-Asterisk integration - High-Availability/Failover Asterisk clusters - Channel Bank connectivity - Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) - Blade Servers where PCI slots are not available Standards Standards Telephony - PRI Switch Compatibility- EuroISDN, AT&T 4ESS, DMS 100, Lucent 5E, NI1/ NI2; Network or CPE - Line Interface- Dual or Quad T1/E1 (RJ45), per port configurable - Line Encoding- AMI/B8ZS for T1, AMI/HDB3 for E1 - Super Frame (SF) and Extended Super Frame (ESF) - Robbed Bit Signaling (RBS/CAS) - Short-haul and Long Haul line build out (LBO) - Adaptive Equalizer for line attenuation conditioning Echo Cancellation - Utilizes on board high-performance digital signal processing with TI's 6000? DSP platform - 128ms (1024 taps) of echo tail - Comfort Noise Generator - Automatic tail search, fast convergence, and no divergence due to double talk - Tone disabler for clear fax and modem communication - Powerful and flexible FPGA based architecture provides for quick field upgradeability Ethernet - 10/100-BASE-TX Half/Full-duplex - 2 x Dedicated RJ45 Specifications - Electrical: DC 500mA Max @ 5V (2.5W) - Environmental: 0 to 50 deg C operating - Physical: Dimensions: 5.00" x 6.50" x 1.00" Weight: 1 pound Mounting: Flange Mount or Desktop 618 Grandstream GXP1200 2-line IP Telephone 48.0000 51.50 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp1200-2line-ip-telephone-p-618.html http://www.voipon.co.uk/images/grandstream_gxp_1200.jpg new Availability: In stock The GXP1200 is a very affordable 2-line entry-level SIP phone delivering excellent sound quality and advanced features. The Grandstream GXP1200 supports traditional telephony features such as caller ID, call-waiting, park, mute, hold, transfer and voicemail as well as advanced enterprise features such as 2 independent SIP accounts, multi-language support, 3-way conferencing and custom ring-tones for distinctive call handling. The GXP1200 also supports a wide range of voice codecs and offers dual 10/100mbps auto-sensing Ethernet ports with integrated Power over Ethernet, headset jack and secure central configuration file for mass deployment. The GXP1200 is quick and easy to install and manage, thanks to GUI web interfaces. Broad interoperability with major SIP based platforms ensures compatibility with any IP telephony system. Its easy installation, hands-free duplex speakerphone, backlit graphic LCD display, 8 dedicated function keys and XML applications make the GXP1200 a very cost-effective choice for any business that needs a feature rich two line IP phone. Technical Features SIP 2.0, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (A record and SRV), DHCP (both client and server), PPPoE, TFTP, NTP, Telnet Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af) 128x32 pixel graphic LCD with backlight Call Features 2 lines appearances with individual SIP account profiles and dual-color LED Call waiting, hold, mute, transfer (blind or attended), forward, and more Support Caller ID display or block 3-way conferencing 5 navigation/menu keys 8 dedicated buttons - hold, mute, speakerphone, send, transfer, conference, headset, messages Handset Features Full duplex speakerphone w/advanced audio echo cancellation Secure and automated provisioning for mass deployment Headset jack (2.5 mm ) for broad headset compatibility SRTP and TLS (pending) for privacy protection And many more enterprise grade features Feature Specifications SIP Compliant and Protocols: SIP, TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP, etc. Networking Interfaces: Dual 10/100mbps Ethernet ports (switched or routed). Voice Codecs: G.711 (a/u-law), G.723.1, G.729A/B, G.726, GSM, iLBC, and G.722 (wideband). Superb Audio: Quality Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG, AGC high fidelity wideband audio (G.722). Custom Ringtone: Software Convert most music files to a Grandstream ringtone. Click here to download. Advanced Functionality: 2-line support, 3-way conferencing, multi-language support (MLS), headset enabled, AES encryption, etc. 619 Grandstream GXP2010 4-line Key System IP Telephone 72.8500 77.50 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp2010-4line-key-system-ip-telephone-p-619.html http://www.voipon.co.uk/images/grandstream_gxp_2010.jpg new Availability: In Stock The new GXP2010 Key System IP phone has a new sleek design and delivers excellent call quality and enterprise grade feature set that includes advanced XML capabilities, multi-party conferencing, multi-language support, presence and BLF (busy lamp field), security protection, automated provisioning, and broad compatibility with leading SIP platforms. The GXP2010 offers 4 lines, 18 programmable keys, 3 dynamic context-sensitive XML soft keys, dual switched 10M/100Mbps auto-sensing Ethernet ports with integrated PoE, and a large high-resolution backlit LCD display. As with all models in the GXP Series, this model supports comprehensive voice codecs and SRTP for privacy protection. The GXP2010 was designed to be the cost-effective choice for the busy call center or small business that wants the feature functionality of a key-system. The 18 programmable speed-dial keys enable one-button access to office personnel and the 11 dedicated keys create one-button access to indispensable telephony features including hold, conference, mute, do-no-disturb, intercom, transfer, speaker and voicemail. This phone is ideal for enhancing office productivity and improving efficiency. Technical Features & Benefits Support SIP (RFC3261), TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP Dual switched 10/100Mbps Ethernet ports w/integrated Power-over-Ethernet (802.3af) Support NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more Backlit 240x120 high resolution graphic LCD with multi-level gray scales Call Features & Benefits 4 lines indicators with individual SIP account profiles 18 programmable hard keys and 3 XML programmable context-sensitive soft keys Multi-line support of up to 22 call appearance lines with dual-color LED indicators Support Caller ID display or block, per call or permanent Call waiting, hold, mute, transfer (blind or attended), forward, and more Multi-party conferencing (up to 4-way) and shared line appearances Expandable through expansion key-modules (pending) And many more enterprise grade features Handset Features & Benefits Full duplex speakerphone w/advanced audio echo cancellation Headset jack (2.5 mm and RJ11) for broad headset compatibility Secure and automated provisioning for mass deployment SRTP and TLS (pending) for privacy protection And many more enterprise grade features Feature Specifications SIP Compliant and Protocols: SIP, TCP/IP/UDP, RTP/RTCP, HTTP, ARP, ICMP, DNS (A record and SRV), DHCP(both client and server), PPPoE, TFTP, NTP, etc. Networking Interfaces: Dual 10/100mbps Ethernet ports (switched or routed). Voice Codecs: G.711 (a/?-law), G.723.1, G.729A/B, G.726, GSM, iLBC, and G.722 (wideband). Superb Audio Quality: Advanced Digital Signal Processing (DSP), Silence suppression, VAD, CNG, AGC high fidelity wideband audio (G.722). Custom Ringtone: Software Convert most music files to a Grandstream ringtone. Click here to download. Advanced Functionality: Multi-line support, multi-party conferencing (4-way), multi-language support (MLS), headset enabled, expandable, intercom, AES encryption, etc. 620 Grandstream Handytone 503 Analog Adaptor 52.0000 54.00 Grandstream Grandstream Adaptors http://www.voipon.co.uk/grandstream-handytone-503-analog-adaptor-p-620.html http://www.voipon.co.uk/images/grandstream_handytone_ht503.jpg new Availability: In stock The HandyTone 503 is the next generation of powerful, affordable, high quality and manageable IP telephony ATA/IAD for residential users and road-warriors. Its compact size, excellent voice quality, packed feature functionality and best-in-class price-performance point enable consumers to maximize the power of IP voice and data connectivity. The HandyTone 503 is based on SIP standard and interoperable with most 3rd party SIP compliant devices and software. It features 1 FXS telephone port, 1 FXO PSTN line port, dual 10M/100Mbps network ports, port status and message waiting LED, and a base stand for vertical positioning. Grandstream HandyTone HT-503 Analog Telephone Adapter with 1 FXS Telephone Port & 1 FXO PSTN Line Port IP connectivity for any phone and fax Hop-on/Hop-off calling Web management for easy configuration and installation Offers traditional and advanced telephony features Portable and compact for use at home or on the road Grandstream HandyTone HT-503 Features 1 FXS telephone port (RJ11), 1 FXO PSTN line port (RJ11) with power-outage life line support Up to 2 SIP account profiles, SIP over TCP/TLS, SRTP Dual 10/100 Mbps network ports (RJ45) with integrated high performance NAT router Status LED for power, telephone, PSTN line, network, and message waiting indication Advanced telephony features o Caller ID from both IP and PSTN o Call waiting, 3-way conference with IP and/or PSTN o Remote call origination and termination from/to PSTN o Hop-on and hop-off calling o Transfer to OR forward to IP or PSTN o Do not disturb o Message waiting indication o Multi-language voice prompts o Flexible dial plan, direct IP calling Comprehensive voice codecs Secure and automated provisioning using HTTP/HTTPS/Telnet/TFTP Symmetric and asymmetric voice codec/RTP in any call sessions T.38 Fax SIP over TCP/TLS   621 Snom M3 Dect VoIP Phone / IP Telephone 104.0000 113.00 Snom Snom DECT SIP Phone http://www.voipon.co.uk/snom-m3-dect-voip-phone-ip-telephone-p-621.html http://www.voipon.co.uk/images/snom_m3.jpg new Availability: In stock Sleek & Modern - the Snom M3 Cordless IP Telephone The m3 is snom ´s first IP DECT phone product. The M3 provides the latest in style, function and mobility without any compromise in audio quality. The snom m3 is the perfect phone for demanding home office, small to medium business and enterprise users. It is an expandable system with multiline functionality for up to eight handsets, handling up to three simultaneous calls at a time. The m3 has a 128 x 128 px colour display with backlight and has operating times of 10 hours talk or 100 hours standby. There are many additional features including three-way conference calling, call transferring, holf music, call forwarding and speed dialing with an address book (up to 100 entries). The m3 provides maximum freedom and ease-of-use. The snom phone has an indoor range of 50 metres with an outdoor range of 100 metres. The snom m3 menu is easily configurable to meet each individual user´s needs – with the option to configure on-site or remotely. Innovative features are both inside and out on the m3. Snom worked with a renowned designer, known for his work in luxury car phone installations (such as Maybach, Porsche and BMW) to create the snom m3\'s sleek, modern and desirable look. Key Features/Specification Display: 128 x 128 pixels, 65536 colors, backlit Li-Ion battery pack for 10 hours of calls or 100 hours standby Range: 50 meters indoors, 100 meters outdoors 12 numerical keys, 5 navigation keys, 2 function keys 8 handsets per base station 8 SIP registrations with different servers/registrars Up to 3 concurrent calls per base station Three-way conference Remote setup, password protection Open DECT GAP standard Speakerphone on mobile handset Polyphonic ringtones Automatic registration of handset Separate charging cradle for handset 622 Snom M3 Handset 60.5000 63.00 Snom Snom DECT SIP Phone http://www.voipon.co.uk/snom-m3-handset-p-622.html http://www.voipon.co.uk/images/snom_m3_handset.jpg new Availability: In stock Individual handset for use with the Snom M3 IP Telephone . Up to 8 handsets can be used with a snom m3 basestation . The snom m3 comes with a 128 x 128 Pixel-Display with colour display and backlight function along with 10 hours talk time/ 100 hours standby. The coverage has an indoor range of 50m and an outdoor range of 100m. 623 Elmeg / Funkwerk IP50 IP Telephone 53.9000 53.90 Elmeg Elmeg IP Telephones http://www.voipon.co.uk/elmeg-funkwerk-ip50-ip-telephone-p-623.html http://www.voipon.co.uk/images/funkwerk_ip50.jpg new Availability: In stock With the VoIP introductory model funkwerk IP50 you can conveniently make your phone calls over the Internet. The compliance with the worldwide SIP standard ensures investment protection and guaranteed applicability in the future. funkwerk IP50 fulfills all requirements in this context: simple operation, up to 3 SIP accounts, an internal phone book with 140 entries, call hold, call switching, call forwarding, and lists of accepted and missed calls leave almost no wish unfulfilled. The most important functions can be operated conveniently via 10 function keys. The settings at the phone can be made in a menu-instructed mode - and conveniently via a web browser. The RFC 3261-compatible SIP stack ensures compatibility with almost all adapters and VoIP routers currently available on the market. Two Ethernet ports allow the direct connection of a PC to funkwerk IP50. Features / Technical Data funkwerk IP50 Key Features Overview: SIP protocol 3 SIP accounts Hands free operation 3 party conference Do not disturb function Configuration via keypad and / or WEB browser Connections: 2x RJ45 Ethernet port (1x network / 1x PC) Handset (RJ plug) 1 x WAN interface 1 x LAN interface Functions Alphanumerical dotmatrix display with 2 rows / 16 characters each Display language: GB 10 function keys (fixed) 1 IP Info 2 Call forwarding, 3 do not disturb, 4 - 6 Memory (No. 1 ? 3 out of the short dial list) 7- 9 line keys (No. 1-3 of the SIP accounts), 10 mute 1 function keys (fixed) Hands free operation 5 soft keys: 1 incoming callers list (11 entries), 2 outgoing callers list (11 entries), 3 Phonebook, 4 Menu, 5 O.K buttom 4 operation keys: 1 Conference, 2 Transfer, 3 Hold, 4 Redial (1 entry) Features: 3 SIP accounts hands free operation Phone book memory with 140 entries 3 Speed dial keys Redial memory (sep. incoming ? outgoing calls) Callers list Call preparation Mute Call time display CLIP Call Hold, Call Waiting and Call transfer Conference (3PTY) DND (Do not disturb) Call forwarding (CFU, CFB, CFNR) Configuration via: Telnet, Browser, Keypad NAT traversal: STUN QoS ToS field In-Band DTMF, RFC-2833, SIP Info Firmware upgrade by TFTP, HTTP, Telnet MD5 for SIP authentification (RFC 2069 / RFC 2617) Hardware Overview: Main Chip ? CM5000 Data Memory ? 16MBit Program Memory ? 64MBit Ethernet Jack ? 2 x 10/100MBit jacks AC/DC adapter ? Input AC230V, output 9V Sound Quality VAD (Voice Activity Detection) CNG (Comfortable Noise Generator) LEC (Line Echo Canceller) Packet Loss Compensation Adaptiv Jitter Buffer Network Protocol SIP v1 (RFC2543), v2(RFC3261) IP / TCP / UDP / RTP / RTCP IP/ ICMP / ARP / RARP / SNTP TFTP Client / DHCP Client / PPPoE Client / Static IP Telnet / HTTP Server DNS Client Audio Codecs: G.711: 64k bit/s (PCM) G.723.1: 6.3k / 5.3k bit/s G.726: 16k / 24k / 32k / 40k bit/s (ADPCM) G.729A: 8k bit/s (CS-ACELP) G.729B: adds VAD & CNG to G.729 Security HTTP 1.1 basic/digest authentication for Web setup MD5 for SIP authentication (RFC2069/ RFC 2617) Electric requirements Voltage: 9V DC Power adaptor: AC/DC input 230V, output 9V Network interface: 2 X RJ-45 Ethernet connectors Dimensions: 198 x 176 x 60 mm (L x W x H Weight: Phone : 839 g. with Adapor. Total weight with giftbox : 1056 g. Operation Temperature:] 5 to 55 C? Relative Humidity Up to 90% Funkwerk Enterprise Communications GmbH - Sue 625 Linksys WBP54G 802.11G WIFI Dongle 25 25.00 Linksys VoIP Accessories http://www.voipon.co.uk/linksys-wbp54g-80211g-wifi-dongle-p-625.html http://www.voipon.co.uk/images/linksys_wbp54g_802.11g_wifi_dongle.jpg new Availability: In Stock The Wireless-G Bridge for Phone Adapters was specially designed to convert your Phone Adapter into a wireless device, so it can connect to your home network without an Ethernet cable. This lets you put your phone where it's most convenient for you, and not be constrained to the area around your Internet connection. The included Setup Wizard makes it easy to configure the Bridge to your wireless network's settings. To protect your privacy, all wireless voice transmissions can be encrypted with WEP or industrial-strength Wi-Fi Protected Access (WPA/WPA2) security. 626 Polycom SoundStation IP4000 Extended Microphones 137.3300 169.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundstation-ip4000-extended-microphones-p-626.html http://www.voipon.co.uk/images/polycom_ip4000_mics-sml.jpg new Availability: In stock Extend Your Polycom Soundsation IP 4000 Quickly & Easily Optional extended microphones expand coverage to 20 x 30 feet for larger conference rooms. Clearer, more natural conferencing and higher productivity Each accessory kit includes two Extended Microphones for optimal coverage.   627 Rhino R1T1 Single T1/E1/PRI PCI Card (no EC) 220.0000 275.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r1t1-single-t1e1pri-pci-card-no-ec-p-627.html http://www.voipon.co.uk/images/rhino_r1t1.gif new Availability: In Stock Linux Open Source Telephony Single T1 PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, uad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. T1/E1 PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interface chip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket All major signaling modes supported (E&M, PRI, Loop, Ground, Kewl, etc.) Loop start signaling for advanced features such as Caller ID and Distinctive Ring 5-year limited warranty CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDNPRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI or HDB3 Line buildouts selections: 0-133 feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655 feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 2.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 628 Rhino R1T1-EC Single T1/E1/PRI PCI Card, with EC 368.0000 460.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r1t1ec-single-t1e1pri-pci-card-with-ec-p-628.html http://www.voipon.co.uk/images/rhino_r1t1.gif new Availability: In Stock Linux Open Source Telephony Single T1 PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, uad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. T1/E1 PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interface chip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket All major signaling modes supported (E&M, PRI, Loop, Ground, Kewl, etc.) Loop start signaling for advanced features such as Caller ID and Distinctive Ring 5-year limited warranty CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDNPRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI or HDB3 Line buildouts selections: 0-133 feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655 feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 2.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 630 Rhino R1T1e-EC Single T1/E1/PRI PCI Card, PCI Express, with EC 368.0000 460.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r1t1eec-single-t1e1pri-pci-card-pci-express-with-ec-p-630.html http://www.voipon.co.uk/images/rhino_r1t1.gif new Availability: In stock Linux Open Source Telephony Single T1 PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, uad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. T1/E1 PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interface chip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket All major signaling modes supported (E&M, PRI, Loop, Ground, Kewl, etc.) Loop start signaling for advanced features such as Caller ID and Distinctive Ring 5-year limited warranty CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDNPRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI or HDB3 Line buildouts selections: 0-133 feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655 feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 2.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 631 Rhino R2T1 Dual T1/E1/PRI PCI Card (no EC) 367.0000 459.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r2t1-dual-t1e1pri-pci-card-no-ec-p-631.html http://www.voipon.co.uk/images/rhino_r2t1.gif new Availability: In Stock Linux Open Source Telephony Dual T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Dual T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code Quad T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interfacechip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket for each individual port All major signaling modes supported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such as Caller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable usingZaptel tools (i.e. zttool) Mechanical Data Size: 3.0? tall, 5.20? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 632 Rhino R2T1-EC Dual T1/E1/PRI PCI Card, with EC 620.0000 775.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r2t1ec-dual-t1e1pri-pci-card-with-ec-p-632.html http://www.voipon.co.uk/images/rhino_r2t1.gif new Availability: In Stock Linux Open Source Telephony Dual T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Dual T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code Quad T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interfacechip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket for each individual port All major signaling modes supported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such as Caller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable usingZaptel tools (i.e. zttool) Mechanical Data Size: 3.0? tall, 5.20? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 633 Rhino R2T1-e-EC Dual T1/E1/PRI PCI Card, PCI Express, with EC 620.0000 775.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r2t1eec-dual-t1e1pri-pci-card-pci-express-with-ec-p-633.html http://www.voipon.co.uk/images/rhino_r2t1.gif new Availability: In Stock Linux Open Source Telephony Dual T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Dual T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code Quad T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interfacechip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket for each individual port All major signaling modes supported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such as Caller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable usingZaptel tools (i.e. zttool) Mechanical Data Size: 3.0? tall, 5.20? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 634 Rhino R4T1-EC Quad T1/E1/PRI PCI Card, with EC 906.0000 1132.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r4t1ec-quad-t1e1pri-pci-card-with-ec-p-634.html http://www.voipon.co.uk/images/rhino_r4t1.gif new Availability: In Stock Linux Open Source Telephony Quad T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asteriskand Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, forlarge scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phonesand wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IPwhen you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can beconfident that Rhino will always work hard in your Open Source Telephony application. Quad T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested andready Zaptel-compliant open sourceLinux module source code Quad T1/E1 embedded CSU Line buildout software selec-table Custom Rhino PCI interfacechip means no excess CPUoverhead Fractional voice and datacapable Field software upgradable T1 crossover cables included Alarm and Link status LEDsvisible from the rear bracket foreach individual port All major signaling modes sup-ported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such asCaller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer,that is software controlled andsoftware programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiverfunctionality Long-haul and short-haul lineinterface for clock/datarecovery and waveshaping Crystal-less jitter attenuator Fully independent transmit andreceive functionality Single chip line interface unit(LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 3.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 635 Rhino R4T1-e Quad T1/E1/PRI PCI Card, PCI Express 653.0000 817.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r4t1e-quad-t1e1pri-pci-card-pci-express-p-635.html http://www.voipon.co.uk/images/rhino_r4t1.gif new Availability: In Stock Linux Open Source Telephony Quad T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asteriskand Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, forlarge scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phonesand wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IPwhen you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can beconfident that Rhino will always work hard in your Open Source Telephony application. Quad T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested andready Zaptel-compliant open sourceLinux module source code Quad T1/E1 embedded CSU Line buildout software selec-table Custom Rhino PCI interfacechip means no excess CPUoverhead Fractional voice and datacapable Field software upgradable T1 crossover cables included Alarm and Link status LEDsvisible from the rear bracket foreach individual port All major signaling modes sup-ported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such asCaller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer,that is software controlled andsoftware programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiverfunctionality Long-haul and short-haul lineinterface for clock/datarecovery and waveshaping Crystal-less jitter attenuator Fully independent transmit andreceive functionality Single chip line interface unit(LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 3.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 636 Rhino R4T1-e-EC Quad T1/E1/PRI PCI Card, PCI Express, with EC 906.0000 1132.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r4t1eec-quad-t1e1pri-pci-card-pci-express-with-ec-p-636.html http://www.voipon.co.uk/images/rhino_r4t1.gif new Availability: In Stock Linux Open Source Telephony Quad T1 PCI Plug-In Card Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asteriskand Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, forlarge scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phonesand wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IPwhen you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to giveyou the support you need, when you need it. Our 5-year, limited warranty means that you can beconfident that Rhino will always work hard in your Open Source Telephony application. Quad T1/E1 PCIPlug-In Card Specifications PCI Card Features Asterisk soft PBX tested andready Zaptel-compliant open sourceLinux module source code Quad T1/E1 embedded CSU Line buildout software selec-table Custom Rhino PCI interfacechip means no excess CPUoverhead Fractional voice and datacapable Field software upgradable T1 crossover cables included Alarm and Link status LEDsvisible from the rear bracket foreach individual port All major signaling modes sup-ported (E&M, PRI, Loop,Ground, Kewl, etc.) Loopstart signaling foradvanced features such asCaller ID and Distinctive Ring CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer,that is software controlled andsoftware programmable - no jumpers Complete T1/DS1/ISDN-PRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiverfunctionality Long-haul and short-haul lineinterface for clock/datarecovery and waveshaping Crystal-less jitter attenuator Fully independent transmit andreceive functionality Single chip line interface unit(LIU) and Framer Zaptel Selections Per channel programmability T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI orHDB3 Line buildouts selections: 0-133feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 3.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1.5 pounds with all included components maximum 637 Rhino R1T1-e Single T1/E1/PRI PCI Card, PCI Express (no EC) 220.0000 275.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-r1t1e-single-t1e1pri-pci-card-pci-express-no-ec-p-637.html http://www.voipon.co.uk/images/rhino_r1t1.gif new Availability: This product has been discontinued.  Please conisder the Rhino R1T1e-EC PCI Express Linux Open Source Telephony Single T1 PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asteriskand Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, uad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? With Rhino you can use lower cost analog phones with digital features and get guaranteed T1 voice quality, all while enjoying Asterisk VoIP technology for off-premise connectivity. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. T1/E1 PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code T1/E1 embedded CSU Line buildout software selectable Custom Rhino PCI interface chip means no excess CPU overhead Fractional voice and data capable Field software upgradable T1 crossover cables included Alarm and Link status LEDs visible from the rear bracket All major signaling modes supported (E&M, PRI, Loop, Ground, Kewl, etc.) Loop start signaling for advanced features such as Caller ID and Distinctive Ring 5-year limited warranty CSU Features All Rhino PCI T1/E1 cards feature a single chip integrated CSU with both LIU and Framer, that is software controlled and software programmable - no jumpers Complete T1/DS1/ISDNPRI/BRI transceiver functionality Complete E1 (CEPT) PCM-30/ISDN-PRI/BRI transceiver functionality Long-haul and short-haul line interface for clock/data recovery and waveshaping Crystal-less jitter attenuator Fully independent transmit and receive functionality Single chip line interface unit (LIU) and Framer Zaptel Selections T1 or E1 T1: D4 or ESF, AMI or B8ZS E1: CAS or CCS, AMI or HDB3 Line buildouts selections: 0-133 feet or 0db, 133-266 feet, 266-399 feet, 399-533 feet, 533-655 feet, -7.5db, -15db, -22.5db Loopback configurable using Zaptel tools (i.e. zttool) Mechanical Data Size: 2.5? tall, 4.75? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 638 Rhino REC1 Echo Cancellation Module for R1T1, R2T1 and R4T1 269.0000 299.00 Rhino Rhino Digital (PRI) Cards http://www.voipon.co.uk/rhino-rec1-echo-cancellation-module-for-r1t1-r2t1-and-r4t1-p-638.html http://www.voipon.co.uk/images/rhino_rec1_r1t1_r2t1_r4t1_daughter_board.gif new Availability: In Stock Rhino REC Digital EC Plug-In Daughter Card Providing reliable, flexible, and leading-edge solutions for a demanding industry. Rhino brings to you the most awaited and most talked about addition to our product line, the 120 channel REC digital echo cancellation daugtercard that is 100% compatible with all Rhino digital T1, E1 and J1 cards. The REC card features the embedded technology from our echo cancellation partners Texas Instruments (TI) and Adaptive Digital Technologies, Inc. (Adaptive Digital). These two Rhino partners bring you a Rhino echo cancellation product line that our competition cannot beat! This daugtercard plugs directly onto the following cards: R1T1 - single port T1-E1-J1 PCI (3V and 5V) R1T1-e - single port T1-E1-J1 PCI Express (available Q4-2007) R2T1 - dual port T1-E1-J1 PCI (3V and 5V) R2T1-e - dual port T1-E1-J1 PCI Express R4T1 - quad port T1-E1-J1 PCI (3V and 5V) R4T1-e - quad port T1-E1-J1 PCI Express The REC is simply plugged onto the Rhino digital telephony card, and with a simple driver upgrade your echo cancellation capability is ready to go. If you have an older card currently in the field, simply install the REC card, upgrade our firmware, then upgrade the driver and the echo cancellation is ready to go. REC Echo Cancellation Features Small form factor, plug-in module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail - 1024 taps Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections 639 Rhino R4FXO-e-EC Quad FXO PCI Express Card w/EC 225.0000 265.00 Rhino Rhino Non-Modular Analog Cards http://www.voipon.co.uk/rhino-r4fxoeec-quad-fxo-pci-express-card-wec-p-639.html http://www.voipon.co.uk/images/r4fxo.jpg new Availability: Late January 08 Quad FXO (R4FXO) PCI Plug-In Card Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R4FXO-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R4FXO requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. R4FXO with EC Quad FXO PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R4FXO loads the PCI bus to no more than the load of a T1 card. Four RJ11 jacks at card bracket Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring 5-year limited warranty Benefits Lowest cost four port FXO card Fully Zaptel and Asterisk compliant Echo Cancellation standard Plug multiple cards into one PC without PCI worries No plug-in modules Unlimited support DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection Echo Cancellation Specs On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Mechanical Data Size: 3.90? tall, 5.50? wide Form Factor: Single PCI slot Shipping Weight: Less than one pound with all included components 641 Rhino R8FXX-e-EC Octal Analog PCI Express Card - base board w/EC 165.0000 190.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-r8fxxeec-octal-analog-pci-express-card-base-board-wec-p-641.html http://www.voipon.co.uk/images/r8fxx.jpg new Availability: Late January 08 8 Port FXS/FXO with Echo Cancellation R8FXX-EC PCI Plug-In Card THIS CARD HAS THE ABILITY TO ADD DUAL FXO AND DUAL FXS CARDS WHICH ARE SOLD SEPERATELY! Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R8FXX-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R8FXX-EC requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Octal FXX PCI Plug-In Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Infineon PEB3268 DualSLIC chip Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R8FXS-EC loads the PCI bus to no more than the load of a T1 card. Eight independant female RJ11 connectors at card bracket Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection SLIC Features On-chip uLaw or aLaw CODEC Integrated ringing generator, 65 Vrms capable +12V power derived from ATX power connector -48V DC on hook voltage, 25mA maximum loop current, loop start feed USA AC and DC impedance characteristics 500msec end-of-call battery interruption, programmable to 3 seconds MWI neon bulb capable On-hook data transmission Echo Cancellation Specs On-board - no module Adaptive Digital Technologies G.168 Echo Canceller is a carrier-class, ITU G.168 compliant line echo canceller, which meets and exceeds G.168-2002 Cancels up to 128msec tail Non-linear processor Comfort Noise Generator Automatic tail search Excellent voice quality Cancels multiple independent tails Fast Convergence No divergence due to doubletalk Tone Disabler disables echo canceller during voiceband modem and FAX connections Mechanical Data Size: 4.0? tall, 6.25? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 642 Rhino R24FXX-e-EC 24 Port Analog PCI Express Card base board EC 195.0000 220.00 Rhino Rhino Modular Analog Cards http://www.voipon.co.uk/rhino-r24fxxeec-24-port-analog-pci-express-card-base-board-ec-p-642.html http://www.voipon.co.uk/images/r24fxx.jpg new Availability: Late January 08 24 Port FXS/FXO with Echo Cancellation PCI Plug-In Card THIS CARD HAS THE ABILITY TO ADD DUAL FXO AND DUAL FXS CARDS WHICH ARE SOLD SEPERATELY! Providing reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Managing your open source telecommunication needs has never been easier than with Rhino products. Rhino PCI plug-in cards satisfy the needs of Open Source Telephony (OST) applications, no matter how stringent the requirement. Rhino Open Source Telephony PCI cards feature Asterisk, Zapata and Linux tested software. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. The Rhino R24FXS-EC is ?ready to roll? with two incredible feature differentiations over our competition - our on-board control element, and our on-board Echo Cancellation circuit. The control element eliminates PCI bus ?bit banging?, which means that the R4FXO requires less CPU power, and more Rhino cards can be used in one computer over alternative, antiquated solutions. The Echo Cancellation circuit provides echo protection no matter what, to ensure that calls are clear, crisp and echo free. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog fixed and mixed mode analog interfaces. And don?t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Using Asterisk? Rhino Open Source Analog Telephony PCI products allow you to utilize analog phones and wiring in conjunction with leading-edge Asterisk technology -- without having to buy expensive IP telephones. Why go IP when you can save on installations by using your proven existing wiring? Rhino allows you to use lower cost analog phones with digital features, get guaranteed T1 voice quality, all with less to worry about while enjoying other Asterisk features. Rhino products are tough. In the rare case of trouble, our technical support staff is ready to provide the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. 24 Port FXX PCI Card Specifications PCI Card Features Asterisk soft PBX tested and ready Zaptel-compliant open source Linux module source code On-board Texas Instruments and Adaptive Digital Technologies Echo Cancellation technology Proven Infineon PEB3268 DualSLIC chip Proven Silicon Labs FXO DAA component - Si3050 Silicon Labs international line interface device - Si3019 Custom Rhino PCI interface chip means no excess CPU overhead Rhino on-board control element eliminates PCI bus bit banging. The R24FXS loads the PCI bus to no more than the load of a T1 card. One female RJ11 connector at card bracket, with Velco strap Field software upgradable All major signaling modes supported Advanced features such as Caller ID and Distinctive Ring 5-year limited warranty DAA Features On-chip uLaw or aLaw CODEC with integrated PCM highway 80db dynamic range Tx/Rx 3 uA on-hook line monitor Programmable digital gains Line voltage and loop current monitor Integrated ring detector Programmable line interface, including AC termination, DC termination, ring detect threshold, ringer impedance to support over 70 countries Tip.Ring polarity reversal detection SLIC Features On-chip uLaw or aLaw CODEC Integrated ringing generator, 65 Vrms capable +12V power derived from ATX power connector -48V DC on hook voltage, 25mA maximum loop current, loop start feed USA AC and DC impedance characteristics 500msec end-of-call battery interruption, programmable to 3 seconds MWI neon bulb capable On-hook data transmission Mechanical Data Size: 4.0? tall, 11.00? wide Form Factor: Single PCI slot Shipping Weight: 1 pound with all included components maximum 643 Rhino Ceros Chassis, 80GB HD (CEROS-80GB-ST) - RAID1 INCLUDED 617.0000 649.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-80gb-hd-ceros80gbst-raid1-included-p-643.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 80GB Hard Drive (plus additional 80GB HDD in RAID1) for Rhino PCI Cards (CEROS-80GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   644 Rhino Ceros Chassis, 160GB Hard Drive (CEROS-160GB-ST) 599.0000 631.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-160gb-hard-drive-ceros160gbst-p-644.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 160GB Hard Drive for Rhino PCI Cards (CEROS-160GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   645 Rhino Ceros Chassis, 160GB HD (CEROS-160GB-ST) - RAID1 INCLUDED 652.0000 686.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-160gb-hd-ceros160gbst-raid1-included-p-645.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 160GB Hard Drive (plus additional 160GB HDD in RAID1) for Rhino PCI Cards (CEROS-160GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   646 Rhino Ceros Chassis, 250GB Hard Drive (CEROS-250GB-ST) 626.0000 659.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-250gb-hard-drive-ceros250gbst-p-646.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 250GB Hard Drive for Rhino PCI Cards (CEROS-250GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   647 Rhino Ceros Chassis, 250GB HD (CEROS-250GB-ST) - RAID1 INCLUDED 688.0000 724.00 Rhino Rhino Ceros Standalone IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-250gb-hd-ceros250gbst-raid1-included-p-647.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 250GB Hard Drive (plus additional 250GB HDD in RAID1) for Rhino PCI Cards (CEROS-250GB-ST) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4 Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   648 Rhino Ceros Chassis, 80GB Hard Drive (CEROS-80GB) 569.0000 579.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-80gb-hard-drive-ceros80gb-p-648.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 80GB Hard Drive for Rhino PCI Cards (CEROS-80GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   649 Rhino Ceros Chassis, 80GB HD (CEROS-80GB) - RAID1 INCLUDED 617.0000 629.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-80gb-hd-ceros80gb-raid1-included-p-649.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 80GB Hard Drive (plus additional 80GB HDD in RAID1) for Rhino PCI Cards (CEROS-80GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   650 Rhino Ceros Chassis, 160GB Hard Drive (CEROS-160GB) 599.0000 611.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-160gb-hard-drive-ceros160gb-p-650.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 160GB Hard Drive for Rhino PCI Cards (CEROS-160GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   651 Rhino Ceros Chassis, 160GB HD (CEROS-160GB) - RAID1 INCLUDED 652.0000 666.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-160gb-hd-ceros160gb-raid1-included-p-651.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 160GB Hard Drive (plus additional 160GB HDD in RAID1) for Rhino PCI Cards (CEROS-160GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration 652 Rhino Ceros Chassis, 250GB Hard Drive (CEROS-250GB) 626.0000 639.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-250gb-hard-drive-ceros250gb-p-652.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 250GB Hard Drive for Rhino PCI Cards (CEROS-250GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   653 Rhino Ceros Chassis, 250GB HD (CEROS-250GB) - RAID1 INCLUDED 688.0000 704.00 Rhino Rhino Ceros with Card IP PBX http://www.voipon.co.uk/rhino-ceros-chassis-250gb-hd-ceros250gb-raid1-included-p-653.html http://www.voipon.co.uk/images/rhino_ceros_sml.jpg new Availability: In Stock Availability: In Stock The Rhino Ceros provides reliable, flexible, and leading-edge solutions for a demanding telecommunications industry, including the Asterisk* community. Rhino Ceros Chassis with 250GB Hard Drive (plus additional 250GB HDD in RAID1) for Rhino PCI Cards (CEROS-250GB) with AMD AM2 4000+ Athlon CPU & 1GB RAM, Preloaded with trixbox CE 2.4. A Rhino PCI card must be purchased with this Ceros - options at the bottom of the page Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple six line by 40 character text file, plus your company graphical logo in .BMP format. The graphic and text in the file is displayed on the LDD at all times for your customers to visually identify your own branded PBX product. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silk screening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer.   Rhino Ceros Features Asterisk soft PBX tested Preloaded with Asterisk distrubution of your choice - currently supported: trixbox CE 2.4, PBX in a Flash, Elastix. Accepts all Rhino PCI and PCI Express cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all required hardware 3U tall, 17" wide x 13.5" deep external dimensions Washable front air filter Rubber bumper feet 19", 23" and wall mounting brackets - all included in box Dual high reliability chassis fans Six line graphical Rhino SLI, serial LCD and 5-button navigation keypad No external monitor or keyboard required Double shipping box 5-year limited warranty Rhino Ceros Base Configurations AMD AM2 4000+ Athlon CPU (Gigabyte GA-M61P-S3) 80GB SATA hard drive (upgradeable to 160 and 250GB, RAID1 also available) 1GB 667MHz DDR2 memory (upgradeable to 4GB) 400W power supply, 100-240VAC, 50-60Hz, Active PFC Optional Dual Redundant 400W power supply, 100-240VAC, 50-60Hz, Active PFC three PCI slots, three PCI Express slots Rhino SLI 6-line graphical LCD and 5-button keypad Video, mouse and keyboard ports Four USB ports, one firewire port, parallel and serial port, audio stack GbE integrated network Physical Specification Size : 3U (5.25") tall 17” wide, 13.5 ” deep All Black design Front panel aluminim colorization plate Single piece top lid, can be opended while rack mouted   Form Factor: Rack mount Wall mount Table top Weight : 25 pounds for base configuration   656 VXI Passport 20V-DC Binaural Headset 59 59.00 VXI Corporation VoIP Accessories http://www.voipon.co.uk/vxi-passport-20vdc-binaural-headset-p-656.html http://www.voipon.co.uk/images/vxi_20v.jpg new Availability: In stock Call center managers know that a quality headset means a happier, more productive agent and fewer problems overall. The ultra-comfortable fit means all-day-wear is stress free and the noise canceling microphone blocks out the sound of surrounding agents for a more professional sounding call. Made with the same impact resistant plastic used in sports equipment to withstand the turbulence of 24/7 operations, Passport headsets are made to handle the intense requirements of the professional contact center. The Passport 10V-DC and Passport 20V-DC models will connect to the QD1026, QD1027, QD1029, QD1095, QD1085, QD1028 Y cord, and QD1000 model cords already offered on this website. Compatibility with these headsets and cords is the same as with the Tuffset headsets. VXI Passport 20V-DC Features & Benefits Compatible with VXI Passport V-series amplifiers and direct connect cords. Noise-canceling microphone greatly reduces background noise for a professional sounding call Durable construction to endure the stress of the 24/7 call center Virtually indestructible, highly flexible gooseneck boom. Rotates for left or right ear placement. Maintains position for consistent voice quality Sturdy quick disconnect allows fast and easy detachment from lower cord for greater mobility Adjustable headband secures positional adjustment, won't slide out of position Concave ear cushion for greater comfort and improved sound 658 OpenVox D110P PCI ISDN PRI Card 190.0000 239.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html http://www.voipon.co.uk/images/openvox_d110p_isdn_card.gif new Availability: In stock D110P is a single span E1/T1/J1 digital line telephony product. D110P is a high-performance, cost-effective single span digital voice card. It provides a compact and powerful interface to asterisk that can support E1/T1/J1 and PRI interface. The D110P supports industry standard protocols including MFR2, PRI, Cisco PPP and Frame relay etc. The low profile allows it to fit within a 2U rack-mount case. Features: 100% hardware compatible with TE110P. Use same device driver without any modify. Industry Standard: PCI 2.2. Both 3.3 volt and 5 volt pci slot can be used. RJ-45 shielded connecter. T1/E1/J1 software configurable. Certificates: CE and FCC RoHS compliant Two year warranty 659 OpenVox D210P PCI ISDN PRI Card 317.0000 364.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-d210p-pci-isdn-pri-card-p-659.html http://www.voipon.co.uk/images/openvox_d210p_isdn_card_s.gif new Availability: In stock The D210P is the latest hardware addition to the Openvox array of digital cards. The D210P supports E1, T1 and J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1 and T1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. With the improved I/O speed, the card reduces CPU usage and increased card density per server. D210P is fully compatible with Asterisk™ application. Also, the open source driver supports an API interface for custom application development. The D210P support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. Other Features includes: Support for both 3.3v and 5v PCI slots. High Performance PCI Bus Master with up to 2Kbytes FIFO Easy to install: Use wct4xxp driver included in original Zaptel without any patch. RoHS compliant Certificates: CE and FCC Two year warranty 660 OpenVox D410P PCI ISDN PRI Card 423.0000 492.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-d410p-pci-isdn-pri-card-p-660.html http://www.voipon.co.uk/images/openvox_d410p_isdn_card_s.jpg new Availability: In stock The D410P is the latest hardware addition to the Openvox array of digital cards. The D410P supports E1, T1 and J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1 and T1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. With the improved I/O speed, the card reduces CPU usage and increased card density per server. D410P is fully compatible with Asterisk application. Also, the open source driver supports an API interface for custom application development. The D410P support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. Other Features includes: Support for both 3.3v and 5v PCI slots. High Performance PCI BusMaster with up to 2Kbytes FIFO. Easy to install: Use wct4xxp driver included in original Zaptel without any patch. Certificates: CE and FCC Two year warranty 661 OpenVox B200P PCI ISDN BRI Card 158.0000 196.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b200p-pci-isdn-bri-card-p-661.html http://www.voipon.co.uk/images/openvox_b200p_isdn_card_s.gif new Availability: In stock The B200P is a PCI 2.2 compliant card supporting 2 BRI S/T interfaces, with a onboard multi NT powerfeeding circuit . NT/TE mode can be independently configured on each of the 2 ports. The B200P can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI bus interface (Spec.2.2) for 3.3V and 5V signal environments DTMF detection on all B-channels Multiparty audio conferences bridge Onboard power feeding PCM bus connectors daisy chaining Each of the 2 ports can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 662 OpenVox B400P PCI ISDN BRI Card 236.0000 280.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b400p-pci-isdn-bri-card-p-662.html http://www.voipon.co.uk/images/openvox_b400p_isdn_card_s.gif new Availability: In stock The B400P is a PCI 2.2 compliant card supporting 4 BRI S/T interfaces, with a onboard multi NT powerfeeding circuit . NT/TE mode can be independently configured on each of the 4 ports. The B400P can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI bus interface (Spec.2.2) for 3.3V and 5V signal environments DTMF detection on all B-channels Multiparty audio conferences bridge Onboard power feeding PCM bus connectors daisy chaining Each of the 4 ports can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 663 OpenVox B800P PCI ISDN BRI Card 345.0000 424.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b800p-pci-isdn-bri-card-p-663.html http://www.voipon.co.uk/images/openvox_b800p_isdn_card_s.gif new Availability: In stock The B800P is a PCI 2.2 compliant card supporting 8 BRI S/T interfaces, with a onboard multi NT powerfeeding circuit . NT/TE mode can be independently configured on each of the 8 ports. The B800P can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI bus interface (Spec.2.2) for 3.3V and 5V signal environments DTMF detection on all B-channels Multiparty audio conferences bridge Onboard power feeding PCM bus connectors daisy chaining Each of the 8 ports can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 668 OpenVox A400P04 - 4 FXO 133.0000 166.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p04-4-fxo-p-668.html http://www.voipon.co.uk/images/openvox_a400p04_sml.jpg new Availability: In stock The A400P03 is an OpenVox A400P bundled with 4 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P01 comes with four FXO modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 669 OpenVox A400P01 - 1 FXO 55.0000 69.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p01-1-fxo-p-669.html http://www.voipon.co.uk/images/openvox_a400p01_sml.jpg new Availability: In stock The A400P01 is an OpenVox A400P bundled with 1 FXO module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P01 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 670 OpenVox A400P02 - 2 FXO 82.0000 102.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p02-2-fxo-p-670.html http://www.voipon.co.uk/images/openvox_a400p02_sml.jpg new Availability: In stock The A400P02 is an OpenVox A400P bundled with 2 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P02 comes with two FXO modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 671 OpenVox A400P03 - 3 FXO 107.0000 134.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p03-3-fxo-p-671.html http://www.voipon.co.uk/images/openvox_a400p03_sml.jpg new Availability: In stock The A400P03 is an OpenVox A400P bundled with 3 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P03 comes with three FXO modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 672 OpenVox A400P10 - 1 FXS 52.0000 65.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p10-1-fxs-p-672.html http://www.voipon.co.uk/images/openvox_a400p10_sml.jpg new Availability: In stock The A400P10 is an OpenVox A400P bundled with 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P10 comes with one FXS module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 673 OpenVox A400P40 - 4 FXS 119.0000 148.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p40-4-fxs-p-673.html http://www.voipon.co.uk/images/openvox_a400p40_sml.jpg new Availability: In stock The A400P40 is an OpenVox A400P bundled with 4 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P40 comes with four FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 674 OpenVox A400P20 - 2 FXS 74.0000 92.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p20-2-fxs-p-674.html http://www.voipon.co.uk/images/openvox_a400p20_sml.jpg new Availability: In stock The A400P20 is an OpenVox A400P bundled with 2 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P20 comes with two FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 675 OpenVox A400P30 - 3 FXS 96.0000 120.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p30-3-fxs-p-675.html http://www.voipon.co.uk/images/openvox_a400p30_sml.jpg new Availability: In stock The A400P30 is an OpenVox A400P bundled with 3 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P30 comes with three FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 676 OpenVox A400P13 - 3 FXO 1 FXS 128.0000 160.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p13-3-fxo-1-fxs-p-676.html http://www.voipon.co.uk/images/openvox_a400p13_sml.jpg new Availability: In stock The A400P13 is an OpenVox A400P bundled with 3 FXO modules and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones.   Key Features and benefits:   Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P13 comes with three FXO and one FXS module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 677 OpenVox A400P21 - 1 FXO 2 FXS 99.0000 124.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p21-1-fxo-2-fxs-p-677.html http://www.voipon.co.uk/images/openvox_a400p21_sml.jpg new Availability: In stock The A400P21 is an OpenVox A400P bundled with 1 FXO module and 2 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P21 comes with one FXO an d two fxs modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 678 OpenVox A400P22 - 2 FXO 2 FXS 125.0000 156.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p22-2-fxo-2-fxs-p-678.html http://www.voipon.co.uk/images/openvox_a400p22_sml.jpg new Availability: In stock The A400P22 is an OpenVox A400P bundled with 2 FXO modules and 2 FXS modules. You can add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P22 comes with two FXO and two fxs modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 679 OpenVox A400P12 - 2 FXO 1 FXS 102.0000 128.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p12-2-fxo-1-fxs-p-679.html http://www.voipon.co.uk/images/openvox_a400p12_sml.jpg new Availability: In stock The A400P12 is an OpenVox A400P bundled with 2 FXO modules and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P12 comes with otwo FXO modules and 1 fxs module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 680 OpenVox A400P11 - 1 FXO 1 FXS 77.0000 96.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p11-1-fxo-1-fxs-p-680.html http://www.voipon.co.uk/images/openvox_a400p11_sml.jpg new Availability: In stock The A400P11 is an OpenVox A400P bundled with 1 FXO module and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P11 comes with one FXO and one fxs module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 681 OpenVox A400P31 - 1 FXO 3 FXS 122.0000 152.00 OpenVox OpenVox A400P http://www.voipon.co.uk/openvox-a400p31-1-fxo-3-fxs-p-681.html http://www.voipon.co.uk/images/openvox_a400p31_sml.jpg new Availability: In stock The A400P31 is an OpenVox A400P bundled with 1 FXO module and 3 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400P is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The A400P supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. (A400P31 comes with one fxo and three FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 682 OpenVox FXS-100 Module 29.0000 36.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-fxs100-module-p-682.html http://www.voipon.co.uk/images/openvox_fxs-100_module_s.gif new Availability: In Stock An OpenVox FXS-100 module allows the A400P, A800P, and A1200P cards to terminate analog telephones. Thanks to the modular design, users can integrate additional ports at any time by adding more FXO-100 or FXS-100 daughter cards to the A400P, A800P, and A1200P. The FXS-100 module passes all the call features that any standard analog telephone will support. Key Features: Easy to upgrade: Simply attach additional FXS-100 or FXO-100 modules to card to upgrade the system. Compatibility: FXS-100 is Pin to Pin compatible with Digium's S100M daughter card. User can use this module with TDM400P and TDM800P cards as well as OpenVox A400P, A800P, and A1200P. RoHS compliant Certificates: CE and FCC Ttwo year Warranty. 683 OpenVox FXO-100 Module 32.0000 37.80 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-fxo100-module-p-683.html http://www.voipon.co.uk/images/openvox_fxo-100_module_s.jpg new Availability: In Stock An OpenVox FXO-100 module allows the A400P, A800P, and A1200P cards to terminate analog telephone lines (POTS). Thanks to the modular design, users can integrate additional ports at any time by adding more FXO-100 or FXS-100 daughter cards to the A400P, A800P, and A1200P. The FXO-100 module passes all the call features that any standard analog telephone line will support. Key Features: Easy to upgrade: Simply attach additional FXO-100 modules to card to upgrade the system. Compatibility: FXO-100 is Pin to Pin compatible with Digium's X100M daughter card. User can use this module with TDM400P and TDM800P cards as well as OpenVox A400P, A800P, and A1200P. RoHS compliant Certificates: CE and FCC Two year Warranty. 684 OpenVox A800P10 - 1 FXS 104.0000 130.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p10-1-fxs-p-684.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P10 is an OpenVox A800P bundled with 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P with 4 FXS and 4 FXO modules installed. A800P10 has 1 FXS100(green) module pre-installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P10 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 685 OpenVox A1200P0100 - 1 FXS 104.0000 130.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0100-1-fxs-p-685.html http://www.voipon.co.uk/images/openvox_a1200p0100_sml.jpg new Availability: In stock The A1200P0100 is an OpenVox A1200P bundled with 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0100 has 1 FXS100(green) module pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0100 comes with one FXS module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 686 OpenVox A800P44 - 4 FXO 4 FXS 233.0000 333.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p44-4-fxo-4-fxs-p-686.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P44 is an OpenVox A800P bundled with 4 FXO modules and 4 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P44 with 4 FXS(green) and 4 FXO(red) modules installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P44 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 687 OpenVox A800P80 - 8 FXS 228.0000 327.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p80-8-fxs-p-687.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P80 is an OpenVox A800P bundled with 8 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P with 4 FXS and 4 FXO modules installed. A800P80 has 8 FXS100(green) modules pre-installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P80 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 688 OpenVox A800P11 - 1 FXO 1 FXS 133.0000 167.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p11-1-fxo-1-fxs-p-688.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P11 is an OpenVox A800P bundled with 1 FXO module and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P with 4 FXS and 4 FXO modules installed. A800P11 has 1 FXO100(red) module, and 1 FXS100(green) module pre-installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P11 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 689 OpenVox A800P01 - 1 FXO 107.0000 134.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p01-1-fxo-p-689.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P01 is an OpenVox A800P bundled with 1 FXO module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P with 4 FXS and 4 FXO modules installed. A800P01 has 1 FXO100(red) module pre-installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P01 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 690 OpenVox A800P08 - 8 FXO 238.0000 340.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-a800p08-8-fxo-p-690.html http://www.voipon.co.uk/images/openvox_a800p_s.jpg new Availability: In stock The A800P08 is an OpenVox A800P bundled with 8 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A800P with 4 FXS and 4 FXO modules installed. A800P08 has 8 FXO100(red) modules pre-installed. OpenVox A800P is designed to be fully compatible with the TDM800P Asterisk Analog Telephony card. User can simply use A800P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Firmware accelerates I/O processes to achieve high stability and highly decreased CPU payload (1x A800P with 8 modules installed uses less CPU cycle than 1x Digium TDM400P with only 4 modules installed). The A800P supports up to a total of 8 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A800P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A800P card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: Full software and hardware compatible with TDM800P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 8 per card. (A800P08 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Compatability: Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Bundle: Includes 2 RJ11 to RJ45 splitters which plug into each of the 2 ports on the card to provide 8 ports for use on a fully expanded A800P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 691 OpenVox A1200P0008 - 8 FXO 238.0000 340.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0008-8-fxo-p-691.html http://www.voipon.co.uk/images/openvox_a1200p0008_sml.jpg new Availability: In stock The A1200P0008 is an OpenVox A1200P bundled with 8 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0008 has 8 FXO100(red) modules pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0008 comes with eight FXO modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 692 OpenVox A1200P0404 - 4 FXO 4 FXS 233.0000 333.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0404-4-fxo-4-fxs-p-692.html http://www.voipon.co.uk/images/openvox_a1200p0404_sml.jpg new Availability: In stock The A1200P0404 is an OpenVox A1200P bundled with 4 FXO modules and 4 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0404 has 4 FXO100(red) modules and 4 FXS100(green) modules pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0404 comes with four FXO modules and four FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 693 OpenVox A1200P0800 - 8 FXS 228.0000 327.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0800-8-fxs-p-693.html http://www.voipon.co.uk/images/openvox_a1200p0800_sml.jpg new Availability: In stock The A1200P0800 is an OpenVox A1200P bundled with 8 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0800 has 8 FXS100(green) modules pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0800 comes with eight FXS modules.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 694 OpenVox A1200P0001 - 1 FXO 107.0000 134.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0001-1-fxo-p-694.html http://www.voipon.co.uk/images/openvox_a1200p0001_sml.jpg new Availability: In stock The A1200P0001 is an OpenVox A1200P bundled with 1 FXO module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0001 has 1 FXO100(red) module pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0001 comes with one FXO module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 695 OpenVox A1200P0101 - 1 FXO 1 FXS 133.0000 167.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0101-1-fxo-1-fxs-p-695.html http://www.voipon.co.uk/images/openvox_a1200p0101_sml.jpg new Availability: In stock The A1200P0101 is an OpenVox A1200P bundled with 1 FXO module and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P with 12 FXO modules installed. A1200P0101 has 1 FXO100(red) module and 1 FXS100(green) module pre-installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 12 per card. (A1200P0101 comes with one FXO module and one FXS module.) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 698 Siemens Gigaset C475IP Dect Phone 65.2000 68.00 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-gigaset-c475ip-dect-phone-p-698.html http://www.voipon.co.uk/images/gigaset_c475ip_dect_phone.jpg new Availability: In stock The attractive VoIP and fixed-line phone for up to 3 parallel calls, plus an integrated answering machine . The Voice over IP revolution is underway – and you can take advantage of cost-efficient internet telephony with the attractive Gigaset C475 IP cordless phone. Featuring an integrated answering machine, this smart hybrid phone is the ultimate in modern communication. It lets you easily switch between internet and fixed-line calls with a single press of a button on the phone’s handset – without a PC. You can also expand your phone system with extra handsets so up to three family members and friends can telephone at the same time. If you’ve missed any calls while out and about, the Gigaset C475 IP offers a handy message notification feature that will send you an SMS² and alert you about new messages on the machine. This Voice over IP model also features Gigaset’s energy-saving ECO DECT technology, which uses up to 60% less energy than our conventional cordless phones – so you can save money and be friendlier to the environment. If you’re looking for an attractive hybrid phone with an integrated answering machine and an array of advanced messaging options, then look no further than the Gigaset C475 IP. An answering machine for important messages You’re always in reach with the Gigaset C475 IP. This model’s integrated answering machine provides up to 30 minutes of recording time and will display a list of your new messages on the handset’s illuminated display. You can also access your messages from anywhere in the world thanks to the remote operation function. For extra convenience, the answering machine lets you individually select the recording time for messages and even gives you the option to record phone calls. Hybrid phone convenience The Gigaset C475 IP is a modern hybrid phone that lets you easily switch between internet and fixed-line phoning with a single press of a key on the phone’s handset – without a PC . You can also add up to 6 extra handsets to the phone system and experience the flexibility of multiline calling. So your family members or friends can make up to 2 internet calls and 1 fixed-line call at the same time. Advanced messaging for busy lifestyles Stay in touch with family and friends no matter where you are in the world – the Gigaset C475 IP offers you an array of options. The handy e-mail message notification function sends an alert to the handset with the time, date and subject of any new e-mails you receive. You don’t even have to be online. When you’re away from home, let this Voice over IP phone conveniently inform you of any missed calls by sending you an SMS – or send your own SMS from the phone’s handset if you’d love to chat, but just don’t have time. A phonebook for all your contacts Store your list of contacts in one convenient place and always have it at hand. The Gigaset C475 IP features an extra-large phonebook that lets you manage the numbers of up to 150 of your family members and friends. Finding, editing or deleting your contacts is also simple thanks to the handset’s easy-to-read illuminated display and intuitive icon-based menu. Talk handsfree in brilliant sound quality Thanks to this phone’s handsfree function, you can enjoy more comfortable telephone conversations. Simply carry on with the day’s tasks or just lie back on the couch and relax with your hands behind your head – and still be able to chat with friends or family in brilliant sound quality. Your world’s at your fingertips with Gigaset.net There’s no quicker way to start phoning with your loved ones over the internet than with Gigaset.net . Purchase the Gigaset C475 IP phone and you’ll automatically become a member of this exclusive Gigaset IP network community. That means you’ll be able to make Voice over IP calls to anyone else with a Gigaset IP phone – anywhere and anytime for free – and you won’t even need an account with a VoIP provider. What’s more, you’ll be able to access free subscription services such as RSS feeds and weather information directly via your Gigaset handset. ECO DECT** for saving energy and costs Thanks to the energy-saving power supply, the Gigaset C475 IP uses up to 60% less energy than our conventional cordless phones – making it easier on the environment and more cost effective. Like all Gigaset cordless phones, it also variably reduces transmitting power from the handset to base station according to their distance apart – but goes further in a number of other ways. Choosing ECO Mode on this model reduces the base station’s transmission power by 80% ³. As a result, you can enjoy a more energy-efficient way of staying connected with the world. Specially designed for keeping in touch with friends and family around the globe, the attractive Gigaset C475 IP offers you all the convenience of cordless Voice over IP communication, plus an integrated answering machine.     Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features All units supplied as base and 1 handset - Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. All units can connect to both a standard BT phone line and have an ethernet connection for VoIP calling. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 699 Siemens Gigaset C47 DECT Handset 35.0000 37.00 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-gigaset-c47-dect-handset-p-699.html http://www.voipon.co.uk/images/gigaset_c47_dect_handset.jpg new Availability: In stock The attractive VoIP and fixed-line phone for up to 3 parallel calls, plus an integrated answering machine . The Voice over IP revolution is underway – and you can take advantage of cost-efficient internet telephony with the attractive Gigaset C47 cordless phone. Featuring an integrated answering machine, this smart hybrid phone is the ultimate in modern communication. It lets you easily switch between internet and fixed-line calls with a single press of a button on the phone’s handset – without a PC. You can also expand your phone system with extra handsets so up to three family members and friends can telephone at the same time. If you’ve missed any calls while out and about, the Gigaset C47 offers a handy message notification feature that will send you an SMS² and alert you about new messages on the machine. This Voice over IP model also features Gigaset’s energy-saving ECO DECT technology, which uses up to 60% less energy than our conventional cordless phones – so you can save money and be friendlier to the environment. If you’re looking for an attractive hybrid phone with an integrated answering machine and an array of advanced messaging options, then look no further than the Gigaset C47. Requires a Siemens Gigaset C475IP.     Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 700 VoiSmart SIP/GSM Gateway - 4 channels (GW-0950-01-4SIM) 1585.0000 1825.00 VoiSmart VoiSmart GSM / SMS Gateways http://www.voipon.co.uk/voismart-sipgsm-gateway-4-channels-gw0950014sim-p-700.html http://www.voipon.co.uk/images/voismart_gateway_gsm.jpg new Availability: In Stock VoiSmart SIP/GSM Gateway – 4 channels is an Asterisk-based IP Gateway which allows you to transform fixed-to-mobile into mobile-to-mobile traffic in the simplest and most efficient way. It's a true “black box”, which can be reached as a SIP client/sever. It allows multiple simultaneously active GSM channels, in different case configurations (4 engines). The VoiSmart Gateways support SIM server functionalities for remote and central storing of any number of Smartcards/SIM. No extra hardware to be added, no installation effort, no technical skills required: a real plug-and-play device, designed to make your business immediately running. You can see it as a standard SIP client/server, and integration is done. LCR and smart routing of incoming calls are immediately available, as well as all its powerful features. A Professional Solution for Professional use Powerful and versatile, think of it as a pure node in the network. VoiSmart SIP/GSM gateway is the building brick on which you will rely to implement high profile solutions, as well as the next door to the world of message communication. A perfect match for Asterisk architectures, ready to be integrated in non-Asterisk environment as well. Main Features SIP / IAX supported All in one: no extra hardware or software required Easy to integrate Easy to install Simultaneous operation on multiple channels (4) SIM Server support for remote storage and management Open Source based (Asterisk) Full Least Cost Routing support Incoming calls routing Echo cancellation on GSM engines High efficiency thanks to multiple channels operation Compact fanless design GSM features Codecs: G.711 PCM at 64 kbps, G.729/Annex A CS-ACELP at 8 kbps, G.723.1 (optional) MP-MLQ/ACELP at 6.3/5.3 kbps Compatible with European, US, Brazil and Japan GSM networks (900/1800/1900 MHz and 850/1800/1900nMHz) Up to 4 SIMs, multiprovider External interface: 4 SMA/F 50 Ohm antenna connectors SMS: supports MT, MO, CB, text and PDU SIM: supports SIM card (3V) IRQ level requests: allocated by PCI BIOS IO base address (hex): allocated by PCI BIOS PCM: a-law or u-law (selectable) Protocols: TCP, UDP, IP, RTP, SSH, Web SIM: supports SIM card (3V) GSM engines: Siemens MC55, Siemens MC56 LCR (Least Cost Routing) also in AutoCLIP mode DISA with voice navigation, user configurable messages for incoming calls Administration: through SSH/Web Interfaces: LAN 10/100 Base-T, RJ45, RS232 SIM Physical Data 110 – 240V CA, output +12V CC, 1,2A max (15W max) Dimensions: 30 x 7 x 20 cm Weight: 3 Kg Temperature: from 0 to 40° C Humidity: from 10 to 95%, noncondensing 701 VoiSmart SMS Gateway - 4 channels (GW-0930-03-4SIM) 1254.0000 1445.00 VoiSmart VoiSmart GSM / SMS Gateways http://www.voipon.co.uk/voismart-sms-gateway-4-channels-gw0930034sim-p-701.html http://www.voipon.co.uk/images/voismart_gateway_sms.jpg new Availability: In Stock VoiSmart SMS Server is an Asterisk-based gateway which allows you to send and receive high volumes of SMS traffic in the simplest and most efficient way. It's a true black box, which can be reached as an IP address, ready to be configured as a Mail server or a POP3 client. It allows multiple RX/TX (4) simultaneously active GSM channels. No extra hardware to be added, no installation effort, no technical skills: a real plug-and earn device, designed to make your business immediately running. Easy to be integrated with professional SMS management systems, it's also an all-in-one system to start sending and receiving SMSs from the most popular mailing applications (Lotus Notes, Outlook, etc.). A Professional Solution for Professional use Powerful and versatile, think of it as a pure node in the network. VoiSmart IP SMS Gateway is the building brick on which you will rely to implement high profile solutions, as well as the next door to the world of message communication. Main Features Configurable either as Mail Server or POP3 Client Fully transparent towards legacy systems All in one: no extra Hardware or Software needed Easy to integrate Easy to install Simultaneous operation on multiple channels (4) Open Source based (Asterisk) Allows broadcasting of SMSs Unicode support for foreign alphabets Gives you freedom of choosing your provider Makes you master your business: no more campaigns to be subscribed and planned in advance! High efficiency thanks to multiple channels operation Compact fanless design SMS Feature Send and receive messages on multiple channels simultaneously Multinumbering format supported Multipart supported, for messages longer than 160 characters Support for SMS delivery verification Support for Unicode for foreign alphabets (Russian, Chinese, Arabic, etc.) Multiple network connections supported SMTP and POP3 supported in native mode GSM features Codecs: G.711 PCM at 64 kbps, G.729/Annex A CS-ACELP at 8 kbps, G.723.1 (optional) MP-MLQ/ACELP at 6.3/5.3 kbps Compatible with European, US, Brazil and Japan GSM networks (900/1800/1900 MHz and 850/1800/1900nMHz) Up to 4 SIMs, multiprovider External interface: 4 SMA/F 50 Ohm antenna connectors SMS: supports MT, MO, CB, text and PDU SIM: supports SIM card (3V) IRQ level requests: allocated by PCI BIOS IO base address (hex): allocated by PCI BIOS PCM: a-law or u-law (selectable) Protocols: TCP, UDP, IP, RTP, SSH, Web SIM: supports SIM card (3V) GSM engines: Siemens MC55, Siemens MC56 LCR (Least Cost Routing) also in AutoCLIP mode DISA with voice navigation, user configurable messages for incoming calls Administration: through SSH/Web Interfaces: LAN 10/100 Base-T, RJ45, RS232 SIM Physical Data 110 – 240V CA, output +12V CC, 1,2A max (15W max) Dimensions: 30 x 7 x 20 cm Weight: 3 Kg Temperature: from 0 to 40° C Humidity: from 10 to 95%, noncondensing 702 Linksys Power supply 7 7.00 Linksys Linksys IP Telephones http://www.voipon.co.uk/linksys-power-supply-p-702.html http://www.voipon.co.uk/images/no_image_available.jpg new Spare UK power supply for Linksys Telephone. 703 VoiSmart SIP/GSM-SMS Gateway - 4 channels (GW-0930-01-4SIM) 2377.0000 2735.00 VoiSmart VoiSmart GSM / SMS Gateways http://www.voipon.co.uk/voismart-sipgsmsms-gateway-4-channels-gw0930014sim-p-703.html http://www.voipon.co.uk/images/voismart_gateway_gsm.jpg new Availability: In stock VoiSmart SIP/GSM-SMS Gateway is an Asterisk-based IP Gateway which allows you to transform fixed-to-mobile into mobile-to-mobile traffic in the simplest and most efficient way. It's a true “black box”, which can be reached as a SIP client/sever. VoiSmart SIP/GSM-SMS Gateway integrate the SMS server feature that allows to send and receive high volumes of SMS traffic in the simplest and most efficient way. It is ready to be configured as a Mail server or a POP3 client. It allows multiple simultaneously active GSM channels, in different case configurations (4 engines). The VoiSmart Gateways support SIM server functionalities for remote and central storing of any number of Smartcards/SIM. No extra hardware to be added, no installation effort, no technical skills required: a real plug-and-play device, designed to make your business immediately running. You can see it as a standard SIP client/server, and integration is done. LCR and smart routing of incoming calls are immediately available, as well as all its powerful features. Easy to be integrated with professional SMS management systems, it's also a all-in-one system to start sending and receiving SMSs from the most popular mailing applications (Lotus Notes, Outlook, etc.). A Professional Solution for Professional use Powerful and versatile, think of it as a pure node in the network. VoiSmart SIP/GSM-SMS Gateway is the building block on which you will rely to implement high profile solutions, as well as the next door to the world of message communication. A perfect match for Asterisk architectures, ready to be integrated in non-Asterisk environment as well. Main Features SIP / IAX supported All in one: no extra hardware or software required Easy to integrate Easy to install Simultaneous operation on multiple channels (4) SIM Server support for remote storage and management Open Source based (Asterisk) Full Least Cost Routing support Incoming calls routing Echo cancellation on GSM engines Allows broadcasting of SMSs Configurable either as Mail Server or POP3 Client High efficiency thanks to multiple channels operation Compact fanless design GSM features Codecs: G.711 PCM at 64 kbps, G.729/Annex A CS-ACELP at 8 kbps, G.723.1 (optional) MP-MLQ/ACELP at 6.3/5.3 kbps Compatible with European, US, Brazil and Japan GSM networks (900/1800/1900 MHz and 850/1800/1900nMHz) Up to 4 SIMs, multiprovider External interface: 4 SMA/F 50 Ohm antenna connectors SMS: supports MT, MO, CB, text and PDU SIM: supports SIM card (3V) IRQ level requests: allocated by PCI BIOS IO base address (hex): allocated by PCI BIOS PCM: a-law or u-law (selectable) Protocols: TCP, UDP, IP, RTP, SSH, Web SIM: supports SIM card (3V) GSM engines: Siemens MC55, Siemens MC56 LCR (Least Cost Routing) also in AutoCLIP mode DISA with voice navigation, user configurable messages for incoming calls Administration: through SSH/Web Interfaces: LAN 10/100 Base-T, RJ45, RS232 SIM SMS Feature Send and receive messages on multiple channels simultaneously Multinumbering format supported Multipart supported, for messages longer than 160 characters Support for SMS delivery verification Support for Unicode for foreign alphabets (Russian, Chinese, Arabic, ecc.) Multiple network connections supported SMTP and POP3 supported in native mode Physical Data 110 – 240V CA, output +12V CC, 1,2A max (15W max) Dimensions: 30 x 7 x 20 cm Weight: 3 Kg Temperature: from 0 to 40° C Humidity: from 10 to 95%, noncondensing 704 VoiSmart SIP/GSM Gateway Rack - 4-32 channels (GW-0930-02-RACK) 1585.0000 1825.00 VoiSmart VoiSmart GSM / SMS Gateways http://www.voipon.co.uk/voismart-sipgsm-gateway-rack-432-channels-gw093002rack-p-704.html http://www.voipon.co.uk/images/voismart_gateway_rack.jpg new Availability: In stock VoiSmart SIP/GSM Gateway Rack Unit is an Asterisk-based IP Gateway which allows you to transform fixed-to-mobile into mobile-to-mobile traffic in the simplest and most efficient way. It's a true “black box”, which can be reached as a SIP client/sever. It allows multiple simultaneously active GSM channels in a scalable solution that can run from 4 to 32 engines (each slot can host a vGSM 4 engines PCI card). The VoiSmart Gateways support SIM server functionalities for remote and central storing of any number of Smartcards/SIM. No extra hardware to be added, no installation effort, no technical skills required: a real plug-and-play device, designed to make your business immediately running. You can see it as a standard SIP client/server, and integration is done. LCR and smart routing of incoming calls are immediately available, as well as all its powerful features. A Professional Solution for Professional use Powerful and versatile, think of it as a pure node in the network. VoiSmart SIP/GSM gateway is the building brick on which you will rely to implement high profile solutions, as well as the next door to the world of message communication. A perfect match for Asterisk architectures, ready to be integrated in non-Asterisk environment as well. Main Features SIP / IAX supported All in one: no extra hardware or software required Easy to integrate Easy to install Simultaneous operation on multiple channels (4 to 32) SIM Server support for remote storage and management Open Source based (Asterisk) Full Least Cost Routing support Incoming calls routing Echo cancellation on GSM engines High efficiency thanks to multiple channels operation Compact fanless design GSM features Codecs: G.711 PCM at 64 kbps, G.729/Annex A CS-ACELP at 8 kbps, G.723.1 (optional) MP-MLQ/ACELP at 6.3/5.3 kbps Compatible with European, US, Brazil and Japan GSM networks (900/1800/1900 MHz and 850/1800/1900nMHz) Up to 4 SIMs, multiprovider External interface: 4 SMA/F 50 Ohm antenna connectors SMS: supports MT, MO, CB, text and PDU SIM: supports SIM card (3V) IRQ level requests: allocated by PCI BIOS IO base address (hex): allocated by PCI BIOS PCM: a-law o u-law (selectable) Protocols: TCP, UDP, IP, RTP, SSH, Web SIM: supports SIM card (3V) GSM engines: Siemens MC55, Siemens MC56 LCR (Least Cost Routing) also in AutoCLIP mode DISA with voice navigation, user configurable messages for incoming calls Administration: through SSH/Web Interfaces: LAN 10/100 Base-T, RJ45, RS232 SIM Physical Data 110 – 240V CA Dimensions: 19 inch Rack mounted, 4U standard Weight: 10 Kg Temperature: from 0 to 40° C Humidity: from 10 to 95%, noncondensing Configurations 8 channels: GW-0930-02-RACK + 2 BD-1100-01-4SIM (2 vGSM 4 channels card) 12 channels: GW-0930-02-RACK + 3 BD-1100-01-4SIM (3 vGSM 4 channels card) 16 channels: GW-0930-02-RACK + 4 BD-1100-01-4SIM (4 vGSM 4 channels card) 20 channels: GW-0930-02-RACK + 5 BD-1100-01-4SIM (5 vGSM 4 channels card) 24 channels: GW-0930-02-RACK + 6 BD-1100-01-4SIM (6 vGSM 4 channels card) 28 channels: GW-0930-02-RACK + 7 BD-1100-01-4SIM (7 vGSM 4 channels card) 32 channels: GW-0930-02-RACK + 8 BD-1100-01-4SIM (8 vGSM 4 channels card) 705 Snom 360 + GN Netcom/Jabra 9120 MidiBoom Headset 312 312.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-gn-netcomjabra-9120-midiboom-headset-p-705.html http://www.voipon.co.uk/images/snom_360_netcom_bundle_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MidiBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it\\'s mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone GN Netcom 9120 MidiBoom wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 706 Snom 320 + GN Netcom/Jabra 9120 Midiboom Headset 283 283.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-gn-netcomjabra-9120-midiboom-headset-p-706.html http://www.voipon.co.uk/images/snom_320_netcom_bundle.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MidiBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it\'s mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone GN Netcom 9120 MidiBoom wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption. 707 Snom 370 + GN Netcom/Jabra 9120 MidiBoom Headset 348 348.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-gn-netcomjabra-9120-midiboom-headset-p-707.html http://www.voipon.co.uk/images/snom_370_netcom_bundle_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MidiBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it\'s mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 370 IP Telephone GN Netcom 9120 MidiBoom wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 708 OpenVox D410E PCI Express ISDN PRI Card 466.0000 558.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-d410e-pci-express-isdn-pri-card-p-708.html http://www.voipon.co.uk/images/openvox_d410e_isdn_card_s.jpg new Availability: In stock The D410E is the latest hardware addition to the Openvox array of digital cards. It is a PCI-Express version of the D410P. The D410E supports E1, T1 and J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1 and T1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. With the improved I/O speed, the card reduces CPU usage and increased card density per server. D410P is fully compatible with Asterisk application. Also, the open source driver supports an API interface for custom application development. The D410E support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. Other Features includes: Support for PCI Express 1.0. High Performance PCI BusMaster with up to 2Kbytes FIFO. Easy to install: Use wct4xxp driver included in original Zaptel without any patch. Certificates: CE and FCC Two year warranty 709 Aastra SIP WLAN 312 IP Telephone 212.0000 220.00 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/aastra-sip-wlan-312-ip-telephone-p-709.html http://www.voipon.co.uk/images/aastra_312_WLAN_s.jpg new Availability: In stock The Aastra 312 WLAN IP phone allows cordless telephony over Wi-Fi networks connecting to any SIP proxy or SIP enabled PBX. It includes a variety of useful features such as a large back-lit graphic colour display, headset connector, hands-free operation and vibrating alert. The memory card can store up to 100 phonebook entries for ease of use. The Aastra 312 WLAN phone supports large infrastructures with easy deployment options and provides diagnostic tools to assist in service and maintenance. Features Supports wireless infrastructures according to IEEE 802.11b/g with access to a SIP proxy or SIP enabled PBX. MEM-Card (for phonebook entries, device-specific data and MAC-address.) Colour display (1,8", 128 x 160 dots, 65536 colours.) Customizable: 30 ringtones available, display background image can be selected, and more. Headset connection (2,5 mm jack.) Vibration alert. Alarm settings. Automatic key lock.   Technical Specification   Standards: 802.11b, 802.11g Security: WEP, WPA1-PSK, WPA2-PSK Quality of Service: WME / 802.11e prority Stand-by-time / Talk timeup to 50 h* / up to 6 h* Dimensions: length / width / depth146 x 50 x 28 mm Battery: Lithium-Polymer battery pack, 1000 mAh, battery compartment locked Weight: 144g incl. batteries 710 Snom 320 + GN Netcom/Jabra 9120 MicroBoom Headset 289 289.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-gn-netcomjabra-9120-microboom-headset-p-710.html http://www.voipon.co.uk/images/snom_320_netcom_micro.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MicroBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone GN Netcom 9120 MicroBoom wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption. 711 Snom 360 + GN Netcom/Jabra 9120 MicroBoom Headset 317 317.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-gn-netcomjabra-9120-microboom-headset-p-711.html http://www.voipon.co.uk/images/snom_360_netcom_micro.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MicroBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone GN Netcom 9120 MicroBoom wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 712 Snom 370 + GN Netcom/Jabra 9120 MicroBoom Headset 353 353.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-gn-netcomjabra-9120-microboom-headset-p-712.html http://www.voipon.co.uk/images/snom_370_netcom_micro.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 MicroBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it\'s mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 370 IP Telephone GN Netcom 9120 MicroBoom wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 713 Snom 320 + GN Netcom/Jabra 9120 FlexBoom Headset 294 294.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-gn-netcomjabra-9120-flexboom-headset-p-713.html http://www.voipon.co.uk/images/snom_320_netcom_flex.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 FlexBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone GN Netcom 9120 FlexBoom wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption. 714 Snom 360 + GN Netcom/Jabra 9120 FlexBoom Headset 322 322.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-gn-netcomjabra-9120-flexboom-headset-p-714.html http://www.voipon.co.uk/images/snom_360_netcom_flex.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 FlexBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone GN Netcom 9120 FlexBoom wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time. Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 715 Snom 370 + GN Netcom/Jabra 9120 FlexBoom Headset 359 359.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-gn-netcomjabra-9120-flexboom-headset-p-715.html http://www.voipon.co.uk/images/snom_370_netcom_flex.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9120 FlexBoom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it\'s mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 370 IP Telephone GN Netcom 9120 FlexBoom wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . GN Netcom 9120 Wireless Headset Features: Long range: up to 150m from the basestation. Long battery life: up to 8 hours of talk time - 1.5 hours recharge time Choice of three microphone booms: FlexBoom with noice-reduction; MicroBoom with noise-filtering - both perfect for filtering out unwanted background noise in louder environments - and MidiBoom with a standard omni-directional microphone for quieter office environments. All calls protected by 64-bit encryption 716 Snom 360 + GN Netcom/Jabra 9350 Wireless Headset 376 376.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-gn-netcomjabra-9350-wireless-headset-p-716.html http://www.voipon.co.uk/images/snom_360_netcom_9350.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9350 Midiboom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone GN Netcom 9350 wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . GN Netcom 9350 Wireless Headset : GN 9350 is the ultimate example of wireless office headset functionality. For use with both traditional desk telephones and IP telephony, the dual connection GN 9350 always meets your needs. GN 9350 makes everything sound better. State-of-the-art Digital Signal Processing (DSP) technology removes impurities from the incoming signal, enriches sound quality, and maintains a safe consistent volume level. The GN 9350's sleek boom arm with noise-cancelling microphone means your voice is transmitted clearly, even in noisy environments. When used with IP telephony applications, GN 9350 is the first wireless headset to provide wideband audio, with an astounding 6.8 kHz range - twice that of conventional telephony. With a wireless range of up to 100 metres, you can roam your office to your heart's content, whilst an optional nine-hour, user-replaceable battery means you can talk around the clock. Features: DSP and IntelliTone for superb sound quality and greater hearing protection. Future-proof investment thanks to built-in USB interface for PC-based IP telephone. LCD display for easy set-up of personal sound preferences. Noise-cancelling microphone for reduced background noise. Around-the-clock talk time with optional second battery. Light-weight design and multiple wearing styles -neckband included. Multi-unit conferencing capability.   717 Snom 320 + GN Netcom/Jabra 9350 Wireless Headset 265 265.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-gn-netcomjabra-9350-wireless-headset-p-717.html http://www.voipon.co.uk/images/snom_320_netcom_9350.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9350 Midiboom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone GN Netcom 9350 wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . GN Netcom 9350 Wireless Headset : GN 9350 is the ultimate example of wireless office headset functionality. For use with both traditional desk telephones and IP telephony, the dual connection GN 9350 always meets your needs. GN 9350 makes everything sound better. State-of-the-art Digital Signal Processing (DSP) technology removes impurities from the incoming signal, enriches sound quality, and maintains a safe consistent volume level. The GN 9350's sleek boom arm with noise-cancelling microphone means your voice is transmitted clearly, even in noisy environments. When used with IP telephony applications, GN 9350 is the first wireless headset to provide wideband audio, with an astounding 6.8 kHz range - twice that of conventional telephony. With a wireless range of up to 100 metres, you can roam your office to your heart's content, whilst an optional nine-hour, user-replaceable battery means you can talk around the clock. Features: DSP and IntelliTone for superb sound quality and greater hearing protection. Future-proof investment thanks to built-in USB interface for PC-based IP telephone. LCD display for easy set-up of personal sound preferences. Noise-cancelling microphone for reduced background noise. Around-the-clock talk time with optional second battery. Light-weight design and multiple wearing styles -neckband included. Multi-unit conferencing capability.   718 Snom 370 + GN Netcom/Jabra 9350 Wireless Headset 412 412.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-gn-netcomjabra-9350-wireless-headset-p-718.html http://www.voipon.co.uk/images/snom_370_netcom_9350.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The GN Netcom 9350 Midiboom Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 370 IP Telephone GN Netcom 9350 wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . GN Netcom 9350 Wireless Headset : GN 9350 is the ultimate example of wireless office headset functionality. For use with both traditional desk telephones and IP telephony, the dual connection GN 9350 always meets your needs. GN 9350 makes everything sound better. State-of-the-art Digital Signal Processing (DSP) technology removes impurities from the incoming signal, enriches sound quality, and maintains a safe consistent volume level. The GN 9350's sleek boom arm with noise-cancelling microphone means your voice is transmitted clearly, even in noisy environments. When used with IP telephony applications, GN 9350 is the first wireless headset to provide wideband audio, with an astounding 6.8 kHz range - twice that of conventional telephony. With a wireless range of up to 100 metres, you can roam your office to your heart's content, whilst an optional nine-hour, user-replaceable battery means you can talk around the clock. Features: DSP and IntelliTone for superb sound quality and greater hearing protection. Future-proof investment thanks to built-in USB interface for PC-based IP telephone. LCD display for easy set-up of personal sound preferences. Noise-cancelling microphone for reduced background noise. Around-the-clock talk time with optional second battery. Light-weight design and multiple wearing styles -neckband included. Multi-unit conferencing capability.   719 Snom 320 + Plantronics CS60 Wireless Headset 304 304.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-plantronics-cs60-wireless-headset-p-719.html http://www.voipon.co.uk/images/snom_320_plantronics_cs60_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . Plantronics CS60 Wireless Headset: The stylish CS60 wireless headset system gives excellent sound quality and hands free wireless freedom up to 100m from your desk (typical office up to 50m). The lightweight and stylish headset can be converted into 3 styles: over the ear, headband and our unique behind the neck design. CS60 has a talk time of up to 9 hours and you can even take calls when you are away from your desk. The IntelliStand™ feature senses when the headset has been removed from the base station and automatically takes the call and directs your call into the headset. Product Features: Totally wireless solution - no headset cables Fully convertible with 3 wearing styles -Unique behind the neck style; Ear-loop style; Over-the-head style. Up to 100m range - up to 50m in a typical office Up to 9 hrs talk time Fast battery recharge IntelliStand™ senses when the headset is removed or replaced in the base unit to automatically pick-up or end a call. Receive volume control Mute button Talk button DECT technology An intuitive flash demo showing how to set up and use the CS60 can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 720 Snom 360 + Plantronics CS60 Wireless Headset 333 333.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-plantronics-cs60-wireless-headset-p-720.html http://www.voipon.co.uk/images/snom_360_plantronics_cs60_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . Plantronics CS60 Wireless Headset: The stylish CS60 wireless headset system gives excellent sound quality and hands free wireless freedom up to 100m from your desk (typical office up to 50m). The lightweight and stylish headset can be converted into 3 styles: over the ear, headband and our unique behind the neck design. CS60 has a talk time of up to 9 hours and you can even take calls when you are away from your desk. The IntelliStand™ feature senses when the headset has been removed from the base station and automatically takes the call and directs your call into the headset. Product Features: Totally wireless solution - no headset cables Fully convertible with 3 wearing styles -Unique behind the neck style; Ear-loop style; Over-the-head style. Up to 100m range - up to 50m in a typical office Up to 9 hrs talk time Fast battery recharge IntelliStand™ senses when the headset is removed or replaced in the base unit to automatically pick-up or end a call. Receive volume control Mute button Talk button DECT technology An intuitive flash demo showing how to set up and use the CS60 can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 721 Snom 370 + Plantronics CS60 Wireless Headset 369 369.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-plantronics-cs60-wireless-headset-p-721.html http://www.voipon.co.uk/images/snom_370_plantronics_cs60_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 370 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . Plantronics CS60 Wireless Headset: The stylish CS60 wireless headset system gives excellent sound quality and hands free wireless freedom up to 100m from your desk (typical office up to 50m). The lightweight and stylish headset can be converted into 3 styles: over the ear, headband and our unique behind the neck design. CS60 has a talk time of up to 9 hours and you can even take calls when you are away from your desk. The IntelliStand™ feature senses when the headset has been removed from the base station and automatically takes the call and directs your call into the headset. Product Features: Totally wireless solution - no headset cables Fully convertible with 3 wearing styles -Unique behind the neck style; Ear-loop style; Over-the-head style. Up to 100m range - up to 50m in a typical office Up to 9 hrs talk time Fast battery recharge IntelliStand™ senses when the headset is removed or replaced in the base unit to automatically pick-up or end a call. Receive volume control Mute button Talk button DECT technology An intuitive flash demo showing how to set up and use the CS60 can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 722 Snom 360 + Plantronics CS70N Wireless Headset 348 348.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-360-plantronics-cs70n-wireless-headset-p-722.html http://www.voipon.co.uk/images/snom_360_plantron_cs70n_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 360 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 360 features can be found here . Plantronics CS70N Wireless Headset: The extremely stylish CS70N lets you talk on the phone whilst up to 50m from your desk, but is so discreet and comfortable you may forget you are wearing it. With a noise cancelling microphone to ensure you are heard clearly, it weighs just 22g but provides 5 hours’ talk-time. Thanks to the Snom Wireless Headset Adapter, you can answer incoming calls when away from your telephone, helping to avoid calls unnecessarily going to voicemail. DECT™ digital wireless technology ensures excellent audio quality and of course is fully compatible with the Snom range. Product Features: Stylish and discreet. Light (22g) and very comfortable. DECT™ technology for excellent sound quality. Noise-cancelling microphone to reduce background noise. Up to 5 hours talk-time, up to 28 hours on standby. Up to 100-metres range (50 metres in typical office). Remote call answering when away from your telephone. An intuitive flash demo showing how to set up and use the CS70N can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 723 Snom 320 + Plantronics CS70N Wireless Headset 319 319.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-320-plantronics-cs70n-wireless-headset-p-723.html http://www.voipon.co.uk/images/snom_320_plantron_cs70n_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 320 features can be found here . Plantronics CS70N Wireless Headset: The extremely stylish CS70N lets you talk on the phone whilst up to 50m from your desk, but is so discreet and comfortable you may forget you are wearing it. With a noise cancelling microphone to ensure you are heard clearly, it weighs just 22g but provides 5 hours’ talk-time. Thanks to the Snom Wireless Headset Adapter, you can answer incoming calls when away from your telephone, helping to avoid calls unnecessarily going to voicemail. DECT™ digital wireless technology ensures excellent audio quality and of course is fully compatible with the Snom range. Product Features: Stylish and discreet. Light (22g) and very comfortable. DECT™ technology for excellent sound quality. Noise-cancelling microphone to reduce background noise. Up to 5 hours talk-time, up to 28 hours on standby. Up to 100-metres range (50 metres in typical office). Remote call answering when away from your telephone. An intuitive flash demo showing how to set up and use the CS70N can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 724 Snom 370 + Plantronics CS70N Wireless Headset 384 384.00 Snom Snom Headset Bundles http://www.voipon.co.uk/snom-370-plantronics-cs70n-wireless-headset-p-724.html http://www.voipon.co.uk/images/snom_370_plantron_cs70n_s.jpg new Availability: In stock With this EHS-bundle you can take incoming calls directly to your ear, even when not situated at your workstation. The Plantronics CS60 Wireless Headset is a fantastic headset which offers wireless functionality up to a range of 100m from the designated base terminal. The headset has a built in rechargeable battery, simply plug it back into it's mains connected holder and it will charge while idle. Simply plug the base terminal of the headset into the Snom headset adapter, which plugs into the phone, and the headset is ready to use. (EHS = Electronic Hook Switch) Bundle Contents: Snom 320 IP Telephone Plantronics CS60 wireless headset Snom Wireless Headset Adapter Snom 370 features can be found here . Plantronics CS70N Wireless Headset: The extremely stylish CS70N lets you talk on the phone whilst up to 50m from your desk, but is so discreet and comfortable you may forget you are wearing it. With a noise cancelling microphone to ensure you are heard clearly, it weighs just 22g but provides 5 hours’ talk-time. Thanks to the Snom Wireless Headset Adapter, you can answer incoming calls when away from your telephone, helping to avoid calls unnecessarily going to voicemail. DECT™ digital wireless technology ensures excellent audio quality and of course is fully compatible with the Snom range. Product Features: Stylish and discreet. Light (22g) and very comfortable. DECT™ technology for excellent sound quality. Noise-cancelling microphone to reduce background noise. Up to 5 hours talk-time, up to 28 hours on standby. Up to 100-metres range (50 metres in typical office). Remote call answering when away from your telephone. An intuitive flash demo showing how to set up and use the CS70N can be found here . Note that the Snom Wireless headset adapter works as an EHS which removes the need for a handset lifter. 725 Rhino Ceros Mini Chassis, 2GB Flash Drive (CerosMini-2GB-ST) 569.0000 599.00 Rhino Rhino Ceros Mini Standalone http://www.voipon.co.uk/rhino-ceros-mini-chassis-2gb-flash-drive-cerosmini2gbst-p-725.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 2GB FLASH for Rhino PCI Cards (CerosMini-2GB-ST) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradeable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox,com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming! CentPBX (www.centpbx.com) - comig! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size :1U (1.75") tall 17.25” wide, 15 ” deep Form Factor: Rack mount, wall mount, table top Weight :20 pounds for base configuration New! Ceros Mini Display 5-buton keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 726 PoE Cable for Polycom 501 14.5000 14.50 Polycom VoIP Accessories http://www.voipon.co.uk/poe-cable-for-polycom-501-p-726.html http://www.voipon.co.uk/images/polycom_poe_cable-sml.jpg new Availability: In Stock PoE Cable for Polycom 501 727 Rhino Ceros Mini Chassis, 80GB HDD (CerosMini-80GB-ST) 599.0000 631.00 Rhino Rhino Ceros Mini Standalone http://www.voipon.co.uk/rhino-ceros-mini-chassis-80gb-hdd-cerosmini80gbst-p-727.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 80GB Hard Drive for Rhino PCI Cards (CerosMini-80GB-ST) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 728 Rhino Ceros Mini Chassis, 80GB HDD (CerosMini-80GB-ST) W/ RAID1 652.0000 686.00 Rhino Rhino Ceros Mini Standalone http://www.voipon.co.uk/rhino-ceros-mini-chassis-80gb-hdd-cerosmini80gbst-w-raid1-p-728.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 80GB Hard Drive for Rhino PCI Cards (CerosMini-80GB-ST) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15\" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5\" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75\") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 729 Rhino Ceros Mini Chassis, 160GB HDD (CerosMini-160GB-ST) 626.0000 659.00 Rhino Rhino Ceros Mini Standalone http://www.voipon.co.uk/rhino-ceros-mini-chassis-160gb-hdd-cerosmini160gbst-p-729.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 160GB Hard Drive for Rhino PCI Cards (CerosMini-160GB-ST) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 730 Rhino Ceros Mini Chassis, 160GB HDD (CerosMini-160GB-ST) W/ RAID 688.0000 724.00 Rhino Rhino Ceros Mini Standalone http://www.voipon.co.uk/rhino-ceros-mini-chassis-160gb-hdd-cerosmini160gbst-w-raid-p-730.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 160GB Hard Drive for Rhino PCI Cards (CerosMini-160GB-ST) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 731 Rhino Ceros Mini Chassis, 2GB Flash Drive (CerosMini-2GB) 569.0000 579.00 Rhino Rhino Ceros Mini with Card http://www.voipon.co.uk/rhino-ceros-mini-chassis-2gb-flash-drive-cerosmini2gb-p-731.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 2GB FLASH for Rhino PCI Cards (CerosMini-2GB) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradeable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox,com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming! CentPBX (www.centpbx.com) - comig! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size :1U (1.75") tall 17.25” wide, 15 ” deep Form Factor: Rack mount, wall mount, table top Weight :20 pounds for base configuration New! Ceros Mini Display 5-buton keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 732 Rhino Ceros Mini Chassis, 80GB HDD (CerosMini-80GB) 599.0000 611.00 Rhino Rhino Ceros Mini with Card http://www.voipon.co.uk/rhino-ceros-mini-chassis-80gb-hdd-cerosmini80gb-p-732.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 80GB Hard Drive for Rhino PCI Cards (CerosMini-80GB) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 733 Rhino Ceros Mini Chassis, 80GB HDD (CerosMini-80GB) W/ RAID1 652.0000 666.00 Rhino Rhino Ceros Mini with Card http://www.voipon.co.uk/rhino-ceros-mini-chassis-80gb-hdd-cerosmini80gb-w-raid1-p-733.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 80GB Hard Drive for Rhino PCI Cards (CerosMini-80GB) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 734 Rhino Ceros Mini Chassis, 160GB HDD (CerosMini-160GB) 626.0000 639.00 Rhino Rhino Ceros Mini with Card http://www.voipon.co.uk/rhino-ceros-mini-chassis-160gb-hdd-cerosmini160gb-p-734.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 160GB Hard Drive for Rhino PCI Cards (CerosMini-160GB) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 735 Rhino Ceros Mini Chassis, 160GB HDD (CerosMini-160GB) W/ RAID1 688.0000 704.00 Rhino Rhino Ceros Mini with Card http://www.voipon.co.uk/rhino-ceros-mini-chassis-160gb-hdd-cerosmini160gb-w-raid1-p-735.html http://www.voipon.co.uk/images/ceros_mini_1u.jpg new Availability: In Stock Providing reliable, flexible, and leading-edge solutions for a demanding industry. Managing your open source telecommunication needs has never been easier than with Rhino products. Ceros provides instant credibility to your Value Added Reseller (VAR) product line, by providing an unmarked PC chassis that is simply customized by editing a simple four line by 40 character text file. The text in the file is displayed on the LCD at all times for your customers to visually identify your own branded PBX product. Rhino Ceros Mini Chassis with 160GB Hard Drive for Rhino PCI Cards (CerosMini-160GB) with Intel 1G-ULV CPU & 512GB RAM, Preloaded with trixbox 2.2. Knowing that Rhino products are ready to perform right out of the box means that you can spend more time developing important customer relationships. Ceros comes preloaded with Liniux and Asterisk, along with all the necessary configurations for the specific installed Rhino hardware. All that is needed is to customize extensions and other outside system environment configurations. The Ceros chassis was designed with VARs in mind. The chassis comes with no external markings or silkscreening, and comes shipped in two cartons -- an outside carton that can be discarded so that a pristine, inside carton can be cleanly shipped to your end customer. Rhino Equipment Corp. offers you a complete line of low cost PCI plug-in cards including Single T1/E1, Dual T1/E1, Quad T1/E1, Quad FXO analog, Octal FXS/FXO and 24-port analog mixed mode analog interfaces. And don’t forget the full line of Rhino Channel Bank products, for large scale analog FXS or FXO applications. Rhino designed products are tough. In the rare case of trouble, our technical support staff is ready to give you the support you need, when you need it. Our 5-year, limited warranty means that you can be confident that Rhino will always work hard in your Open Source Telephony application. Standard Ceros Mini 1U Features Asterisk soft PBX tested Preloaded with your Asterisk distribution of choice Accepts all Rhino PCI cards. Full length cards have special anchor brackets to fully secure cards in the chassis Fully tested and integrated with all install Rhino hardware 1U tall, 17.25” wide x 15" deep external dimensions Fanless design Rubber bumper feet 19”mounting brackets - all included in box Six line Rhino serial OSLI and 5-button navigation keypad No external monitor or keyboard required 5-year limited warranty Base Configuration Intel 1G-ULV CPU 2GB IDE flash drive (upgradable to 2.5" hard drives, RAID1 also available) 512MB 667MHz DDR2 memory 180W power supply Two PCI slots Rhino SLI 6-line OSLI and 5-button keypad Video, mouse and keyboard ports Four USB ports, four serial port One GbE (LAN) and one 10/100 (WAN) integrated network ports Ceros Software New for 2008 is the ability to have Rhino install any available Asterisk-based Linux distribution. Here is a short list of the expected distributions that will be supported. RhinOSterisk (www.rhinosterisk.com) - available now! trixbox ce 2.4 (www.trixbox.com) - RC2 available now! Elastix (www.elastix.org) - 0.91 available now! PBX in a Flash (http://pbxinaflash.net) - coming soon! CentPBX (www.centpbx.com) - coming soon! And more to come! Ceros Upgrades: 80GB and 160GB hard drive RAID1 (two mirrored drives) 1GB DRAM (from 512MB) Others upon request Chassis Mechanical Data Size: 1U (1.75") tall 17.25” wide, 15” deep Form Factor: Rack mount, wall mount, table top Weight: 20 pounds for base configuration New! Ceros Mini Display 5-button keypad Reset and Power Off functions by pressing two buttons BMP file insertion allows for your Logo 6 lines for your company information Menu system allows for viewing important system information Fully graphical - layout graphic and text on the same display 736 Digium 8HPECLIC High Performance Echo Cancellation 5 5.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-8hpeclic-high-performance-echo-cancellation-p-736.html http://www.voipon.co.uk/images/no_image_available.jpg new Digium 8HPECLIC High Performance Echo Cancellation 737 Polycom Kirk 4020 DECT Handset 120.0000 133.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-4020-dect-handset-p-737.html http://www.voipon.co.uk/images/kirk_4020_small.jpg new Availability: In stock The KIRK 4020 handset is the entry level phone from the 4000 range. It is built for long term dependability and provides workplace mobility in a robust but stylish package. The 4020 is a strong, well designed and price competitive handset. It meets demands for free mobility and is built for long-term dependability in office environments. The 4020 can connect seamlessly with the KIRK Wireless Server 600v3. Upto 35 handsets can be connected to the system running as a single-cell solution or upto 1500 handsets when run as a multi-cell solution. Frequency bands: 1881.792-1897.344 MHz 1902.528-1918.080 MHz 2400 MHz Graphic display Alarm key - ready for alarm application CLIP (10 caller-ID presentations) * Internal/external ring pattern Volume control LED indication of incoming and unanswered calls Telephone book with room for 200 names and numbers Auto Login - Roaming between 10 different systems Silent mode (mute all sounds) Redial function (the last 10 numbers) Speed dial/Alarm key speed dial Programming pause Key lock Auto key lock Auto hook - when removed from charger Any key answer 9 different ringer tones and adjustable ringer volume Microphone mute Automatic Off-Hook (B-answer) Loudspeaker on B-answer 10 menu languages (UK, DE, FR, IT, ES, NL, PT, SE, DK, NO) Possibility for 1 customer specific language SMS function (Supported on KIRK systems only), stores 14 messages of 72 characters each Editing possibilities for start-up text User defined stand-by text R-key for transfer and special services Speech/stand by time > 16/150 hours ERP (Average Radiated Power): 10mW Temperature compensated charging Weight incl. battery: 130g Size (LxWxH): 148x50x28mm * = CLIP for internal calls - for external calls only if supported by PBX and external application 738 Polycom Kirk 4040 DECT Handset 171.0000 190.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-4040-dect-handset-p-738.html http://www.voipon.co.uk/images/kirk_4040_small.jpg new Availability: In stock The KIRK 4040 handset is the best selling phone from the 4000 range. It is built for long term dependability and provides workplace mobility in a robust but stylish package. The KIRK 4040 handset is a stong, well designed and full feature handset. It meets demands for easy mobility and is for built for long-term dependability in harsh environments. In order to meet industry standards, the KIRK 4040 is IP54 classified, which means that it is dust-protected and protected against splashing water. The 4040 can connect seamlessly with the KIRK Wireless Server 600v3. Upto 35 handsets can be connected to the system running as a single-cell solution or upto 1500 handsets when run as a multi-cell solution. Frequency bands: 1881.792-1897.344 MHz 1902.528-1918.080 MHz 2400 MHz Graphic display Alarm key - ready for alarm application Speed dial/Alarm key speed dial CLIP (10 caller-ID presentations) * Internal/external ring pattern Volume control LED indication of incoming and unanswered calls Telephone book with room for 200 names and numbers Auto Login - roaming between 10 different systems Silent mode (mute all sounds) Auto hook - when removed from charger Redial function (the last 10 numbers) Any key answer Programming pause Key lock Auto key lock 9 different ringer tone and adjustable ringer volume Headset alerting Microphone mute Headset jack Support for headset lead button Adjustable volume in headset Loud speaking/Handsfree Vibrator Automatic Off-Hook (B-answer) Loudspeaker on B-answer 10 menu languages (UK, DE, FR, IT, ES, NL, PT, SE, DK, NO) Possibility for 1 customer specific language SMS function (Supported on KIRK systems only), stores 14 messages of 72 characters each Editing possibilities for start-up text User defined stand-by text R-key for transfer and special services IP 54 classification: Dust protected/Splashing water Speech/stand by time > 16/150 hours ERP (Average Radiated Power): 10mW Temperature compensated charging Weight incl. battery: 130g Size (LxWxH): 148x50x28mm * = CLIP for internal calls - for external calls only if supported by PBX and external application 739 Polycom Kirk 4080 DECT Handset 419.0000 465.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-4080-dect-handset-p-739.html http://www.voipon.co.uk/images/kirk_4080_small.jpg new Availability: In stock The KIRK 4080 is the ultimate handset for environments with special conditions and regulations. The KIRK 4080 holds the same features as the KIRK 4040, but is furthermore IP 64 classified and ATEX and IEC approved according to European and international standards. The KIRK 4080 is targeted at potentially explosive work environments. Areas of application include, but are not restricted to, oil, gas and chemical production as well as dust-filled environments within the wood-processing, ingredients and food industries. The 4080 can connect seamlessly with the KIRK Wireless Server 600v3. Upto 35 handsets can be connected to the system running as a single-cell solution or upto 1500 handsets when run as a multi-cell solution. Frequency bands: 1881.792-1897.344 MHz Graphic display Alarm key - ready for alarm application Speed dial/Alarm key speed dial CLIP (10 caller-ID presentations) * Internal/external ring pattern Volume control LED indication of incoming and unanswered calls Telephone book with room for 200 names and numbers Auto login - roaming between 10 different systems Silent mode (mute all sounds) Redial function (the last 10 numbers) Programming pause Key lock Auto key lock 9 different ringer tones and adjustable ringer volume Microphone mute Headset jack Headset alerting Support for headset lead button Adjustable volume in headset Loud speaking/Handsfree Vibrator Automatic Off-Hook (B-answer) Loudspeaker on B-answer Any key answer Auto hook - when removed from charger 10 menu languages (UK, DE, FR, IT, ES, NL, PT, SE, DK, NO) Possibility for 1 customer specific language SMS function (Supported on KIRK systems only), stores 14 messages of 72 characters each Editing possibilities for start-up text R-key for transfer and special services IP 64 classification: Dust tight/Splashing water ATEX approved: II2G II3D T60°C EEx ib IIC T3 IEC approved: Ex ib IIC T3 Speech/stand by time > 16/150 hours ERP (Average Radiated Power): 10mW Temperature compensated charging Weight incl. battery: 130g Size (LxWxH): 148x50x28mm * = CLIP for internal calls - for external calls only if supported by PBX and external application 740 Polycom Kirk 5020 DECT Handset 171.0000 190.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-5020-dect-handset-p-740.html http://www.voipon.co.uk/images/kirk_5020_small.jpg new Availability: In stock The KIRK 5020 is an elegant handset positioned towards adminstrative environments across various vertical markets as it meets the needs for a wireless device that mobile workers within these market have, including: - Easy to use phonebook - Hands-free / headset / Call register with time stamp - Easy to recognise ring tone (internal vs. external) - Easy to use and to navigate The 5020 can connect seamlessly with the KIRK Wireless Server 600v3. Upto 35 handsets can be connected to the system running as a single-cell solution or upto 1500 handsets when run as a multi-cell solution. Frequency bands: 1881.792-1897.344 MHz 1902.528-1918.080 MHz TFT colour display (65.000 colours, 8 lines of text/icons) Li-ion battery 4 Way navigation key 2 Softkeys CLIP (40 caller-ID presentations) Date and time in display Internal/external ring pattern Volume control Telephone book with 250 name entries (4 numbers per name) Auto login - roaming between 10 different installations Silent mode (mutes all alerts/calls) Alerting on silent mode (choice from display flash, vibrator or short ring) Call list of incoming/missed/received (last 40 entries) Redial function from call list Speed dial Auto answer with different settings (after 1st ring/when lifted from charger/on headset/loud speaker on) 10 different ringer toners and adjustable ringer volume Key lock Auto key lock Vibrating alert Any key answer 11 menu languages (UK, FR, DE, ES, IT, NL, CZ, PO, DK, NO, SE) Headset connection Ring tone in headset Adjustable volume in headset Answer/end calls via headset button Microphone mute Speaker on auto-answer R-key for transfer and special services Adjustable alerting volume (low battery/low coverage/incoming message) Adjustable backlight delay (for max. battery conservation) Text messaging - max. 72 characters per message (system dependant) 10 user defined messaging templates Stores 20 messages Speech/stand by time: Up to 20/200 hours Temperature compensated charging Weight incl. battery: 110g Size (LxWxH): 146x48x19mm 2 types of chargers (w/wo USB 2.0 connection) 741 Polycom Kirk 3040 DECT Handset 158.0000 175.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-3040-dect-handset-p-741.html http://www.voipon.co.uk/images/kirk_3040_small.jpg new Availability: In stock KIRK telecom products are well-known throughout the world for their simple, classic design - and the KIRK 3040 is no exception . The elegance of the classic design, the user-friendly keys and the well-lit display make using the KIRK 3040 a real pleasure. The 3040 can connect seamlessly with the KIRK Wireless Server 600v3. Upto 35 handsets can be connected to the system running as a single-cell solution or upto 1500 handsets when run as a multi-cell solution. Frequency bands: 1881.792-1897.344 MHz 1902.528-1918.080 MHz 2401.2800-2481.1520 MHz Large alphanumeric display with backlight CLIP (10 caller-ID presentations) * Internal/external ring pattern Volume control LED indication of incoming and unanswered calls Telephone book with room for 80 numbers Vibrator Auto login - roaming between 10 different systems Silent mode (mute all sounds) Redial function (the last 10 numbers) Programming pause Key lock Auto key lock 9 different ringertones and adjustable ringer volume Microphone mute Headset jack Automatic Off-Hook (B-answer) 6 menu languages (UK, DE, FR, IT, ES, NL) Possibility for 1 customer specific language SMS function (Supported on KIRK systems only), stores 14 messages of 72 characters each Editing possibilities for start-up text R-key for call transfer and special services Speech/stand by time: > 16/150 hours ERP (Average Radiated Power): 10 mW Temperature compensated charging Weight incl. battery: 121g Size:143 x 48 x 26 mm * = CLIP for internal calls - for external calls only if supported by PBX and external application. 742 Polycom Kirk 600v3 Wireless IP Server 746.0000 829.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-600v3-wireless-ip-server-p-742.html http://www.voipon.co.uk/images/kirk_dect_600v3_small.jpg new Availability: In stock The KIRK Wireless Server 600v3 is a scalable solution capable of registering up to 1500 wireless users making it an excellent choice for small and medium-sized businesses across a wide range of vertical markets. Using an IP interface to the PBX, power over Ethernet and IP routing between the wireless servers, the KIRK Wireless Server 600v3 is a cost-effective system that allows for re-use of existing network infrastructure and components. The KIRK Wireless Server 600v3 can be deployed as either a single-cell or a multi-cell solution and can be adjusted to fit the exact size and needs of the individual customer. A single-cell version, consisting of one KIRK Wireless Server 600v3 and up to 6 KIRK Repeaters, can support up to 35 users. When there is a need for more than 35 users at a location, additional KIRK Wireless Server 600v3s can be installed in a multi-cell configuration. Up to 256 radio units (a mix of KIRK Wireless Server 600v3s and KIRK Repeaters) can be used to provide the necessary radio coverage. A multi-site solution can be further customized to include number of single-cell and multi-cell deployments depending on the size of each individual location. A KIRK Wireless Server 600v3 is installed directly on the LAN and must be managed as part of the corporate network. Features all protocols (Cisco Skinny, H.323 (innovaphone) and SIP : · Up to 1500 wireless users · Up to 256 KIRK Wireless Server/3 in one network cluster · Synchronisation over the air · Built-in redundancy · Power over LAN · One RJ45 for 100 Megabit connection to Ethernet · Browser configured - no additional SW needed · Codec G.711 and G.729 743 Polycom Kirk Repeater - Wall-Mounted - 2 Channel 129.0000 143.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-repeater-wallmounted-2-channel-p-743.html http://www.voipon.co.uk/images/kirk_dect_repeater_small.jpg new Availability: In stock The KIRK Repeater is a building block to be used to extend the coverage area in a KIRK solution. The KIRK Repeater does not increase the number of traffic channels, but merely provides a larger physical spreading of the traffic channels and hereby increases the coverage area established with the KIRK Base Stations. KIRK Repeaters are mainly used in areas with limited traffic. The KIRK Repeater is available with either 2 or 4 voice channels. It is wireless and does not need physical connection to the KIRK Wireless Server, making it very easy to install. This repeater is a 4-channel wall-mounted Repeater Wall or Ceiling-Mounted There are two different versions of the KIRK Repeater hardware; one for mounting on the wall and another for mounting on the ceiling. The ceiling-mounted KIRK Repeaters are especially useful for the hospitality and healthcare verticals in which there is often a need to assure radio coverage in long corridors. This is also the case for office environments within a wide range of vertical markets. Being able to place the KIRK Repeater on the ceiling makes it easier and less expensive to ascertain the needed coverage. Ceiling-mounting furthermore facilitates the work involved with assuring radio coverage in large factory buildings as they can be placed in the middle of the facility where the speech channels are needed the most, instead of along the walls. Repeater Jumps The KIRK Repeaters can be installed in a cascade style setup to provide coverage in a large area using only KIRK Repeaters. A maximum of three KIRK Repeaters can be placed on the string providing the possibility to assure maximum coverage of a large area without physical cabling. Repeater jumps should only be used to expand coverage in areas with limited traffic as the total area will have to share the same number of traffic channels available to the base station the repeaters are linked to (4 channels, unless a 2-channel repeater is used, in which case there will be a limit of 2 channels) Repeater jumps are possible with 2-channel repeaters in 1.8GHz, SAM and 2.4GHz and with 4-channel repeaters in 1.8GHz and SAM. 744 Polycom Kirk Repeater - Wall-Mounted - 4 Channel 155.0000 172.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-repeater-wallmounted-4-channel-p-744.html http://www.voipon.co.uk/images/kirk_dect_repeater_small.jpg new Availability: In stock The KIRK Repeater is a building block to be used to extend the coverage area in a KIRK solution. The KIRK Repeater does not increase the number of traffic channels, but merely provides a larger physical spreading of the traffic channels and hereby increases the coverage area established with the KIRK Base Stations. KIRK Repeaters are mainly used in areas with limited traffic. The KIRK Repeater is available with either 2 or 4 voice channels. It is wireless and does not need physical connection to the KIRK Wireless Server, making it very easy to install. This repeater is a 4-channel wall-mounted Repeater Wall or Ceiling-Mounted There are two different versions of the KIRK Repeater hardware; one for mounting on the wall and another for mounting on the ceiling. The ceiling-mounted KIRK Repeaters are especially useful for the hospitality and healthcare verticals in which there is often a need to assure radio coverage in long corridors. This is also the case for office environments within a wide range of vertical markets. Being able to place the KIRK Repeater on the ceiling makes it easier and less expensive to ascertain the needed coverage. Ceiling-mounting furthermore facilitates the work involved with assuring radio coverage in large factory buildings as they can be placed in the middle of the facility where the speech channels are needed the most, instead of along the walls. Repeater Jumps The KIRK Repeaters can be installed in a cascade style setup to provide coverage in a large area using only KIRK Repeaters. A maximum of three KIRK Repeaters can be placed on the string providing the possibility to assure maximum coverage of a large area without physical cabling. Repeater jumps should only be used to expand coverage in areas with limited traffic as the total area will have to share the same number of traffic channels available to the base station the repeaters are linked to (4 channels, unless a 2-channel repeater is used, in which case there will be a limit of 2 channels) Repeater jumps are possible with 2-channel repeaters in 1.8GHz, SAM and 2.4GHz and with 4-channel repeaters in 1.8GHz and SAM. 745 Polycom Kirk Repeater - Ceiling-Mounted - 4 Channel 167.0000 185.00 Polycom Kirk Polycom Kirk DECT IP600v3 http://www.voipon.co.uk/polycom-kirk-repeater-ceilingmounted-4-channel-p-745.html http://www.voipon.co.uk/images/kirk_dect_repeater_c_s.jpg new Availability: In stock The KIRK Repeater is a building block to be used to extend the coverage area in a KIRK solution. The KIRK Repeater does not increase the number of traffic channels, but merely provides a larger physical spreading of the traffic channels and hereby increases the coverage area established with the KIRK Base Stations. KIRK Repeaters are mainly used in areas with limited traffic. The KIRK Repeater is available with either 2 or 4 voice channels. It is wireless and does not need physical connection to the KIRK Wireless Server, making it very easy to install. This repeater is a 4-channel wall-mounted Repeater Wall- or Ceiling-Mounted There are two different versions of the KIRK Repeater hardware; one for mounting on the wall and another for mounting on the ceiling. The ceiling-mounted KIRK Repeaters are especially useful for the hospitality and healthcare verticals in which there is often a need to assure radio coverage in long corridors. This is also the case for office environments within a wide range of vertical markets. Being able to place the KIRK Repeater on the ceiling makes it easier and less expensive to ascertain the needed coverage. Ceiling-mounting furthermore facilitates the work involved with assuring radio coverage in large factory buildings as they can be placed in the middle of the facility where the speech channels are needed the most, instead of along the walls. Repeater Jumps The KIRK Repeaters can be installed in a cascade style setup to provide coverage in a large area using only KIRK Repeaters. A maximum of three KIRK Repeaters can be placed on the string providing the possibility to assure maximum coverage of a large area without physical cabling. Repeater jumps should only be used to expand coverage in areas with limited traffic as the total area will have to share the same number of traffic channels available to the base station the repeaters are linked to (4 channels, unless a 2-channel repeater is used, in which case there will be a limit of 2 channels) Repeater jumps are possible with 2-channel repeaters in 1.8GHz, SAM and 2.4GHz and with 4-channel repeaters in 1.8GHz and SAM. 746 Polycom SoundPoint IP560 185.0000 195.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundpoint-ip560-p-746.html http://www.voipon.co.uk/images/polycom_ip550-ip560_sml.jpg new Availability: In stock Polycom Soundpoint IP 560 - Future-proof SIP phone that combines GigE support & Polycom HD Voice experience The SoundPoint IP 560 desktop phone with GigE is a four-line SIP phone that delivers calls of unprecedented richness and clarity and supports a comprehensive range of cutting-edge features to futureproof your network infrastructure investment. The IP 560 is ideal for professionals and managers with demanding collaborative communication needs. Higher Productivity, Less Fatigue The SoundPoint IP 560 desktop phone features Polycom’s revolutionary HD Voice technology, which brings life-like richness and clarity to calls. Polycom HD Voice technology ecompasses wideband audio for more than twice the voice clarity and Polycom’s patented Acoustic Clarity Technology for crystal-clear, noise- and echo-free sound with a best-in-class system design for high-fidelity, faithful voice reproduction. The Polycom SoundPoint IP 560 desktop phone is designed and engineered to make installation, configuration, and upgrading as simple and efficient as possible. The phone’s built-in IEEE 802.3af PoE circuitry and dual-port Gigabit Ethernet switch allow flexible deployment options and savings on cabling cost. The IP560 supports remote, zero-touch provisioning and upgrades from a variety of servers as well as boot and call server redundancy to ensure reliable, uninterrupted performance. Polycom IP 560 - Key Features & Benefits 320 x 160-pixel backlit grayscale graphical LCD Up to 4 lines with up to 2 concurrent calls per line Polycom HD Voice technology delivers life-like voice quality for each audio path - the handset, the hands-free speakerphone, and the optional headset. Adaptive jitter buffers Packet loss concealment Acoustic echo cancellation Background noise suppression Multifaceted productivity enhancements - Allowing for fast data and application transmission between the network and user’s computer - Delivering unparalleled, lifelike audio experience, powered by Polycom HD Voice - XHTML micro-browser for Web applications - Shared call/bridged line appearance - Busy lamp field - Presence and buddy list 747 Atcom IP04 4-port Asterisk IP PBX - 4 FXO 275.0000 289.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-4-fxo-p-747.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 4 x FXO modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 748 Digium Wildcard TE121 PCI Express ISDN PRI Card 264.0000 278.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te121-pci-express-isdn-pri-card-p-748.html http://www.voipon.co.uk/images/digium_te121_small.jpg new Availability: In Stock High Performance Digium TE121P PCI-Express VoIP Card The Digium TE121 is a high-performance, cost effective digital telephony interface card with the power to create a seamless network, interconnecting traditional telephony systems with emerging Voice over IP technologies. digium & Asterisk® The TE121 is a single span, selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) card. The card utilizes Digium's VoiceBus™ technology. VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts. Improving upon the prior TE120P which does not offer expansion capabilities, the TE121 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode. Bundled with the VPMADT032, the product SKU is TE121B. The TE121 supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice, routed directly to the Asterisk Open Source PBX, and 12 to data, handled by the underlying Linux operating system, thus eliminating the need for an external router. The TE121 works in both 3.3V and 5V PCI slots by auto detecting the slot's voltage. By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can easily connect multiple PCs equipped with the TE121 and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE121 cards to each individual PC. Add additional cards as you need them for your expanding applications. The TE121 supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Using this card in concert with Digium's Asterisk® software, standard PC hardware, and the Linux® OS, you can upgrade your PBX to a sophisticated telephony environment capable of supporting both voice and data channels. 749 Digium Wildcard TE122P PCI ISDN PRI Card 264.0000 278.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te122p-pci-isdn-pri-card-p-749.html http://www.voipon.co.uk/images/digium_te122_small.jpg new Availability: In Stock High Performance Digium TE122 PCI VoIP Card The Digium TE122 is a high-performance, cost effective digital telephony interface card with the power to create a seamless network, interconnecting traditional telephony systems with emerging Voice over IP technologies. digium & Asterisk® The TE122 is a single span, selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) card. The card utilizes Digium's VoiceBus™ technology. VoiceBus technology allows the TE122 to use an industry standard bus-mastering PCI interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts. Improving upon the prior TE120P which does not offer expansion capabilities, the TE122 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode. Bundled with the VPMADT032, the product SKU is TE122B. The TE122 supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice, routed directly to the Asterisk Open Source PBX, and 12 to data, handled by the underlying Linux operating system, thus eliminating the need for an external router. The TE122 works in both 3.3V and 5V PCI slots by auto detecting the slot's voltage. By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can easily connect multiple PCs equipped with the TE122 and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE122 cards to each individual PC. Add additional cards as you need them for your expanding applications. T he TE122 supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Using this card in concert with Digium's Asterisk® software, standard PC hardware, and the Linux® OS, you can upgrade your PBX to a sophisticated telephony environment capable of supporting both voice and data channels. 750 OpenVox D210E PCI Express ISDN PRI Card 285.0000 356.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-d210e-pci-express-isdn-pri-card-p-750.html http://www.voipon.co.uk/images/openvox_d210e_isdn_card_s.jpg new Availability: In stock The D210E is the latest hardware addition to the Openvox array of digital cards. It is a PCI-Express version of the D210P. The D210E supports E1, T1 and J1 environments and is selectable on a per-card or per-port basis. This feature enables signaling translation between E1 and T1 equipment and allows inexpensive T1 channel banks to connect with E1 circuits. With the improved I/O speed, the card reduces CPU usage and increased card density per server. D210P is fully compatible with Asterisk™ application. Also, the open source driver supports an API interface for custom application development. The D210E support industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported, also included are advanced call features. Other Features includes: Support for PCI-Express 1.0. High Performance PCI Bus Master with up to 2Kbytes FIFO Easy to install: Use wct4xxp driver included in original Zaptel without any patch. RoHS compliant Certificates: CE and FCC Two year warranty 751 VoiSmart Fax Server with 2 channel licences (FX-0700-01-HW02C) 1340.0000 1489.00 VoiSmart VoiSmart Fax Server http://www.voipon.co.uk/voismart-fax-server-with-2-channel-licences-fx070001hw02c-p-751.html http://www.voipon.co.uk/images/voismart_fax_server_s.jpg new Availability: In Stock The VoiSmart FaxServer is the easiest way to manage, deliver, store, record faxes directly from the end users\' PC. Through a simple web interface, each user can send or receive fax, decide whether to get them via mail or not and define how and where to store them. VoiSmart FaxServer allows any organization to save money otherwise spent in cartridge, paper, expensive hardware and IT maintenance, moreover it allows increasing efficiency and productivity due to the fact that users don\'t have to waste time moving to and from fax machines and waiting to the fax to be sent. VoiSmart FaxServer is a powerful solution for confidential documents, as each fax is addressed directly to the receiver and it cannot be missed or changed, but if some faxes need to be shared within an office, they can be addressed to specific groups. VoiSmart FaxServer can be used for direct marketing purposes by scheduling delivery time and number of attempts. The FDR functionality shows the result of the fax activity detailing the statistics and the outcome of each sending. VoiSmart FaxServer is a powerful solution, which can manage up to 120 simultaneous faxes (depending on the hardware), with a scalable structure that easily adds fax capability to new applications and infrastructure according to the organization growth. Moreover it\'s fully integrated with the Voip Technology: using the T38 real time fax protocol, it guarantees an outstanding level of communication and the cost saving features of the Voip network. T.38 protocol can relay fax even on noisy and jittery lines, where standard G.711 fax passthrough fails. This hardware platform supports up to 50 channels, licences for 2 channels are included in this model. (FX-0700-01-HW02C) Additional licences can be purchased here . Main Highlights • Intuitive and user friendly interface • Full access through Web browser. • Multiple languages availabile • Unified environment customizable at single user level • Easy setup • Full access to Fax server set-ups and configuration • Full FDR • FDR data export in CSV format 752 VoiSmart Fax Server with 50 channel licences (FX-0700-02-HW50C) 4815.0000 5345.00 VoiSmart VoiSmart Fax Server http://www.voipon.co.uk/voismart-fax-server-with-50-channel-licences-fx070002hw50c-p-752.html http://www.voipon.co.uk/images/voismart_fax_server_s.jpg new Availability: In Stock The VoiSmart FaxServer is the easiest way to manage, deliver, store, record faxes directly from the end users' PC. Through a simple web interface, each user can send or receive fax, decide whether to get them via mail or not and define how and where to store them. VoiSmart FaxServer allows any organization to save money otherwise spent in cartridge, paper, expensive hardware and IT maintenance, moreover it allows increasing efficiency and productivity due to the fact that users don't have to waste time moving to and from fax machines and waiting to the fax to be sent. VoiSmart FaxServer is a powerful solution for confidential documents, as each fax is addressed directly to the receiver and it cannot be missed or changed, but if some faxes need to be shared within an office, they can be addressed to specific groups. VoiSmart FaxServer can be used for direct marketing purposes by scheduling delivery time and number of attempts. The FDR functionality shows the result of the fax activity detailing the statistics and the outcome of each sending. VoiSmart FaxServer is a powerful solution, which can manage up to 120 simultaneous faxes (depending on the hardware), with a scalable structure that easily adds fax capability to new applications and infrastructure according to the organization growth. Moreover it's fully integrated with the Voip Technology: using the T38 real time fax protocol, it guarantees an outstanding level of communication and the cost saving features of the Voip network. T.38 protocol can relay fax even on noisy and jittery lines, where standard G.711 fax passthrough fails. This hardware platform supports up to 50 channels, licences for 50 channels are included in this model. (FX-0700-02-HW50C) Main Highlights • Intuitive and user friendly interface • Full access through Web browser. • Multiple languages availabile • Unified environment customizable at single user level • Easy setup • Full access to Fax server set-ups and configuration • Full FDR • FDR data export in CSV format 753 VoiSmart Fax Server 2 channel licence upgrade (FX-0700-03-SW02C) 275.0000 305.00 VoiSmart VoiSmart Fax Server http://www.voipon.co.uk/voismart-fax-server-2-channel-licence-upgrade-fx070003sw02c-p-753.html http://www.voipon.co.uk/images/voismart_fax_cd_s.jpg new Availability: In Stock This is a 2 channel licence upgrade for the VoiSmart Fax Server - hardware or software platform. Either platform supports upto 50 channels.   754 VoiSmart Fax Software Version - 2 channels (FX-0700-03-SW02C) 689.0000 765.00 VoiSmart VoiSmart Fax Server http://www.voipon.co.uk/voismart-fax-software-version-2-channels-fx070003sw02c-p-754.html http://www.voipon.co.uk/images/voismart_fax_cd_s.jpg new Availability: In Stock This is a software platform that allows a PC to act as a Fax Server. By installing the software onto a PC, users will have access to a GUI which allows them to configure settings that would otherwise be found only in the hardware VoiSmart Fax Server. The VoiSmart FaxServer is the easiest way to manage, deliver, store, record faxes directly from the end users' PC. Through a simple web interface, each user can send or receive fax, decide whether to get them via mail or not and define how and where to store them. VoiSmart FaxServer allows any organization to save money otherwise spent in cartridge, paper, expensive hardware and IT maintenance, moreover it allows increasing efficiency and productivity due to the fact that users don't have to waste time moving to and from fax machines and waiting to the fax to be sent. VoiSmart FaxServer is a powerful solution for confidential documents, as each fax is addressed directly to the receiver and it cannot be missed or changed, but if some faxes need to be shared within an office, they can be addressed to specific groups. VoiSmart FaxServer can be used for direct marketing purposes by scheduling delivery time and number of attempts. The FDR functionality shows the result of the fax activity detailing the statistics and the outcome of each sending. VoiSmart FaxServer is a powerful solution, which can manage up to 120 simultaneous faxes (depending on the hardware), with a scalable structure that easily adds fax capability to new applications and infrastructure according to the organization growth. Moreover it's fully integrated with the Voip Technology: using the T38 real time fax protocol, it guarantees an outstanding level of communication and the cost saving features of the Voip network. T.38 protocol can relay fax even on noisy and jittery lines, where standard G.711 fax passthrough fails. This software platform supports up to 50 channels, licences for 2 channels are included in this model. (FX-0700-03-SW02C) Additional licences can be purchased here . Main Highlights • Intuitive and user friendly interface • Full access through Web browser. • Multiple languages availabile • Unified environment customizable at single user level • Easy setup • Full access to Fax server set-ups and configuration • Full FDR • FDR data export in CSV format 755 VoiSmart Traffic Shaper (VS-0800-01-TSV) 895.0000 995.00 VoiSmart VoiSmart Traffic Shaper http://www.voipon.co.uk/voismart-traffic-shaper-vs080001tsv-p-755.html http://www.voipon.co.uk/images/voismart_traffic-shaper_s.jpg new Availability: In Stock VoiSmart Traffic Shaper is a revolutionary device, which will allow quality calls to be performed even when a network is overloaded and VoIP calls could not usually be established. The result of a joint research project between VoiSmart and Politecnico di Milano, based on a traffic estimation engine and an innovative traffic control algorithm, VoiSmart Traffic Shaper is a Level 2 Bridge for data and a SIP transparent proxy for VoIP calls. It is totally transparent to traffic, and ready to be used: it allocates resources as a function of the measured traffic and of the voice call requests. It also controls a selective attribution of the data (non-voice) traffic, giving the counterpart an indication of the call drop (if any). It can be also used as a measurement tool of the effective bandwidth supplied by your internet provider. Dynamic Bandwidth Management VoiSmart Traffic Shaper performs a real time analysis of traffic, allocating, as a function of the detected codec, the necessary bandwidth to the new IP call. Call Admission Control inhibits any new call whenever the minimum quality levels cannot be guaranteed. Nonvoice traffic is slowed down, being assigned a lower priority class, but not interrupted. If no IP calls are present, the full bandwidth is dynamically reassigned to the normal data traffic. Moreover, it performs bandwidth shaping also within non-voice data traffic (Web, E-mail, SSH...) Main Features • SIP Firewall supported • All-in-one: no HW or SW to be added • Easy to install and to integrate • Astonishing Quality of Service • Level 2 Bridge • Embedded Linux OS • Dynamical evaluation of available bandwidth • Web interface • Borrowing as a function of detected Codec and within non-voice traffic • Multiple Codecs supported • VoIP optimized CPU • Use from 640 kbps (ADSL) upwards • Advanced Call Admission Control: inhibits calls to be issued if adequate band is not available • Graphic statistics 756 UK VoIP Geographic Number Renewal 19.0000 21.99 Incoming Numbers http://www.voipon.co.uk/uk-voip-geographic-number-renewal-p-756.html http://www.voipon.co.uk/images/0102.gif new Please purchase this product only when your Geographic number expries. 757 Grandstream GXW-4024 Analog FXS Gateway 445 445.00 Grandstream Grandstream Gateways http://www.voipon.co.uk/grandstream-gxw4024-analog-fxs-gateway-p-757.html http://www.voipon.co.uk/images/grandstream_gxw-4024_s.jpg new Availability: In Stock The GXW4024 is a high density SIP based analog telephone VoIP gateway that is fully interoperable with leading IP-PBX and Softswitch systems. It features 24 telephone ports, rich telephony functionalities, superb voice quality, easy provisioning, flexible dial plans and advanced security protection. The GXW4024 gateway enables small to medium sized businesses to create a cost-effective hybrid IP and analog telephone system - thus enjoy the benefits of VoIP communications while preserving investment on existing analog phones and traditional PBX systems. The GXW FXS series, comes in three different port configurations: 4 port , 8 port , and 24 port. Main Benefits: Simple and flexible configuration options as a VoIP-FXS Gateway IP enabler for analogue phones, faxes and legacy PBX systems Secure and easy management using Web browser, automated provisioning tools, and integrated IVR High quality voice and fax communication with significant cost savings Features 24 FXS ports with PSTN lifeline One 10M/100Mbps network port Multiple SIP server profiles (2 per system) and independent account per port Supports audio codecs: G.711, G.723, G.726, G.729 A/B/E, iLBC T.38 Fax G.168 Echo Cancellation 758 Digium TDM808EF 2 Quad Port FXO Module with EC 470.0000 640.00 Digium Digium TDM800P http://www.voipon.co.uk/digium-tdm808ef-2-quad-port-fxo-module-with-ec-p-758.html http://www.voipon.co.uk/images/digium_tdm800p.jpg new Availability: In Stock The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium\'s VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC. The TDM800P provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single PCI bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the TDM800P reduces part complexity, cable clutter, and points of failure. The TDM800P contains two module banks. Each bank supports up to four analog interfaces. The module banks may be filled either with one standard Digium® quad analog module (FXS - S400M, FXO - X400M), or up to two standard Digium® single analog modules (FXS- S110M, FXO - X100M) enabling the creation of any combination of ports The TDM800P may be used in conjunction with Digium\'s High Performance Echo Canceler (HPEC), a commercial and toll quality hybrid echo cancellation solution. Digium\'s HPEC provides 16ms to 128ms of selectable near-end ITU G.168 compliant echo cancellation in software. Using this card in concert with Digium\'s Asterisk® software, standard PC hardware, and the Linux® OS, you can create SME or SOHO telephony environments capable of satisfying the needs of small or medium business applications at an industry-leading price. 759 Digium TDM401B - 1 FXO PCI Card 104.0000 114.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm401b-1-fxo-pci-card-p-759.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM401B is a TDM410 card with 1 FXO module attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM401B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM401B or other Digium analog interface cards. The TDM401B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 760 Digium TDM401E - 1 FXO PCI Card with Echo Cancellation VPMADT032 212.0000 238.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm401e-1-fxo-pci-card-with-echo-cancellation-vpmadt032-p-760.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM401E is a TDM410 card with 1 FXO module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM401E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM401E or other Digium analog interface cards. The TDM401E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 761 Digium AEX801B - 1 FXO PCI Express Card 174.0000 189.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex801b-1-fxo-pci-express-card-p-761.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX801B is an AEX800 card with 1 x FXO module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX801B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 762 Digium AEX801E - 1 FXO PCI Express Card with Echo Cancellation 282.0000 312.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex801e-1-fxo-pci-express-card-with-echo-cancellation-p-762.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX801E is an AEX800 card with 1 x FXO module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX801E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 763 Digium AEX2401E - 1 FXO PCI Express Card with Echo Cancellation 366.0000 405.00 Digium Digium AEX2400 PCI Express http://www.voipon.co.uk/digium-aex2401e-1-fxo-pci-express-card-with-echo-cancellation-p-763.html http://www.voipon.co.uk/images/digium_aex2400_sml.jpg new Availability: In stock The Digium AEX2401E is an AEX2400 card with hardware echo cancellation and 1 x Quad FXO module. The Digium® AEX2400 is a full-length PCI-Express 1.0-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to six quad port station or trunk modules for a total of 24 lines. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the AEX2401E eliminates the requirement for separate channel bank and T1 interface cards, with industry-leading performance and price. The quad trunk and quad station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional AEX2400 interface cards. The image shown is an AEX2400 with 4 x Quad FXO modules, 4 Quad FXS modules, and the hardware echo cancellation module. 764 Digium AEX2401B - 1 x Quad FXO PCI Express Card 255.0000 281.00 Digium Digium AEX2400 PCI Express http://www.voipon.co.uk/digium-aex2401b-1-x-quad-fxo-pci-express-card-p-764.html http://www.voipon.co.uk/images/digium_aex2400_sml.jpg new Availability: In stock The Digium AEX2401B is an AEX2400 card with 1 x Quad FXO module. The Digium® AEX2400 is a full-length PCI-Express 1.0-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to six quad port station or trunk modules for a total of 24 lines. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the AEX2401B eliminates the requirement for separate channel bank and T1 interface cards, with industry-leading performance and price. The quad trunk and quad station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional AEX2400 interface cards. The image shown is an AEX2400 with 4 x Quad FXO modules, 4 Quad FXS modules, and the hardware echo cancellation module. 765 Digium Wildcard TE121B PCI Express ISDN PRI Card with Echo Cancellation 365.0000 385.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te121b-pci-express-isdn-pri-card-with-echo-cancellation-p-765.html http://www.voipon.co.uk/images/digium_te121_small.jpg new Availability: In Stock High Performance Digium TE121V PCI-Express VoIP Card with 32 channel Hardware Echo Cancellation The Digium TE121 is a high-performance, cost effective digital telephony interface card with the power to create a seamless network, interconnecting traditional telephony systems with emerging Voice over IP technologies. digium & Asterisk® The TE121 is a single span, selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) card. The card utilizes Digium's VoiceBus™ technology. VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts. Improving upon the prior TE120P which does not offer expansion capabilities, the TE121 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode. Bundled with the VPMADT032, the product SKU is TE121B. The TE121 supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice, routed directly to the Asterisk Open Source PBX, and 12 to data, handled by the underlying Linux operating system, thus eliminating the need for an external router. The TE121 works in both 3.3V and 5V PCI slots by auto detecting the slot's voltage. By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can easily connect multiple PCs equipped with the TE121 and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE121 cards to each individual PC. Add additional cards as you need them for your expanding applications. The TE121 supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Using this card in concert with Digium's Asterisk® software, standard PC hardware, and the Linux® OS, you can upgrade your PBX to a sophisticated telephony environment capable of supporting both voice and data channels. 766 Digium Wildcard TE122B PCI ISDN PRI Card with Echo Cancellation 365.0000 385.00 Digium Digium Digital (PRI) Cards http://www.voipon.co.uk/digium-wildcard-te122b-pci-isdn-pri-card-with-echo-cancellation-p-766.html http://www.voipon.co.uk/images/digium_te122_small.jpg new Availability: In Stock High Performance Digium TE122 PCI VoIP Card with 32 Channel Hardware Echo Cancellation The Digium TE122 is a high-performance, cost effective digital telephony interface card with the power to create a seamless network, interconnecting traditional telephony systems with emerging Voice over IP technologies. digium & Asterisk® The TE122 is a single span, selectable T1 (24-channel), E1 (32-channel), or J1 (24-channel) card. The card utilizes Digium's VoiceBus™ technology. VoiceBus technology allows the TE122 to use an industry standard bus-mastering PCI interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts. Improving upon the prior TE120P which does not offer expansion capabilities, the TE122 may be combined with Digium's VPMADT032 echo cancellation module to deliver 128ms of echo cancellation across 30 channels in E1 mode or 24 channels in T1 mode. Bundled with the VPMADT032, the product SKU is TE122B. The TE122 supports both voice and data modes on its single span. For example, the card can support 12 channels dedicated to voice, routed directly to the Asterisk Open Source PBX, and 12 to data, handled by the underlying Linux operating system, thus eliminating the need for an external router. The TE122 works in both 3.3V and 5V PCI slots by auto detecting the slot's voltage. By utilizing our TDMoE (TDM over Ethernet) technology, an exclusive Digium process, you can easily connect multiple PCs equipped with the TE122 and achieve voice quality on par with single PBX implementations. Scalability for this product is derived from adding multiple TE122 cards to each individual PC. Add additional cards as you need them for your expanding applications. T he TE122 supports industry standard telephony and data protocols, including both RBS and Primary Rate ISDN (PRI) protocol families for voice and PPP, Cisco HDLC, and Frame Relay data modes. The board drives both line-side and trunk-side interfaces, including call features. Using this card in concert with Digium's Asterisk® software, standard PC hardware, and the Linux® OS, you can upgrade your PBX to a sophisticated telephony environment capable of supporting both voice and data channels. 767 Digium TDM402B - 2 FXO PCI Card 124.0000 137.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm402b-2-fxo-pci-card-p-767.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM402B is a TDM410 card with 2 FXO modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM402B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM402B or other Digium analog interface cards. The TDM402B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 768 Digium TDM402E - 2 FXO PCI Card with Echo Cancellation VPMADT032 232.0000 261.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm402e-2-fxo-pci-card-with-echo-cancellation-vpmadt032-p-768.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM402E is a TDM410 card with 2 FXO modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM402E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM402E or other Digium analog interface cards. The TDM402E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 769 Digium TDM403B - 3 FXO PCI Card 144.0000 160.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm403b-3-fxo-pci-card-p-769.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM403B is a TDM410 card with 3 FXO modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and a standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM403B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM403B or other Digium analog interface cards. The TDM403B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 770 Digium TDM403E - 3 FXO PCI Card with Echo Cancellation VPMADT032 252.0000 284.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm403e-3-fxo-pci-card-with-echo-cancellation-vpmadt032-p-770.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM403E is a TDM410 card with 3 FXO modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and a standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM403E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM403E or other Digium analog interface cards. The TDM403E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 771 Digium TDM404B - 4 FXO PCI Card 164.0000 183.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm404b-4-fxo-pci-card-p-771.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM404B is a TDM410 card with 4 FXO modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM404B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM404B or other Digium analog interface cards. The TDM404B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 772 Digium TDM404E - 4 FXO PCI Card with Echo Cancellation VPMADT032 272.0000 307.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm404e-4-fxo-pci-card-with-echo-cancellation-vpmadt032-p-772.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM404E is a TDM410 card with 4 FXO modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM404E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM404E or other Digium analog interface cards. The TDM404E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 773 Digium TDM410B - 1 FXS PCI Card 102.0000 112.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm410b-1-fxs-pci-card-p-773.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM410B is a TDM410 card with 1 FXS module attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and a standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM410B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM410B or other Digium analog interface cards. The TDM410B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 774 Digium TDM410E - 1 FXS PCI Card with Echo Cancellation VPMADT032 210.0000 236.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm410e-1-fxs-pci-card-with-echo-cancellation-vpmadt032-p-774.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM410E is a TDM410 card with 1 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM410E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM410E or other Digium analog interface cards. The TDM410E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 775 Digium TDM420B - 2 FXS PCI Card 121.0000 134.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm420b-2-fxs-pci-card-p-775.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM420B is a TDM410 card with 2 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM420B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM420B or other Digium analog interface cards. The TDM420B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 776 Digium TDM420E - 2 FXS PCI Card with Echo Cancellation VPMADT032 229.0000 257.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm420e-2-fxs-pci-card-with-echo-cancellation-vpmadt032-p-776.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM420E is a TDM410 card with 2 FXS modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM420E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM420E or other Digium analog interface cards. The TDM420E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 777 Digium TDM430B - 3 FXS PCI Card 140.0000 155.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm430b-3-fxs-pci-card-p-777.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM430B is a TDM410 card with 3 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM430B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM430B or other Digium analog interface cards. The TDM430B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 778 Digium TDM430E - 3 FXS PCI Card with Echo Cancellation VPMADT032 248.0000 279.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm430e-3-fxs-pci-card-with-echo-cancellation-vpmadt032-p-778.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM430E is a TDM410 card with 3 FXS modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM430E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM430E or other Digium analog interface cards. The TDM430E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 779 Digium TDM440B - 4 FXS PCI Card 159.0000 177.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm440b-4-fxs-pci-card-p-779.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM440B is a TDM410 card with 4 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM440B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM440B or other Digium analog interface cards. The TDM440B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 780 Digium TDM440E - 4 FXS PCI Card with Echo Cancellation VPMADT032 267.0000 301.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm440e-4-fxs-pci-card-with-echo-cancellation-vpmadt032-p-780.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM440E is a TDM410 card with 4 FXS modules attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM440E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM440E or other Digium analog interface cards. The TDM440E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 781 Digium TDM411B - 1 FXO 1 FXS PCI Card 122.0000 135.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm411b-1-fxo-1-fxs-pci-card-p-781.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM411B is a TDM410 card with 1 FXO and 1 FXS module attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM411B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM411B or other Digium analog interface cards. The TDM411B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 782 Digium TDM411E - 1 FXO 1 FXS PCI Card with Echo Cancellation VPMADT032 230.0000 259.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm411e-1-fxo-1-fxs-pci-card-with-echo-cancellation-vpmadt032-p-782.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM411E is a TDM410 card with 1 FXO and 1 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM411E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM411E or other Digium analog interface cards. The TDM411E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 783 Digium TDM412B - 2 FXO 1 FXS PCI Card 143.0000 159.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm412b-2-fxo-1-fxs-pci-card-p-783.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM412B is a TDM410 card with 2 FXO and 1 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM412B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM412B or other Digium analog interface cards. The TDM412B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 784 Digium TDM412E - 2 FXO 1 FXS PCI Card with Echo Cancellation VPMADT032 251.0000 282.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm412e-2-fxo-1-fxs-pci-card-with-echo-cancellation-vpmadt032-p-784.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM412E is a TDM410 card with 2 FXO and 1 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM412E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM412E or other Digium analog interface cards. The TDM412E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 785 Digium TDM413B - 3 FXO 1 FXS PCI Card 163.0000 182.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm413b-3-fxo-1-fxs-pci-card-p-785.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM413B is a TDM410 card with 3 FXO and 1 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM413B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM413B or other Digium analog interface cards. The TDM413B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 786 Digium TDM413E - 3 FXO 1 FXS PCI Card with Echo Cancellation VPMADT032 271.0000 305.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm413e-3-fxo-1-fxs-pci-card-with-echo-cancellation-vpmadt032-p-786.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM413E is a TDM410 card with 3 FXO and 1 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM413E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM413E or other Digium analog interface cards. The TDM413E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 787 Digium TDM421B - 1 FXO 2 FXS PCI Card 141.0000 157.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm421b-1-fxo-2-fxs-pci-card-p-787.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM421B is a TDM410 card with 1 FXO and 2 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM421B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM421B or other Digium analog interface cards. The TDM421B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 788 Digium TDM421E - 1 FXO 2 FXS PCI Card with Echo Cancellation VPMADT032 249.0000 281.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm421e-1-fxo-2-fxs-pci-card-with-echo-cancellation-vpmadt032-p-788.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM421E is a TDM410 card with 1 FXO and 2 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM421E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM421E or other Digium analog interface cards. The TDM421E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 789 Digium TDM431B - 1 FXO 3 FXS PCI Card 160.0000 178.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm431b-1-fxo-3-fxs-pci-card-p-789.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM431B is a TDM410 card with 1 FXO and 3 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM431B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM431B or other Digium analog interface cards. The TDM431B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 790 Digium TDM431E - 1 FXO 3 FXS PCI Card with Echo Cancellation VPMADT032 268.0000 302.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm431e-1-fxo-3-fxs-pci-card-with-echo-cancellation-vpmadt032-p-790.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM431E is a TDM410 card with 1 FXO and 3 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM431E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM431E or other Digium analog interface cards. The TDM431E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 791 Digium TDM422B - 2 FXO 2 FXS PCI Card 161.0000 180.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm422b-2-fxo-2-fxs-pci-card-p-791.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM422B is a TDM410 card with 2 FXO and 2 FXS modules attached and no echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM422B eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM422B or other Digium analog interface cards. The TDM422B provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium's hardware-based echo cancellation modules. The image shown is a TDM410E card with 2 FXS and 2 FXO modules and the optional echo cancellaton module. 792 Digium TDM422E - 2 FXO 2 FXS PCI Card with Echo Cancellation VPMADT032 269.0000 304.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-tdm422e-2-fxo-2-fxs-pci-card-with-echo-cancellation-vpmadt032-p-792.html http://www.voipon.co.uk/images/digium_410p_sml.jpg new Availability: In stock The TDM422E is a TDM410 card with 2 FXO and 2 FXS module attached and echo cancellation module ( VPMADT032) . Used in conjunction with Asterisk and standard PC hardware, one can create a telephony system that includes all of the cutting edge features of a high-quality business PBX. Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus technology, the TDM422E eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM422E or other Digium analog interface cards. The TDM422E provides the same functional capacity as the TDM400P, but improves significantly upon its system compatibility and provides the capability of connecting to Digium\'s hardware-based echo cancellation modules. 793 Digium VPMADT032 - 128ms Hardware Echo Cancellation Module 114.0000 143.00 Digium Digium TDM410P http://www.voipon.co.uk/digium-vpmadt032-128ms-hardware-echo-cancellation-module-p-793.html http://www.voipon.co.uk/images/VPMADT032-echo-sml.jpg new Availability: In stock The VPMADT032 is a 128ms per-channel echo cancellation module for use with Digium telephony interface cards. It is compatible with the following Digium cards: TDM410 TDM800P AEX800 TDM2400P AEX2400 This module, which offers 128ms (1024 taps) of echo cancellation across all of its channels, improves upon the older VPM100M echo cancellation module, that only offered 32ms of echo cancellation. The VPMADT032 provides the same toll-quality G.168 compliant echo cancellation found in Digium's HPEC software-based commercial echo canceller. The VPMADT032 is immune from any system CPU spikes that might otherwise affect a software-based solution. The option to purchase the standalone module is ideal should a user already have one of the above cards without the module, as it provides them with the ability to upgrade to full hardware echo cancellation without having to invest in new interface cards. 794 OpenVox A400E01 - 1 FXO PCI Express Card 96.0000 120.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e01-1-fxo-pci-express-card-p-794.html http://www.voipon.co.uk/images/openvox_a400e01_sml.jpg new Availability: In stock The A400E01 is an OpenVox A400E bundled with 1 FXO module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E01 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E01 is the PCI-Express version of the A400P. 795 OpenVox A400E02 - 2 FXO PCI Express Card 123.0000 153.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e02-2-fxo-pci-express-card-p-795.html http://www.voipon.co.uk/images/openvox_a400e02_sml.jpg new Availability: In stock The A400E02 is an OpenVox A400E bundled with 2 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E02 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E02 is the PCI-Express version of the A400P. 796 OpenVox A400E03 - 3 FXO PCI Express Card 148.0000 187.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e03-3-fxo-pci-express-card-p-796.html http://www.voipon.co.uk/images/openvox_a400e03_sml.jpg new Availability: In stock The A400E03 is an OpenVox A400E bundled with 3 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E03 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E03 is the PCI-Express version of the A400P. 797 OpenVox A400E04 - 4 FXO PCI Express Card 170.0000 214.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e04-4-fxo-pci-express-card-p-797.html http://www.voipon.co.uk/images/openvox_a400e04_sml.jpg new Availability: In stock The A400E04 is an OpenVox A400E bundled with 4 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E04 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E04 is the PCI-Express version of the A400P. 798 OpenVox A400E10 - 1 FXS PCI Express Card 86.7000 108.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e10-1-fxs-pci-express-card-p-798.html http://www.voipon.co.uk/images/openvox_a400e10_sml.jpg new Availability: In stock The A400E10 is an OpenVox A400E bundled with 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E10 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E10 is the PCI-Express version of the A400P. 799 OpenVox A400E20 - 2 FXS PCI Express Card 111.0000 138.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e20-2-fxs-pci-express-card-p-799.html http://www.voipon.co.uk/images/openvox_a400e20_sml.jpg new Availability: In stock The A400E20 is an OpenVox A400E bundled with 2 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E20 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E20 is the PCI-Express version of the A400P. 800 OpenVox A400E30 - 3 FXS PCI Express Card 133.0000 166.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e30-3-fxs-pci-express-card-p-800.html http://www.voipon.co.uk/images/openvox_a400e30_sml.jpg new Availability: In stock The A400E30 is an OpenVox A400E bundled with 3 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E30 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E30 is the PCI-Express version of the A400P. 801 OpenVox A400E40 - 4 FXS PCI Express Card 158.0000 196.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e40-4-fxs-pci-express-card-p-801.html http://www.voipon.co.uk/images/openvox_a400e40_sml.jpg new Availability: In stock The A400E40 is an OpenVox A400E bundled with 4 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E40 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E40 is the PCI-Express version of the A400P. 802 OpenVox A400E11 - 1 FXO 1 FXS PCI Express Card 119.0000 148.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e11-1-fxo-1-fxs-pci-express-card-p-802.html http://www.voipon.co.uk/images/openvox_a400e11_sml.jpg new Availability: In stock The A400E11 is an OpenVox A400E bundled with 1 FXO and 1 FXS module. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E11 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E11 is the PCI-Express version of the A400P. 803 OpenVox A400E22 - 2 FXO 2 FXS PCI Express Card 166.0000 206.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e22-2-fxo-2-fxs-pci-express-card-p-803.html http://www.voipon.co.uk/images/openvox_a400e22_sml.jpg new Availability: In stock The A400E22 is an OpenVox A400E bundled with 2 FXO and 2 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E22 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E22 is the PCI-Express version of the A400P. 804 OpenVox A400E21 - 1 FXO 2 FXS PCI Express Card 140.0000 176.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e21-1-fxo-2-fxs-pci-express-card-p-804.html http://www.voipon.co.uk/images/openvox_a400e21_sml.jpg new Availability: In stock The A400E21 is an OpenVox A400E bundled with 1 FXO and 2 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E21 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E21 is the PCI-Express version of the A400P. 805 OpenVox A400E31 - 1 FXO 3 FXS PCI Express Card 163.0000 204.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e31-1-fxo-3-fxs-pci-express-card-p-805.html http://www.voipon.co.uk/images/openvox_a400e31_sml.jpg new Availability: In stock The A400E31 is an OpenVox A400E bundled with 1 FXO and 3 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E31 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E31 is the PCI-Express version of the A400P. 806 OpenVox A400E12 - 2 FXO 1 FXS PCI Express Card 143.0000 180.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e12-2-fxo-1-fxs-pci-express-card-p-806.html http://www.voipon.co.uk/images/openvox_a400e12_sml.jpg new Availability: In stock The A400E12 is an OpenVox A400E bundled with 2 FXO and 1 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E12 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E12 is the PCI-Express version of the A400P. 807 OpenVox A400E13 - 3 FXO 1 FXS PCI Express Card 170.0000 213.00 OpenVox OpenVox A400E PCI-Express http://www.voipon.co.uk/openvox-a400e13-3-fxo-1-fxs-pci-express-card-p-807.html http://www.voipon.co.uk/images/openvox_a400e13_sml.jpg new Availability: In stock The A400E13 is an OpenVox A400E bundled with 3 FXO and 1 FXS modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. OpenVox A400E13 is designed to be fully compatible with the TDM400P Asterisk Analog Telephony card. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400E card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400E card to terminate analog telephones. Key Features and benefits: Scalable: Just add additional cards to extend the system. Easy to use: A400E is full software and hardware compatible with TDM400P. Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. PCI-Express 1.0 compatible. Modules are Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400E13 is the PCI-Express version of the A400P. 808 Hitachi DMP330 (DMP-330) Dual Mode Handset 152.0000 159.00 Hitachi Hitachi IP Telephone http://www.voipon.co.uk/hitachi-dmp330-dmp330-dual-mode-handset-p-808.html http://www.voipon.co.uk/images/hitachi_dmp330_sml.jpg new Availability: In Stock The Hitachi DMP330 is a Dual mode handset supporting both GSM (900/1800/1900) and WLAN (WiFi) networks. The handset can be used like a GSM mobile phone in normal circumstances, and when located in a WiFi hot spot, at home or in the office, the DMP330 can switch to WLAN mode and be used to make VoIP over WiFi calls. The DMP330 is multimedia feature rich as well as practical, featuring a 1.3 megapixel camera, MP3 player, video player, internet browser and more. All this weighing in at under 90g. Key features: 1.8" LCD screen (175*220 resolution) Tri-band GSM VoIP over WiFi (SIP) WLAN (802.11b/g) MP3 Player (MP3, AAC, AAC+) Video Player (MPEG4, H.263, 3GP) 1.3 Megapixel Camera 1000 name phone book Speakerphone Java MIDP 2.0 Built in 64MB memory Upgradable memory slot (MicroSD) Internet Browser E-mail (POP3/ IMAP4) Bluetooth WPA2 Security 64 Polyphonic ringtones/ Vibration alert Long battery life: Talk time: 240 mins (GSM mode) 160 mins (WiFi mode) Standby time: 180 hours (GSM mode) 150 hours (WiFi mode) 809 3CX Windows IP PBX Free edition - 8 Simultaneous calls (3CXPSFRE) 0 0.00 3CX 3CX Windows IP PBX http://www.voipon.co.uk/3cx-windows-ip-pbx-free-edition-8-simultaneous-calls-3cxpsfre-p-809.html http://www.voipon.co.uk/images/3cx-free.jpg new Availability: In stock The free version of 3CX (3CXPSFRE) can be downloaded here . Evolve your communications with 3CX Phone System for Windows - an IP Phone System that completely replaces your proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. 3CX Phone System is far less expensive than a traditional PBX and can reduce call costs substantially by using a VOIP service provider. Its web-based administration makes phone system management easy. 3CX Phone System eliminates the phone wiring network and allows users to hot desk simply by taking their phone. Key Features: Complete phone system - Provides call switching, routing & queueing Purchase cost dramatically lower than a traditional hardware PBX Scaleable - Unlimited extensions and phone lines. No proprietary expansion modules needed! Web based configuration & status indication - Easy phone system management! Unified messaging - Receive voice mail via e-mail Auto attendant (e.g. 1 for sales, 2 for support, etc.) Reduce long distance and inter office call costs No more expensive proprietary system phones - Use standard SIP phones Eliminate the phone wiring and make moving offices easier All of the commercial versions have the same features, however, depending on which version you choose, more simultaneous calls are supported. FREE Edition Small Business Edition Professional Edition Enterprise Edition Extensions & Calls Number of Extensions Unlimited Unlimited Unlimited Unlimited Number of Simultaneous Calls Up to 8 Up to 8 Up to 16 Up to 32 Compare the free and commercial versions of 3CX. FREE Edition Commercial Editions General Phone System Features Personal Call History Call Logging Call Reporting Blind Call Transfer Attended Call Transfer Call Forward on Busy Call Forward on No Answer Call Parking   Call Pickup   Call Routing (DID) Caller ID Conference Calling (if supported by phone) Auto Attendant / Digital Receptionist Voice Mail Music on Hold Ring Groups Hunt Groups MWI - Message Waiting Indicator   Call Queueing   BLF Status Updates   E-mail & live chat support   Management Features Web-based management console Real time web-based system status Integrated web server, no web server required Phone provisioning   Unified Communications & Mobility Features Public SIP ID for extensions Supports Exchange 2007 UM   Branch Office Integration Sends voice mail via email Fax   3CX bridge: connect branch offices   Call recording   Remote configuration of extension options   3CX VOIP Client Compact Windows system tray applet Included mini VPN – firewall friendly Shows status of other extensions   Shows incoming calls Shows caller ID Shows personal call history Ability to place calls Ability to divert calls to voice mail Ability to transfer calls TAPI driver for integration with Outlook Shows callers in queue   Devices Supports SIP hardware phones Supports SIP software phones Supports popular SIP VOIP Gateways Supports popular SIP VOIP Providers SIP Trunking support Scalability Integrated enterprise database (PostgreSQL) Ability to run on a Virtual Machine Restore and backup to compressed XML file High performance web server (Apache) Supports Windows clustering Codecs G729   G711 (a law and u law) GSM Speex Iblc 810 3CX Windows IP PBX Small Business Edition - 8 Simultaneous calls (3CXPSSB) 265 265.00 3CX 3CX Windows IP PBX http://www.voipon.co.uk/3cx-windows-ip-pbx-small-business-edition-8-simultaneous-calls-3cxpssb-p-810.html http://www.voipon.co.uk/images/3cx-small_bus.jpg new Availability: In stock This is the Small Business Edition of 3CX (3CXPSSB). It supports upto eight simultaneous calls and all of the commercial edition features listed below. Evolve your communications with 3CX Phone System for Windows - an IP Phone System that completely replaces your proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. 3CX Phone System is far less expensive than a traditional PBX and can reduce call costs substantially by using a VOIP service provider. Its web-based administration makes phone system management easy. 3CX Phone System eliminates the phone wiring network and allows users to hot desk simply by taking their phone. Key Features: Complete phone system - Provides call switching, routing & queueing Purchase cost dramatically lower than a traditional hardware PBX Scaleable - Unlimited extensions and phone lines. No proprietary expansion modules needed! Web based configuration & status indication - Easy phone system management! Unified messaging - Receive voice mail via e-mail Auto attendant (e.g. 1 for sales, 2 for support, etc.) Reduce long distance and inter office call costs No more expensive proprietary system phones - Use standard SIP phones Eliminate the phone wiring and make moving offices easier All of the commercial versions have the same features, however, depending on which version you choose, more simultaneous calls are supported. FREE Edition Small Business Edition Professional Edition Enterprise Edition Extensions & Calls Number of Extensions Unlimited Unlimited Unlimited Unlimited Number of Simultaneous Calls Up to 8 Up to 8 Up to 16 Up to 32 Compare the free and commercial versions of 3CX. FREE Edition Commercial Editions General Phone System Features Personal Call History Call Logging Call Reporting Blind Call Transfer Attended Call Transfer Call Forward on Busy Call Forward on No Answer Call Parking   Call Pickup   Call Routing (DID) Caller ID Conference Calling (if supported by phone) Auto Attendant / Digital Receptionist Voice Mail Music on Hold Ring Groups Hunt Groups MWI - Message Waiting Indicator   Call Queueing   BLF Status Updates   E-mail & live chat support   Management Features Web-based management console Real time web-based system status Integrated web server, no web server required Phone provisioning   Unified Communications & Mobility Features Public SIP ID for extensions Supports Exchange 2007 UM   Branch Office Integration Sends voice mail via email Fax   3CX bridge: connect branch offices   Call recording   Remote configuration of extension options   3CX VOIP Client Compact Windows system tray applet Included mini VPN – firewall friendly Shows status of other extensions   Shows incoming calls Shows caller ID Shows personal call history Ability to place calls Ability to divert calls to voice mail Ability to transfer calls TAPI driver for integration with Outlook Shows callers in queue   Devices Supports SIP hardware phones Supports SIP software phones Supports popular SIP VOIP Gateways Supports popular SIP VOIP Providers SIP Trunking support Scalability Integrated enterprise database (PostgreSQL) Ability to run on a Virtual Machine Restore and backup to compressed XML file High performance web server (Apache) Supports Windows clustering Codecs G729   G711 (a law and u law) GSM Speex Iblc 811 3CX Windows IP PBX Professional Edition - 16 Simultaneous calls (3CXPSPRO) 565 565.00 3CX 3CX Windows IP PBX http://www.voipon.co.uk/3cx-windows-ip-pbx-professional-edition-16-simultaneous-calls-3cxpspro-p-811.html http://www.voipon.co.uk/images/3cx-pro.jpg new Availability: In stock This is the Professional Edition of 3CX (3CXPSPRO). It supports upto 16 simultaneous calls and all of the commercial edition features listed below. Evolve your communications with 3CX Phone System for Windows - an IP Phone System that completely replaces your proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. 3CX Phone System is far less expensive than a traditional PBX and can reduce call costs substantially by using a VOIP service provider. Its web-based administration makes phone system management easy. 3CX Phone System eliminates the phone wiring network and allows users to hot desk simply by taking their phone. Key Features: Complete phone system - Provides call switching, routing & queueing Purchase cost dramatically lower than a traditional hardware PBX Scaleable - Unlimited extensions and phone lines. No proprietary expansion modules needed! Web based configuration & status indication - Easy phone system management! Unified messaging - Receive voice mail via e-mail Auto attendant (e.g. 1 for sales, 2 for support, etc.) Reduce long distance and inter office call costs No more expensive proprietary system phones - Use standard SIP phones Eliminate the phone wiring and make moving offices easier All of the commercial versions have the same features, however, depending on which version you choose, more simultaneous calls are supported. FREE Edition Small Business Edition Professional Edition Enterprise Edition Extensions & Calls Number of Extensions Unlimited Unlimited Unlimited Unlimited Number of Simultaneous Calls Up to 8 Up to 8 Up to 16 Up to 32 Compare the free and commercial versions of 3CX. FREE Edition Commercial Editions General Phone System Features Personal Call History Call Logging Call Reporting Blind Call Transfer Attended Call Transfer Call Forward on Busy Call Forward on No Answer Call Parking   Call Pickup   Call Routing (DID) Caller ID Conference Calling (if supported by phone) Auto Attendant / Digital Receptionist Voice Mail Music on Hold Ring Groups Hunt Groups MWI - Message Waiting Indicator   Call Queueing   BLF Status Updates   E-mail & live chat support   Management Features Web-based management console Real time web-based system status Integrated web server, no web server required Phone provisioning   Unified Communications & Mobility Features Public SIP ID for extensions Supports Exchange 2007 UM   Branch Office Integration Sends voice mail via email Fax   3CX bridge: connect branch offices   Call recording   Remote configuration of extension options   3CX VOIP Client Compact Windows system tray applet Included mini VPN – firewall friendly Shows status of other extensions   Shows incoming calls Shows caller ID Shows personal call history Ability to place calls Ability to divert calls to voice mail Ability to transfer calls TAPI driver for integration with Outlook Shows callers in queue   Devices Supports SIP hardware phones Supports SIP software phones Supports popular SIP VOIP Gateways Supports popular SIP VOIP Providers SIP Trunking support Scalability Integrated enterprise database (PostgreSQL) Ability to run on a Virtual Machine Restore and backup to compressed XML file High performance web server (Apache) Supports Windows clustering Codecs G729   G711 (a law and u law) GSM Speex Iblc 812 3CX Windows IP PBX Enterprise Edition - 32 Simultaneous calls (3CXPSENT) 821 821.00 3CX 3CX Windows IP PBX http://www.voipon.co.uk/3cx-windows-ip-pbx-enterprise-edition-32-simultaneous-calls-3cxpsent-p-812.html http://www.voipon.co.uk/images/3cx-enterprise.jpg new Availability: In stock This is the Enterprise Edition of 3CX (3CXPSENT). It supports upto 32 simultaneous calls and all of the commercial edition features listed below. Evolve your communications with 3CX Phone System for Windows - an IP Phone System that completely replaces your proprietary PBX, supports standard SIP soft/hard phones, VOIP services and traditional PSTN phone lines. 3CX Phone System is far less expensive than a traditional PBX and can reduce call costs substantially by using a VOIP service provider. Its web-based administration makes phone system management easy. 3CX Phone System eliminates the phone wiring network and allows users to hot desk simply by taking their phone. Key Features: Complete phone system - Provides call switching, routing & queueing Purchase cost dramatically lower than a traditional hardware PBX Scaleable - Unlimited extensions and phone lines. No proprietary expansion modules needed! Web based configuration & status indication - Easy phone system management! Unified messaging - Receive voice mail via e-mail Auto attendant (e.g. 1 for sales, 2 for support, etc.) Reduce long distance and inter office call costs No more expensive proprietary system phones - Use standard SIP phones Eliminate the phone wiring and make moving offices easier All of the commercial versions have the same features, however, depending on which version you choose, more simultaneous calls are supported. FREE Edition Small Business Edition Professional Edition Enterprise Edition Extensions & Calls Number of Extensions Unlimited Unlimited Unlimited Unlimited Number of Simultaneous Calls Up to 8 Up to 8 Up to 16 Up to 32 Compare the free and commercial versions of 3CX. FREE Edition Commercial Editions General Phone System Features Personal Call History Call Logging Call Reporting Blind Call Transfer Attended Call Transfer Call Forward on Busy Call Forward on No Answer Call Parking   Call Pickup   Call Routing (DID) Caller ID Conference Calling (if supported by phone) Auto Attendant / Digital Receptionist Voice Mail Music on Hold Ring Groups Hunt Groups MWI - Message Waiting Indicator   Call Queueing   BLF Status Updates   E-mail & live chat support   Management Features Web-based management console Real time web-based system status Integrated web server, no web server required Phone provisioning   Unified Communications & Mobility Features Public SIP ID for extensions Supports Exchange 2007 UM   Branch Office Integration Sends voice mail via email Fax   3CX bridge: connect branch offices   Call recording   Remote configuration of extension options   3CX VOIP Client Compact Windows system tray applet Included mini VPN – firewall friendly Shows status of other extensions   Shows incoming calls Shows caller ID Shows personal call history Ability to place calls Ability to divert calls to voice mail Ability to transfer calls TAPI driver for integration with Outlook Shows callers in queue   Devices Supports SIP hardware phones Supports SIP software phones Supports popular SIP VOIP Gateways Supports popular SIP VOIP Providers SIP Trunking support Scalability Integrated enterprise database (PostgreSQL) Ability to run on a Virtual Machine Restore and backup to compressed XML file High performance web server (Apache) Supports Windows clustering Codecs G729   G711 (a law and u law) GSM Speex Iblc 813 Polycom CX200 OCS IP Phone 64.6000 68.00 Polycom Polycom OCS IP Phones http://www.voipon.co.uk/polycom-cx200-ocs-ip-phone-p-813.html http://www.voipon.co.uk/images/polycom_cx200-sml.jpg new Availability: In stock The CX range from Polycom are designed for Microsoft Office Communication Server. The CX200 intergrates seamlessly with Microsoft Office Communicator 2007 to provide users with outstanding high definition wideband audio quality. With its sleek, high quality handset and full duplex, hands-free speakerphone, the Polycom CX200 ensures crystal-clear conversations without any feedback or echo. With the convenience of USB plug-and-play connectivity, and no extra power cables to carry - thanks to power over USB, its clear the CX200 is built for convenience and simplicity. Key features: Unified communication - seamlessly integrates and supports the capabilities of Microsoft Office Communication Server 2007. Freedom - Clear one-on-one conversations, or high quality speakerphone modes. High definition - Incredible quality high definition wideband audio for the best voice call quality around. Presence - Enhanced presence indicator allows monitoring of presence status to allow efficient routing of calls. Plug-and-play - The CX200 is connected and powered via a USB port on the PC, minimalising cable clutter. Monitor - Voicemail, message waiting, and calls forwarded indicators keep users up to date. Audio Enhancements - Automatic Gain Control, Dynamic Noice Reduction, and Acoustic Echo Cancellation all help keep call quality at a maximum. 814 Polycom CX700 OCS IP Phone 274.0000 297.00 Polycom Polycom OCS IP Phones http://www.voipon.co.uk/polycom-cx700-ocs-ip-phone-p-814.html http://www.voipon.co.uk/images/polycom_cx700-sml.jpg new Availability: In stock The CX range from Polycom are designed for Microsoft Office Communication Server. The CX700 integrates seamlessly with Microsoft Office Communicator 2007 to provide users with outstanding high definition wideband audio quality. It comes embedded with an Office Communicator 2007 client - displayed on the 5.7" touch-screen TFT LCD (320x240 resolution). With its sleek, high quality handset; full duplex, hands-free speakerphone; as well as the headset mode, the Polycom CX700 ensures crystal-clear conversations without any feedback or echo. Key features: Integrated Office Commnicator 2007 client - thanks to the 5.7" touch-screen LCD, all features of Microsoft Office Communicator can be reached through the phone - eliminating the need for a PC to access the advanced VoIP features of Office Communicator. An on-screen keyboard can be used to type, call presence monitored, and much more. Easy Call Management - Monitor and manage calls through the Microsoft Windows CE-based touch-screen interface. Click-to-call with name-based calling from a user's contact list or call logs. Enhanced incoming call information lets users quickly decide how to handle a caller. Personal Configuration - The built in biometric fingerprint reader or on-screen keyboard can be used to set up individual user communications preferences. High definition - Incredible quality high definition wideband audio for the best voice call quality around. Presence - On screen presence details of the users contact list - ability to set the presence of the phone via the touch-screen. A large LED is used to diplay current presence status. Flexibility - Access voicemail, wireless headset can be plugged in via standard RJ-9, connect to a computer via the Ethernet port, and more. Audio Enhancements - Automatic Gain Control, Dynamic Noice Reduction, and Acoustic Echo Cancellation all help keep call quality at a maximum.   815 AC Adaptor for RFP 32 16.38 16.38 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/ac-adaptor-for-rfp-32-p-815.html http://www.voipon.co.uk/images/no_image_available.jpg new Availability: In stock AC Adaptor for RFP 32 816 Aastra DECT Activation CD 16.84 16.84 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/aastra-dect-activation-cd-p-816.html http://www.voipon.co.uk/images/aastra_install_cd_sml.jpg new Availability: In stock Aastra OMM Activation CD 817 Polycom CX100 OCS IP Speakerphone 59.8500 63.00 Polycom Polycom OCS IP Phones http://www.voipon.co.uk/polycom-cx100-ocs-ip-speakerphone-p-817.html http://www.voipon.co.uk/images/polycom_cx100-sml.jpg new Availability: In stock The CX range by Polycom is designed for Microsoft Office Communication Server. The CX100 is a full-duplex speakerphone ideal for handset and headset free conference calls. The CX100 is designed to be ultra portable, it is connected and powered by USB, and the USB cable can be stored inside the unit for transport - making the device portable and convenient. The CX100 is tightly integrated with Microsoft Office Communicator - delivering funtionality for handsfree computer calling, optimised to provide a seamless unified communication experience. Key features: The high-fidelity speaker ensures that audio quality is kept to a maximum, and no echo is heard - and doubles as a high quality solution for CD quality music and presentation audio. For private conversations, just plug in stereo headphones to the headphone jack. Two stereo microphones make sure that audio quality is always high and every word is heard clearly - a full 360 degrees of pickup ensures that everyone is heard during a conference. Convenient buttons and integrated functionality with Microsoft Office Communicator 2007 enable you to pick up and hang up calls, control volume, and mute the microphones on the active call. Plug-n-Play capability of USB ensures no driver downloads are required. Ultimate portability - USB cable stored inside unit for convenience and reduced cable clutter. 818 Digium AEX802B - 2 FXO PCI Express Card 193.0000 212.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex802b-2-fxo-pci-express-card-p-818.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX802B is an AEX800 card with 2 x FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX802B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 819 Digium AEX803E - 3 FXO PCI Express Card with Echo Cancellation VPMADT032 324.0000 358.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex803e-3-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-819.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX803E is an AEX800 card with 3 x FXO modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX803E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 820 Digium AEX803B - 3 FXO PCI Express Card 214.0000 235.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex803b-3-fxo-pci-express-card-p-820.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX803B is an AEX800 card with 3 x FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX803B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 821 Digium AEX802E - 2 FXO PCI Express Card with Echo Cancellation VPMADT032 303.0000 335.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex802e-2-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-821.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX802E is an AEX800 card with 2 x FXO modules attached as well as the echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX802E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 822 Digium AEX8S4B - 4 FXO PCI Express Card 234.1900 258.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex8s4b-4-fxo-pci-express-card-p-822.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX8S4B is an AEX800 card with 4 x FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX8S4B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 823 Digium AEX8S4E - 4 FXO PCI Express Card with Echo Cancellation VPMADT032 345.0000 382.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex8s4e-4-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-823.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX8S4E is an AEX800 card with 4 x FXO modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX8S4E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 824 Patton SmartNode 4554 2 port (4-channel) ISDN BRI Gateway 240.0000 259.00 Patton Inalp Patton ISDN BRI Gateways http://www.voipon.co.uk/patton-smartnode-4554-2-port-4channel-isdn-bri-gateway-p-824.html http://www.voipon.co.uk/images/patton_4554_sml.jpg new Availability: In Stock SmartNode 2-port BRI VoIP GW-Router, 1x10/100baseTX WAN, integrated 10/100bTX LAN switch, H.323 or SIP, External UI Power. The SmartNode 4552 enables integration of ISDN network users into their local phone service, a remote PBX and the data/VPN network through a single tightly integrated access device. Connecting to the local ISDN Phone as well as the local PSTN port, Patton's SessionRouter® technology links any standard ISDN telephone or PBX to VoIP while still connecting to legacy PSTN services. Built-in Lifeline support ensures phone connection to the PSTN in the event of failure. Gateway functions use standard CODECs such as G.723 and G.729 as well as industry standard SIP, H.323 and MGCP/IUA to ensure seamless connection and compatibility for all voice services. Broadband network connectivity integrates with any fixed IP, DHCP or PPPoE service. An integrated 10/100 Ethernet LAN switch, with advanced routing features such as NAT, Firewall/ACL, DynDNS as well as optional IPSec VPN, fulfills the requirements of demanding network users. Quality of Service (QoS) features complete the offering with advanced voice prioritization and traffic management. Patton\'s patent-pending DownStreamQoS® ensures voice without interruptions even over best-effort Internet connections. Features & Benefits 4-channel VoIP Media Gateway and router with dual ISDN BRI/S0 ports with NT and TE support. SessionRouter® with enhanced circuit-switched call routing allows user programmable call handling based on hunt groups, caller/called ID, and time of day. Pass local PSTN call through without packet conversion. SIP, H.323, and MGCP All SmartNodes support industry standard call-control signalling with standards-based compatibility Fast-Ethernet WAN and integrated 4-port 10/100 LAN switch with auto MDI-X. Access router with NAT, Firewall, PPPoE, DHCP, and DynDNS Quality of Service Advanced adaptive traffic management and shaping for maximum voice quality. Voice and data prioritization and DownStreamQoS® Built-in Web based management, SNMP, Command Line Interface and Auto-Provisioning for automated configuration distribution and software upgrades 825 Digium AEX804B - 1 Quad FXO PCI Express Card 234.0000 257.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex804b-1-quad-fxo-pci-express-card-p-825.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX804B is an AEX800 card with 1 x  Quad FXO module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX804B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 826 Digium AEX804E - 1 Quad FXO PCI Express Card with Echo Cancellation VPMADT032 344.0000 381.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex804e-1-quad-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-826.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX804E is an AEX800 card with 1 x Quad FXO modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX804E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 827 Digium AEX805B - 1 Quad FXO 1 FXO PCI Express Card 254.0000 281.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex805b-1-quad-fxo-1-fxo-pci-express-card-p-827.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX805B is an AEX800 card with 1 x Quad FXO and 1 FXO module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX805B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 828 Digium AEX805E - 1 Quad FXO 1 FXO PCI Express Card with Echo Cancellation VPMADT032 365.0000 404.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex805e-1-quad-fxo-1-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-828.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX805E is an AEX800 card with 1 x Quad FXO and 1 x FXO module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX805E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 829 Digium AEX806B - 1 Quad FXO 2 FXO PCI Express Card 275.0000 304.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex806b-1-quad-fxo-2-fxo-pci-express-card-p-829.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX806B is an AEX800 card with 1 x Quad FXO and 2 FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX806B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 830 Digium AEX806E - 1 Quad FXO 2 FXO PCI Express Card with Echo Cancellation VPMADT032 386.0000 427.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex806e-1-quad-fxo-2-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-830.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX806E is an AEX800 card with 1 x Quad FXO and 2 x FXO modules attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX806E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 831 Digium AEX808B - 2 Quad FXO PCI Express Card 316.0000 350.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex808b-2-quad-fxo-pci-express-card-p-831.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX808B is an AEX800 card with 2 x Quad FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX808B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 832 Digium AEX808E - 2 Quad FXO PCI Express Card with Echo Cancellation VPMADT032 426.0000 473.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex808e-2-quad-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-832.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX808E is an AEX800 card with 2 x Quad FXO modules attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX808E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 833 Digium AEX810B - 1 FXS PCI Express Card 170.0000 187.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex810b-1-fxs-pci-express-card-p-833.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX810B is an AEX800 card with 1 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX810B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 834 Digium AEX810E - 1 FXS PCI Express Card with Echo Cancellation 281.0000 311.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex810e-1-fxs-pci-express-card-with-echo-cancellation-p-834.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX810E is an AEX800 card with 1 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX810E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 835 Digium AEX820B - 2 FXS PCI Express Card 190.0000 208.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex820b-2-fxs-pci-express-card-p-835.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX820B is an AEX800 card with 2 x FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX820B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 836 Digium AEX820E - 2 FXS PCI Express Card with Echo Cancellation VPMADT032 300.0000 332.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex820e-2-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-836.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX820E is an AEX800 card with 2 x FXS modules attached as well as the echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX820E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 837 Digium AEX811B - 1 FXO 1 FXS PCI Express Card 191.0000 210.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex811b-1-fxo-1-fxs-pci-express-card-p-837.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX811B is an AEX800 card with 1 x FXO and 1 FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX811B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 838 Digium AEX811E - 1 FXO 1 FXS PCI Express Card with Echo Cancellation 301.0000 334.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex811e-1-fxo-1-fxs-pci-express-card-with-echo-cancellation-p-838.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX811E is an AEX800 card with 1 x FXO & 1 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX811E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 839 Digium AEX812B - 2 FXO 1 FXS PCI Express Card 212.0000 233.95 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex812b-2-fxo-1-fxs-pci-express-card-p-839.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX812B is an AEX800 card with 2 x FXO and 1 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX812B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 840 Digium AEX812E - 2 FXO 1 FXS PCI Express Card with Echo Cancellation VPMADT032 323.0000 357.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex812e-2-fxo-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-840.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX812E is an AEX800 card with 2 x FXO & 1 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX812E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 841 Digium AEX813B - 3 FXO 1 FXS PCI Express Card 233.0000 256.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex813b-3-fxo-1-fxs-pci-express-card-p-841.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX813B is an AEX800 card with 3 x FXO and 1 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX813B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 842 Digium AEX813E - 3 FXO 1 FXS PCI Express Card with Echo Cancellation VPMADT032 344.0000 380.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex813e-3-fxo-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-842.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX813E is an AEX800 card with 3 x FXO & 1 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX813E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 843 Digium AEX814B - 1 Quad FXO 1 FXS PCI Express Card 253.0000 279.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex814b-1-quad-fxo-1-fxs-pci-express-card-p-843.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX814B is an AEX800 card with 1 x Quad FXO and 1 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX814B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 844 Digium AEX814E - 1 Quad FXO 1 FXS PCI Express Card with Echo Cancellation VPMADT032 364.0000 403.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex814e-1-quad-fxo-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-844.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX814E is an AEX800 card with 1 x Quad FXO & 1 x FXS module attached as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX814E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 845 Digium AEX815B - 1 Quad FXO 1 FXO 1 FXS PCI Express Card 273.0000 302.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex815b-1-quad-fxo-1-fxo-1-fxs-pci-express-card-p-845.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX815B is an AEX800 card with 1 x Quad FXO, 1 FXO and 1 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX815B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 846 Digium AEX815E - 1 Quad FXO 1 FXO 1 FXS PCI Express Card with Echo Cancellation VPMADT032 385.0000 426.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex815e-1-quad-fxo-1-fxo-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-846.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX815E is an AEX800 card with 1 x Quad FXO, 1 FXO & 1 x FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX815E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 847 Digium AEX821B - 1 FXO 2 FXS PCI Express Card 210.0000 232.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex821b-1-fxo-2-fxs-pci-express-card-p-847.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX821B is an AEX800 card with 1 x FXO and 2 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX821B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 848 Digium AEX821E - 1 FXO 2 FXS PCI Express Card with Echo Cancellation VPMADT032 321.0000 355.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex821e-1-fxo-2-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-848.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX821E is an AEX800 card with 1 x FXO & 2 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX821E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 849 Digium AEX822B - 2 FXO 2 FXS PCI Express Card 231.0000 255.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex822b-2-fxo-2-fxs-pci-express-card-p-849.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX822B is an AEX800 card with 2 x FXO and 2 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX822B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 850 Digium AEX822E - 2 FXO 2 FXS PCI Express Card with Echo Cancellation VPMADT032 342.0000 378.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex822e-2-fxo-2-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-850.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX822E is an AEX800 card with 2 x FXO & 2 x FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX822E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 851 Digium AEX824B - 1 Quad FXO 2 FXS PCI Express Card 272.4400 301.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex824b-1-quad-fxo-2-fxs-pci-express-card-p-851.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX824B is an AEX800 card with 1 x Quad FXO and 2 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX824B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 853 Digium AEX824E - 1 Quad FXO 2 FXS PCI Express Card with Echo Cancellation VPMADT032 383.0000 424.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex824e-1-quad-fxo-2-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-853.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX824E is an AEX800 card with 1 x Quad FXO & 2 x FXS module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX824E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 854 Digium AEX830B - 3 FXS PCI Express Card 209.0000 230.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex830b-3-fxs-pci-express-card-p-854.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX830B is an AEX800 card with 3 x FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX830B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 855 Digium AEX830E - 3 FXS PCI Express Card with Echo Cancellation VPMADT032 320.0000 354.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex830e-3-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-855.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX830E is an AEX800 card with 3 x FXS modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX830E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 856 Digium AEX831B - 3 FXS 1 FXO PCI Express Card 229.0000 253.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex831b-3-fxs-1-fxo-pci-express-card-p-856.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX831B is an AEX800 card with 3 x FXS and 1 FXO module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX831B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 857 Digium AEX831E - 3 FXS 1 FXO PCI Express Card with Echo Cancellation VPMADT032 340.0000 377.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex831e-3-fxs-1-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-857.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX831E is an AEX800 card with 3 x FXS and 1 FXO module attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX831E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 858 Digium AEX84SB - 4 FXS PCI Express Card 228.0000 252.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex84sb-4-fxs-pci-express-card-p-858.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX84SB is an AEX800 card with 4 x FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX84SB provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 859 Digium AEX84SE - 4 FXS PCI Express Card with Echo Cancellation VPMADT032 338.0000 375.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex84se-4-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-859.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX84SE is an AEX800 card with 4 x FXS modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX84SE provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 860 Digium AEX840B - 1 Quad FXS PCI Express Card 243.0000 268.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex840b-1-quad-fxs-pci-express-card-p-860.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX840B is an AEX800 card with 1 x Quad FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX840B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 861 Digium AEX840E - 1 Quad FXS PCI Express Card with Echo Cancellation VPMADT032 353.0000 392.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex840e-1-quad-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-861.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX840E is an AEX800 card with 1 x Quad FXS modules attached and the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX840E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 862 Digium AEX841B - 1 Quad FXS 1 FXO PCI Express Card 264.0000 291.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex841b-1-quad-fxs-1-fxo-pci-express-card-p-862.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX841B is an AEX800 card with 1 x Quad FXS and 1 FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX841B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 864 Digium AEX841E - 1 Quad FXS 1 FXO PCI Express Card with Echo Cancellation VPMADT032 374.0000 415.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex841e-1-quad-fxs-1-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-864.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX841E is an AEX800 card with 1 x Quad FXS and 1 x FXO module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX841E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 865 Digium AEX842B - 1 Quad FXS 2 FXO PCI Express Card 284.0000 314.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex842b-1-quad-fxs-2-fxo-pci-express-card-p-865.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX842B is an AEX800 card with 1 x Quad FXS and 2 FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX842B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 866 Digium AEX842E - 1 Quad FXS 2 FXO PCI Express Card with Echo Cancellation VPMADT032 395.0000 438.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex842e-1-quad-fxs-2-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-866.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX842E is an AEX800 card with 1 x Quad FXS and 2 x FXO modules attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX842E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 867 Digium AEX844B - 1 Quad FXS 1 Quad FXO PCI Express Card 325.0000 360.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex844b-1-quad-fxs-1-quad-fxo-pci-express-card-p-867.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX844B is an AEX800 card with 1 Quad FXS and 1 Quad FXO modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX844B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 868 Digium AEX844E - 1 Quad FXS 1 Quad FXO PCI Express Card with Echo Cancellation VPMADT032 436.0000 484.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex844e-1-quad-fxs-1-quad-fxo-pci-express-card-with-echo-cancellation-vpmadt032-p-868.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX844E is an AEX800 card with 1 x Quad FXS and 1 x Quad FXO module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX844E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 869 Digium AEX850B - 1 Quad FXS 1 FXS PCI Express Card 262.0000 290.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex850b-1-quad-fxs-1-fxs-pci-express-card-p-869.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX850B is an AEX800 card with 1 x Quad FXS and 1 FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX850B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 870 Digium AEX850E - 1 Quad FXS 1 FXS PCI Express Card with Echo Cancellation VPMADT032 373.0000 413.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex850e-1-quad-fxs-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-870.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX850E is an AEX800 card with 1 x Quad FXS and 1 x FXS module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX850E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 871 Digium AEX860B - 1 Quad FXS 2 FXS PCI Express Card 282.0000 311.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex860b-1-quad-fxs-2-fxs-pci-express-card-p-871.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX860B is an AEX800 card with 1 x Quad FXS and 2 FXS module attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX860B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 872 Digium AEX860E - 1 Quad FXS 2 FXS PCI Express Card with Echo Cancellation VPMADT032 392.0000 435.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex860e-1-quad-fxs-2-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-872.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX860E is an AEX800 card with 1 x Quad FXS and 2 x FXS module attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX860E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 873 Digium AEX851B - 1 Quad FXS 1 FXO 1 FXS PCI Express Card 283.0000 313.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex851b-1-quad-fxs-1-fxo-1-fxs-pci-express-card-p-873.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX851B is an AEX800 card with 1 x Quad FXS, 1 FXO and 1 FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX851B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 874 Digium AEX851E - 1 Quad FXS 1 FXO 1 FXS PCI Express Card with Echo Cancellation VPMADT032 394.0000 436.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex851e-1-quad-fxs-1-fxo-1-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-874.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX851E is an AEX800 card with 1 x Quad FXS, 1 FXO & 1 x FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX851E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 875 Digium AEX880B - 2 Quad FXS PCI Express Card 335.0000 371.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex880b-2-quad-fxs-pci-express-card-p-875.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX880B is an AEX800 card with 2 x Quad FXS modules attached. The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX880B provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 876 Digium AEX880E - 2 Quad FXS PCI Express Card with Echo Cancellation VPMADT032 446.0000 498.00 Digium Digium AEX800 PCI Express http://www.voipon.co.uk/digium-aex880e-2-quad-fxs-pci-express-card-with-echo-cancellation-vpmadt032-p-876.html http://www.voipon.co.uk/images/digium_aex800_sml.jpg new Availability: In stock The Digium AEX880E is an AEX800 card with 2 x Quad FXS modules attached, as well as the hardware echo cancellation module (VPMADT032). The AEX800 is a half-length PCI-Express 1.0-compliant, 8 port analog interface card. It supports combinations of station and/or trunk/line modules for a total of 8 interfaces. Like the Digium PCI-based TDM800P, the AEX880E provides an industry first with 8 standard two-wire, RJ-11 interfaces on a single card bracket. This eliminates the need for multiple brackets, external dongles, or splitters. In doing so, the AEX800 reduces part complexity, cable clutter, and points of failure. The AEX800 contains two module banks. Each bank supports up to four analog interfaces and may be filled either with one standard Digium® quad station (S400M) or line/trunk (X400M) analog module, or up to two standard Digium single analog station (S110M) or line/trunk (X100M) modules, enabling the creation of any combination of ports. The optional VPMADT032 hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior echo cancellation on both types of analog interfaces. The image shown is an AEX800 card bundled with the echo cancellation module (purple), 1 x Quad FXS module (green), and 1 x Quad FXO module (red). 877 Intertex SurfinBird IX78 ADSL AIR GW2 184.0000 189.00 Intertex Intertex Routers http://www.voipon.co.uk/intertex-surfinbird-ix78-adsl-air-gw2-p-877.html http://www.voipon.co.uk/images/intertex-ix78-wlan-sml.jpg new Availability: In stock The Intertex SurfinBird IX78 is a SIP capable router firewall with a built-in n ADSL2+ modem supporting ADSL speeds up to 24 Mbps downstream and up to 3 Mbps upstream using the new Annex M standard. It can also be used as an Ethernet firewall and then handles bandwidths up to 90 Mbps. The IX78 includes 4 Ethernet built-in ports with the capability of setting up DMZs and VLANs. Several other features are included in this router: high speed 802.11 b/g wireless access point, 2 ports for analog telephones or faxes (FXS), SIP gateway to the traditional telephone network (FXO), VPN termination, far-end NAT traversal, and the Intertex IP-PBX SIP Switch. Additionally, the IX78 offers TR-069 provisioning which enables service providers to automatically configure their remotely installed customer base.The IX78 also serves as a Triple Play router enabling all multimedia services to become available to all clients. The IX78 has a unique and extensive SIP (Session Initiation Protocol) support. The IX78 handles global SIP based IP communication and allows usage of all SIP applications on the LAN and over the Internet. A built-in SIP proxy and registrar dynamically controlling the NAT and firewall, enables SIP communication between the private LAN and the Internet. This allows users behind the IX78 to utilize the full power of SIP based live IP communication, whether offered by service providers or by using the built-in SIP proxy and registrar as a stand-alone SIP server. The two FXS ports allow analogue telephones to be connected and used as IP telephones. The IX78 also includes a gateway between SIP based IP telephony and the traditional telephony network via its FXO port. This gateway is useful for handling emergency “911” calls and for local back up for the IP telephony. In addition, it integrates calls received on the ordinary telephone line with the IP telephony system. Multiple SIP phones and soft PC SIP clients can also be connected to the LAN and become fully integrated with any SIP service. The IX78 includes: Router/Firewall with 4-5 Ethernet Switch ADSL2+ modem with Annex A/B/M (24 Mbps DS, 3 Mbps US) Ethernet WAN Triple play and various routing configuration possibilities 2 FXS ports for analog telephones FXO port: Real SIP/PSTN gateway + Fallback on WAN loss Wireless 802.11b/g as Access Point Unique SIP support, Proxy and Registrar PBX-like functionality Advanced QoS for voice IPSec VPN Server TR-069 and proprietary flexible provision system 878 Grandstream GXP-2000 Spare PSU 7 7.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp2000-spare-psu-p-878.html http://www.voipon.co.uk/images/no_image_available.jpg new Availability: In Stock Power supply unit designed to work with the Grandstream GXP-2000.       879 Snom DECT Repeater 108.0000 120.00 Snom Snom DECT SIP Phone http://www.voipon.co.uk/snom-dect-repeater-p-879.html http://www.voipon.co.uk/images/snom_dect_repeater.jpg new Availability: In Stock The snom DECT repeater can be deployed when there is a need to extend the range of the snom m3 dect IP phone, or the desire arises to increase limited coverage or improve reception in far away areas. Not all repeaters are created the same. In order to maximise improvements to the DECT telephone's operational ability (whether increased range, improved reception, or both), until now it was important to select the according repeaters. The snom DECT repeaters comply with all this requirements. The snom DECT Repeater is designed in accordance with the DECT standard. It is intended for a use in residential, business and public DECT applications based on the GAP profile. So they built an ideal addition to yoursnom m3. The snom DECT Repeater has two internal antennas and can be used to double the effective range of a DECT Base Station. And the snom DECT Repeater can ensure the seamless intercell handover from the base station area to the repeater area and vice versa. It is possible to register up to six repeaters to a single base station. Furthermore the snom DECT Repeater uses a new, automatic registration method. With this new method, the repeater automatically seeks and acquires the DECT/GAP base station that emits the most powerful signal. For the end user this means that the repeater installation has been simplified to Plug-and-Play. 880 OpenVox A400M01 - 1 FXO Mini-PCI Card 107.0000 136.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m01-1-fxo-minipci-card-p-880.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M01 is an OpenVox A400M bundled with 1 FXO module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M01 is the mini-PCI version of the A400P. 881 OpenVox A400M02 - 2 FXO Mini-PCI Card 133.0000 168.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m02-2-fxo-minipci-card-p-881.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M02 is an OpenVox A400M bundled with 2 FXO module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M02 is the mini-PCI version of the A400P. 882 OpenVox A400M03 - 3 FXO Mini-PCI Card 156.0000 197.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m03-3-fxo-minipci-card-p-882.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M03 is an OpenVox A400M bundled with 3 FXO module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M03 is the mini-PCI version of the A400P. 883 OpenVox A400M04 - 4 FXO Mini-PCI Card 177.0000 224.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m04-4-fxo-minipci-card-p-883.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M04 is an OpenVox A400M bundled with 4 FXO module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M04 is the mini-PCI version of the A400P. 884 OpenVox A400M10 - 1 FXS Mini-PCI Card 107.0000 136.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m10-1-fxs-minipci-card-p-884.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M10 is an OpenVox A400M bundled with 1 FXS module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M10 is the mini-PCI version of the A400P. 885 OpenVox A400M11 - 1 FXS 1 FXO Mini-PCI Card 127.0000 160.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m11-1-fxs-1-fxo-minipci-card-p-885.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M11 is an OpenVox A400M bundled with 1 FXS and 1 FXO module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M11 is the mini-PCI version of the A400P. 886 OpenVox A400M12 - 1 FXS 2 FXO Mini-PCI Card 152.0000 192.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m12-1-fxs-2-fxo-minipci-card-p-886.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M12 is an OpenVox A400M bundled with 1 FXS and 2 FXO modules. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M12 is the mini-PCI version of the A400P. 887 OpenVox A400M13 - 1 FXS 3 FXO Mini-PCI Card 175.0000 221.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m13-1-fxs-3-fxo-minipci-card-p-887.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M13 is an OpenVox A400M bundled with 1 FXS 3 FXO modules. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M13 is the mini-PCI version of the A400P. 888 OpenVox A400M20 - 2 FXS Mini-PCI Card 133.0000 168.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m20-2-fxs-minipci-card-p-888.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M20 is an OpenVox A400M bundled with 2 FXS module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M13 is the mini-PCI version of the A400P. 889 OpenVox A400M21 - 2 FXS 1 FXO Mini-PCI Card 149.0000 189.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m21-2-fxs-1-fxo-minipci-card-p-889.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M21 is an OpenVox A400M bundled with 2 FXS 1 fxo module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M21 is the mini-PCI version of the A400P. 890 OpenVox A400M22 - 2 FXS 2 FXO Mini-PCI Card 177.0000 224.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m22-2-fxs-2-fxo-minipci-card-p-890.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M22 is an OpenVox A400M bundled with 2 FXS 2 fxo module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M22 is the mini-PCI version of the A400P. 891 OpenVox A400M30 - 3 FXS Mini-PCI Card 158.0000 200.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m30-3-fxs-minipci-card-p-891.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M30 is an OpenVox A400M bundled with 3 FXS 2 fxo module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M30 is the mini-PCI version of the A400P. 892 OpenVox A400M31 - 3 FXS 1 FXO Mini-PCI Card 177.0000 224.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m31-3-fxs-1-fxo-minipci-card-p-892.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M31 is an OpenVox A400M bundled with 3 FXS 1 fxo module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M31 is the mini-PCI version of the A400P. 893 OpenVox A400M40 - 4 FXS Mini-PCI Card 177.0000 224.00 OpenVox OpenVox A400M mini-PCI http://www.voipon.co.uk/openvox-a400m40-4-fxs-minipci-card-p-893.html http://www.voipon.co.uk/images/openvox_a400m.jpg new Availability: In stock The A400M40 is an OpenVox A400M bundled with 4 FXS module. The Openvox A400M is a 4-port analog card with mini-PCI and works with PSTN line. It can be used to build a PBX system based on Asterisk open source platform. The above picture is an A400M22 You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The card supports up to a total of 4 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A400M card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A400M card to terminate analog telephones. Key Features and benefits: Support mini PCI type III Designed for low-power systems Support AskoziaPBX system Support VIA, PC Engines motherboard and AMD geode based motherboard Support trixbox, Elastix and other asterisk based distributions Easy to use: Full GPL license driver- Zaptel World Wide Usable: Configurable line interface to meet global telephone line interface requirements Modular design: Can upgrade at any time by adding extra FXO or FXS modules up to a total of 4 per card. Excellent value: High quality at a low price. Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty The A400M40 is the mini-PCI version of the A400P. 894 OpenVox B100P PCI ISDN BRI Card 55.0000 71.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b100p-pci-isdn-bri-card-p-894.html http://www.voipon.co.uk/images/openvox_b100p.jpg new Availability: In stock B100P is a PCI 2.2 compliant card supporting 1 BRI S/T interface. NT/TE mode can be configured on the port. It can be used to build Asterisk based systems such as ISDN PBX and Voip gateway­. The OpenVox B100P can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: One integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI bus interface (Spec.2.2) for 3.3V and 5V signal environments Can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 895 OpenVox B200E PCI Express ISDN BRI Card 217.0000 264.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b200e-pci-express-isdn-bri-card-p-895.html http://www.voipon.co.uk/images/openvox_b200e.jpg new Availability: In stock The B200E is a PCI Express 2.2 compliant card supporting 2 BRI S/T interfaces, with a onboard multi NT powerfeeding circuit . NT/TE mode can be independently configured on each of the 2 ports. The Openvox B200E can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI Express bus interface (Spec.2.2) for 3.3V and 5V signal environments DTMF detection on all B-channels Multiparty audio conferences bridge Onboard power feeding PCM bus connectors daisy chaining Each of the 2 ports can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 896 OpenVox B400E PCI Express ISDN BRI Card 271.0000 330.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b400e-pci-express-isdn-bri-card-p-896.html http://www.voipon.co.uk/images/openvox_b400e.jpg new Availability: In stock The B400E is a PCI Express 2.2 compliant card supporting 4 BRI S/T interfaces, with a onboard multi NT powerfeeding circuit . NT/TE mode can be independently configured on each of the 4 ports. The B400E can be used for building Open Source Asterisk based systems such as ISDN PBX and VoIP gateways­. Target applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode Integrated PCI Express bus interface (Spec.2.2) for 3.3V and 5V signal environments DTMF detection on all B-channels Multiparty audio conferences bridge Onboard power feeding PCM bus connectors daisy chaining Each of the 4 ports can be independently configured for TE or NE mode Application ready: Use Asterisk to build your IP-PBX/Voicemail system RoHS compliant Certificates: CE and FCC Two year warranty 897 OpenVox SP140 -- A1200P RJ45 to RJ11 splitter (3 sets per package) 11.0000 14.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-sp140-a1200p-rj45-to-rj11-splitter-3-sets-per-package-p-897.html http://www.voipon.co.uk/images/openvox_sp140.jpg new Availability: In stock The OpenVox SP140 is a splitter used to split 8 pins in an RJ45 jack that A1200P uses to connect 4 telephone lines. Each SP140 contain 3 set split plus 90cm wire cable to satisfy 12 telephone lines required by A1200P. Features: RoHS compliant Certificates: CE and FCC Two year warranty 898 OpenVox SP141- A800P RJ45 to RJ11 splitter (2 sets per package) 7.6000 9.00 OpenVox OpenVox A800P http://www.voipon.co.uk/openvox-sp141-a800p-rj45-to-rj11-splitter-2-sets-per-package-p-898.html http://www.voipon.co.uk/images/openvox_sp140.jpg new Availability: In stock The OpenVox SP141 is a splitter used to split 8 pins in an RJ45 jack that A800P uses to connect 2 telephone lines. Each SP141 contains 2 set split plus 90cm wire cable to satisfy 8 telephone lines required by A800P. Features: RoHS compliant Certificates: CE and FCC Two year warranty 899 Grandstream GXE5024 IP PBX 284.0000 294.95 Grandstream Grandstream IP PBX http://www.voipon.co.uk/grandstream-gxe5024-ip-pbx-p-899.html http://www.voipon.co.uk/images/grandstream_gxe502x.jpg new Availability: In stock Grandstream GXE5024 IPPBX Appliance with 4 FXO Ports Grandstream's IP-PBX product segment consists of the GXE5024 and GXE5028. The GXE502x appliance is a powerful all-in-one voice + video + fax + data communication solution for the small to medium sized business, especially companies with sub-30 seats per location. The GXE502x takes modern business communication systems to a heightened level of innovation, quality, reliability, ease of deployment and affordability. Designed from ground up to support distributed IP communications, intelligent unified messaging, advanced application integration and popular PBX features, the GXE502x product family also optimally integrates legacy PSTN trunk and telephone interfaces for fail-safe hybrid communication needs in all circumstances including power or network loss. Key Features Integrated high performance data router with advanced voice/video QoS support Integrated legacy PSTN trunks, analog phone/FAX ports & unlimited SIP trunk options Integrated session border controller (SBC) for NAT/firewall traversal and secure telecommuting Integrated conference bridges that allow any combination of IP or PSTN calls using any codecs(built-in transcoding) Unified messaging including voicemail-to-email, fax-to-email and video-to-email (pending) Power and network failure survivability and recovery; Integrated PoE (802.3af) Support true & local emergency call routing in all circumstances Automated detection and provisioning of IP phones, video phones, ATA and other endpoints for easy deployment Rich PBX features such as presence, shared line appearance, call park & pickup, call queue, ACD, intercom & paging, ring group, customizable auto attendant & IVR, personal music-on-hold, branch office system peering Hardware accelerated encryption engine to ensure strongest security protection using SRTP and TLS Personal Web portal to manage individual phone/call setting, personal greeting, new or saved voice/fax/video messages for each extension user Flexible dial plan, call routing and call recording (pending) Feature Specifications Feature Specifications GXE5024 GXE5028 FXO Ports 4 FXO 8 FXO FXS Ports 2 2 Ethernet Ports 1 x WAN, 1 x LAN (10/100Mbps, integrated PoE) 1 x WAN, 1 x LAN (10/100Mbps, integrated PoE) PSTN Life Line Ports 2 PSTN fail-over life lines 2 PSTN fail-over life lines Peripheral Ports USB, Audio In, Audio Out USB, Audio In, Audio Out Conference Rooms 2 4 Unified Message Storage 75 hours of voicemail, 5000 fax pages, 2 hours of video mail 150 hours of voicemail, 10000 fax pages, 4 hours of video mail Registered Extensions 100 100 Voice Codecs G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay Video Codecs H.264, H.263/H.263+ H.264, H.263/H.263+ Communication/Security Protocols TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP Compliance FCC : Part 68 & 15B; CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick : AS-ACIF S002, AS/NZS60950 UL (power supply) FCC : Part 68 & 15B; CE : EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick : AS-ACIF S002, AS/NZS60950 UL (power supply) Universal Power Supply Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp Configuration & Management HTTP/HTTPS, TELNET, Syslog, TR-069 (pending) HTTP/HTTPS, TELNET, Syslog, TR-069 (pending) 900 Grandstream GXE5028 IP PBX 379.0000 394.95 Grandstream Grandstream IP PBX http://www.voipon.co.uk/grandstream-gxe5028-ip-pbx-p-900.html http://www.voipon.co.uk/images/grandstream_gxe502x.jpg new Availability: In stock Grandstream GXE5028 IPPBX Appliance with 8 FXO Ports Grandstream's IP-PBX product segment consists of the GXE5024 and GXE5028. The GXE502x appliance is a powerful all-in-one voice + video + fax + data communication solution for the small to medium sized business, especially companies with sub-30 seats per location. The GXE502x takes modern business communication systems to a heightened level of innovation, quality, reliability, ease of deployment and affordability. Designed from ground up to support distributed IP communications, intelligent unified messaging, advanced application integration and popular PBX features, the GXE502x product family also optimally integrates legacy PSTN trunk and telephone interfaces for fail-safe hybrid communication needs in all circumstances including power or network loss. Key Features Integrated high performance data router with advanced voice/video QoS support Integrated legacy PSTN trunks, analog phone/FAX ports & unlimited SIP trunk options Integrated session border controller (SBC) for NAT/firewall traversal and secure telecommuting Integrated conference bridges that allow any combination of IP or PSTN calls using any codecs(built-in transcoding) Unified messaging including voicemail-to-email, fax-to-email and video-to-email (pending) Power and network failure survivability and recovery; Integrated PoE (802.3af) Support true & local emergency call routing in all circumstances Automated detection and provisioning of IP phones, video phones, ATA and other endpoints for easy deployment Rich PBX features such as presence, shared line appearance, call park & pickup, call queue, ACD, intercom & paging, ring group, customizable auto attendant & IVR, personal music-on-hold, branch office system peering Hardware accelerated encryption engine to ensure strongest security protection using SRTP and TLS Personal Web portal to manage individual phone/call setting, personal greeting, new or saved voice/fax/video messages for each extension user Flexible dial plan, call routing and call recording (pending) Feature Specifications Feature Specifications GXE5024 GXE5028 FXO Ports 4 FXO 8 FXO FXS Ports 2 2 Ethernet Ports 1 x WAN, 1 x LAN (10/100Mbps, integrated PoE) 1 x WAN, 1 x LAN (10/100Mbps, integrated PoE) PSTN Life Line Ports 2 PSTN fail-over life lines 2 PSTN fail-over life lines Peripheral Ports USB, Audio In, Audio Out USB, Audio In, Audio Out Conference Rooms 2 4 Unified Message Storage 75 hours of voicemail, 5000 fax pages, 2 hours of video mail 150 hours of voicemail, 10000 fax pages, 4 hours of video mail Registered Extensions 100 100 Voice Codecs G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay G.711, G.723, G.729 A/B/E, G.726, iLBC, T.38 fax relay Video Codecs H.264, H.263/H.263+ H.264, H.263/H.263+ Communication/Security Protocols TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DDNS, DHCP, NTP, TFTP, TELNET, HTTP/HTTPS, PPoE, SIP(RFC3261),STUN, SRTP, TLS/SIP Compliance FCC : Part 68 & 15B; CE: EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick : AS-ACIF S002, AS/NZS60950 UL (power supply) FCC : Part 68 & 15B; CE : EN55022, EN55024, TBR21, EN60950, C-Tick: AS/NZX CISPR22, CIS PR24 A-Tick : AS-ACIF S002, AS/NZS60950 UL (power supply) Universal Power Supply Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp Input: 100-240V, 50-60Hz Output: 12 Vdc, 1.25Amp Configuration & Management HTTP/HTTPS, TELNET, Syslog, TR-069 (pending) HTTP/HTTPS, TELNET, Syslog, TR-069 (pending) 901 Polycom SpectraLink 8002 Series Wireless Telephone 158.0000 174.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-spectralink-8002-series-wireless-telephone-p-901.html http://www.voipon.co.uk/images/polycom_spectralink_8002.jpg new Availability: In stock Cost-Effective Business Wireless: The SpectraLink 8002 Wireless Telephone is a cost-effective, business-grade mobile telephony solution for small to medium-sized businesses. Using the Wi-Fi Alliance's WMM QoS standard, the SpectraLink 8002 Wireless Telephone interoperates with most consumer-grade and SMB access point infrastructure devices, alleviating the need to install and maintain additional hardware while still providing enterprise-level security and quality voice. In keeping with the needs of the market segment, the SpectraLink 8002 is packaged in complete, ready-to-install bundles that include a single or dual charger, Battery Pack and a power supply. This lightweight handset offers a rich set of features, including a backlit keypad and high-resolution display, menu driven functions and messaging capabilities. With its reinforced yet ergonomic design, the SpectraLink 8002 is a lightweight handset that also provides shock resistance at an affordable price. Polycom's advanced manufacturing and test processes also deliver leading-class durability ensuring that the SpectraLink 8002 Wireless Telephone withstands the rigors of daily operation, thereby protecting your investment. Both administrators and users will appreciate the SpectraLink 8002 Wireless Telephone's ease-of-use and minimal training requirements, as well as the complete line of carrying options and accessories available to suit users in every work environment. Polycom helps thousands of commercial enterprises improve productivity, responsiveness and customer service with SpectraLink Wireless Telephones. The combination of a standards-based architecture, enterprise telephone system integration and exceptional voice quality make SpectraLink Wireless Telephones the right choice for workplace Wi-Fi telephony. SpectraLink 8002 Features: Free Asterisk Support Asterisk Ready 802.11b (Wi-Fi) standard-compatible WMM QoS standards-compatible Text messaging support via Open Application Interface (v 2.0) Four programmable soft keys supporting up to 16 programmable features Audible and vibrating ringers Integrated TFTP client DHCP or static IP addressing Features Offers cost-effective mobility solution for small to medium-sized businesses (SMBs) Dramatically improves responsiveness and productivity Incorporates enterprise-grade durability for investment protection Increases productivity through text messaging applications Integrates with SIP-based PBXs and a broad array of wireless infrastructure devices Minimizes training and administration time through intuitive handset operation Rounds out Polycom's comprehensive portfolio of SIP telephony endpoints 902 Siemens S685IP Dect SIP Phone 80.6000 84.00 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-s685ip-dect-sip-phone-p-902.html http://www.voipon.co.uk/images/siemens_s685ip.jpg new Availability: In stock VoIP with answering machine and multi accounts, unrivaled wideband audio quality. Thanks to High Definition Sound Performance (HDSP), the Gigaset S685IP lets you experience VoIP calls in unrivaled wideband audio quality. The extended frequency range means you can enjoy a conversation in twice the clarity of a normal phone call and get closer to the person you're talking to. This elegant dual-mode phone lets you easily switch between internet and fixed-line calls at the touch of a button – and offers you the added convenience of an integrated answering machine. Sporting a big, brilliant colour display and intuitive icon-based menu, the Gigaset S685 IP is also both extremely easy on the eyes and a joy to use. It offers you the convenience of e-mail notification, which means you can find out the time, date and subject of new messages without having to open your e-mail account. It even features energy-saving ECO DECT technology that makes calling better for the environment – and more cost-effective for you. The Gigaset S685 IP with High Definition Sound Performance also offers you an array of easy-to-use advanced features that are the perfect complement to its superior wideband audio. The new technology provides double frequency range compared to conventional telephony and thereby comes up to the human voice frequency range. Key Features: With up to 30 mins recording time High Definition Sound Performance1 offering 8 kHz wideband audio thanks to greatly increased data compression provided by audio Codec G.722 Dual mode: easy switch from internet calls to fixed-line calls by single keypress Easy configuration of internet telephony (VoIP) without a PC Expandable phone system with multiline functionality for up to 6 handsets and 6 SIP accounts from different providers Up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call with multiple handsets Gigaset.net-ready E-mail notification with time; date and subject Instant Messaging (buddy list; chatting; presence status) Increased virus protection thanks to protected operating system Address book for 250 vCard entries to store first name; last name; 3 numbers; e-mail address and birthday of all your contacts    Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features All units supplied as base and 1 handset - Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. All units can connect to both a standard BT phone line and have an ethernet connection for VoIP calling. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 903 Siemens S68 Dect SIP Phone 40.8000 42.50 Siemens Siemens Dect SIP Phones http://www.voipon.co.uk/siemens-s68-dect-sip-phone-p-903.html http://www.voipon.co.uk/images/siemens_s68.jpg new Availability: In stock VoIP with answering machine and multi accounts, unrivaled wideband audio quality. Thanks to High Definition Sound Performance (HDSP), the Gigaset S685IP lets you experience VoIP calls in unrivaled wideband audio quality. The extended frequency range means you can enjoy a conversation in twice the clarity of a normal phone call and get closer to the person you're talking to. This elegant dual-mode phone lets you easily switch between internet and fixed-line calls at the touch of a button – and offers you the added convenience of an integrated answering machine. Sporting a big, brilliant colour display and intuitive icon-based menu, the Gigaset S685 IP is also both extremely easy on the eyes and a joy to use. Requires a Siemens Gigaset S685IP.     Siemens DECT handset comparison chart Phone Simultaneous Calls SIP Identities Answering Machine Wide band audio Bluetooth Phone Book Display Siemens C460IP 1 IP Call 1 Landline Call 1 SIP Account N N N 100 Numbers with name High Quality Colour Display 101*80 pixels Siemens S450IP 2 IP Calls 1 Landline Call 6 SIP Accounts N N N 150 Numbers + name and birthday notifications High Quality Colour Display 128*128 pixels Siemens C475IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y N N 150 Numbers + name High Quality Colour Display 128*128 pixels Siemens S685IP 2 IP Calls 1 Landline Call 6 SIP Accounts Y Y Y 250 vCard entries; VIP-entries (a specific melody and predefined picture can be assigned to each directory entry High Quality Colour Display 128*160 pixels Common Features Up to 5 Additional Handsets may be connected, giving 6 handsets in total. All supplied to UK spec with UK power supply. These phones do not depend on your PC being switched on. Notes Bluetooth is connectivity only with S85H handset Wideband Audio (G.722) only with S85H handset S675IP will not be launched in the UK. S685IP is the equivalent. Specifications subject to change without notice 904 Redfone foneBRIDGE2 Single T1/E1 Ethernet Bridge 494.0000 549.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2-single-t1e1-ethernet-bridge-p-904.html http://www.voipon.co.uk/images/redfone_fonebridge2_1port.jpg new Availability: In stock foneBRIDGE2 Single port without Echo Cancellation The foneBRIDGE2 Single port is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated blackbox "appliance" designed to streamline installation and enable redundant design of Asterisk ® based VoIP systems. Perfect for simple T1/E1 standalone Asterisk installs or complex clustered environmentsthat require High Availability with minimal downtime. Dual port capability out of the box without the upfront investment. Through a simple and cost effective upgrade license the 2nd port can be enabled on demand. foneBRIDGE2 eliminates the need to install proprietary TDM hardware cards in approved/compatible server configurations. Instead, foneBRIDGE2 terminates T1/E1 PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using the Asterisk TDMoE protocol. Key Features Failover & Asterisk HA Enabled High-Availability through rapid failover capabilities ensures your critical telephony services are alwayson and available Flexible Configuration Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment. Field upgradeable Solid State Embedded Appliance No moving components Low heat and power consumption High MTBF (mean time before failure) Optional DSP based Carrier Grade Echo Cancellation Target Applications • T1/E1 PRI Trunk termination • Legacy PBX-to-Asterisk integration • High-Availability/Failover Asterisk clusters • Channel Bank connectivity • Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) • Blade Servers where PCI slots are not available Telephony • PRI Switch Compatibility– EuroISDN, AT&T 4ESS, DMS 100, Lucent 5E, NI1/NI2; Network or CPE • Line Interface– Dual T1/E1 (RJ45). • Line Encoding– AMI/B8ZS for T1, AMI/HDB3 for E1 • Framing– SF, ESF, CCS, CAS • Robbed Bit Signaling (RBS/CAS) • Short and Long Haul line build out (LBO) • Adaptive Equalizer for line attenuation conditioning Ethernet • 10/100-BASE-TX Half/Full-duplex • 2 x Dedicated RJ45 Specifications Electrical: DC 500mA Max @ 5V (2.5W) • Environmental: 0 to 50 deg C operating • Physical: Dimensions: 5.00” x 6.50” x 1.00” Weight: 1 pound Mounting: Flange Mount or Desktop MODEL NUMBERS • No Echo Cancellation: 750-5050-Single • With Echo Cancellation: 750-5050-ECSingle • Upgrade License: 750-DUAL-Upgrade 905 Redfone foneBRIDGE2-EC Single T1/E1 Ethernet Bridge with EC 731.0000 819.00 RedFone Communications RedFone Ethernet Bridge http://www.voipon.co.uk/redfone-fonebridge2ec-single-t1e1-ethernet-bridge-with-ec-p-905.html http://www.voipon.co.uk/images/redfone_fonebridge2_1port.jpg new Availability: In stock foneBRIDGE2 Single port with Echo Cancellation The foneBRIDGE2 Single port is a T1/E1 PRI-to-Ethernet Bridge. It is an integrated blackbox "appliance" designed to streamline installation and enable redundant design of Asterisk ® based VoIP systems. Perfect for simple T1/E1 standalone Asterisk installs or complex clustered environmentsthat require High Availability with minimal downtime. Dual port capability out of the box without the upfront investment. Through a simple and cost effective upgrade license the 2nd port can be enabled on demand. foneBRIDGE2 eliminates the need to install proprietary TDM hardware cards in approved/compatible server configurations. Instead, foneBRIDGE2 terminates T1/E1 PRI lines on the trunk side and provides direct Ethernet communication to a network of Asterisk servers using the Asterisk TDMoE protocol. Key Features Failover & Asterisk HA Enabled High-Availability through rapid failover capabilities ensures your critical telephony services are alwayson and available Flexible Configuration Configurable on a per-port basis, foneBRIDGE2 allows you to mix multiple telephony standards on a single deployment. Field upgradeable Solid State Embedded Appliance No moving components Low heat and power consumption High MTBF (mean time before failure) Optional DSP based Carrier Grade Echo Cancellation Target Applications • T1/E1 PRI Trunk termination • Legacy PBX-to-Asterisk integration • High-Availability/Failover Asterisk clusters • Channel Bank connectivity • Mixed telephony environments (ex. E1 PRIs + T1 Channel Banks) • Blade Servers where PCI slots are not available Telephony • PRI Switch Compatibility– EuroISDN, AT&T 4ESS, DMS 100, Lucent 5E, NI1/NI2; Network or CPE • Line Interface– Dual T1/E1 (RJ45). • Line Encoding– AMI/B8ZS for T1, AMI/HDB3 for E1 • Framing– SF, ESF, CCS, CAS • Robbed Bit Signaling (RBS/CAS) • Short and Long Haul line build out (LBO) • Adaptive Equalizer for line attenuation conditioning Ethernet • 10/100-BASE-TX Half/Full-duplex • 2 x Dedicated RJ45 Specifications Electrical: DC 500mA Max @ 5V (2.5W) • Environmental: 0 to 50 deg C operating • Physical: Dimensions: 5.00” x 6.50” x 1.00” Weight: 1 pound Mounting: Flange Mount or Desktop MODEL NUMBERS • No Echo Cancellation: 750-5050-Single • With Echo Cancellation: 750-5050-ECSingle • Upgrade License: 750-DUAL-Upgrade 906 Sangoma A301 Clear Channel T3/E3 card 799.0000 999.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-a301-clear-channel-t3e3-card-p-906.html http://www.voipon.co.uk/images/sangoma_a301.jpg new Availability: In stock The A301 card is a 2U PCI adapter that supports clear channel T3 (44.7 Mbps) and E3 (34.4 Mbps) over dual BNC connectors. Based on bus mastering PCI technology supported by a ring-buffer DMA architecture, the A301 provides 132 Mbps of PCI bus transfer capacity with minimal real-time processor load. The A301 is ideal for larger organizations or ISPs that have outgrown their multiple T1 or DSL links. The A301 allows these users to take advantage of the relatively low cost of T3/E3 connections when compared to multiple T1 or E1 lines. Technical specifications Dimensions: 2U Form factor: 120mm x 55 mm for use in restricted chassis. 32 bit bus master DMA data exchanges across PCI interface at 132Mbytes/sec for minimum host processor intervention. Ring buffer DMA handling for minimum host intervention and guaranteed data integrity on high volume systems. Intelligent hardware: Downloadable Field Programmable Gate Array programming with multiple operating modes. Allows new features to be added when they become available. Power: 800mA peak, operational 300mA max at +3.3v or 5v. Autosense compatibility with 5v and 3.3v PCI busses. Temperature range: 0 - 50°C. T3/E3 Standards DS3/E3 Framer: ANSI T1.107_1988, ANSI T1.107_1989, T1.404, ITU-T G.751, ITU-T G.832, Bellcore TR-TSY-000009 and AT&T Pub 54014. LIU Receiver: G.775, GR-449, ITU G.751, ITU G.752, ITU G.755, ETSI TBR 24 and GR-499-CORE, 1995. LIU Transmitter: Bellcore GR-499, GR-253 and ANSI T1.102. Jitter Attenuator: T1.105.03b, ETSI TBR-24, Bellcore GR-253 and GR-499. Operating systems Linux (all versions, releases and distributions from 1.0 up). FreeBSD. Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. Higher level protocols IP/IPX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950 Production quality ISO 9002 Warranty Five years parts and labor. 907 Sangoma A142 Synchronous Dual Serial Card 252.0000 315.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-a142-synchronous-dual-serial-card-p-907.html http://www.voipon.co.uk/images/sangoma_a142_serial_card.jpg new Availability: In stock Now newly redesigned for improved efficiency, the Sangoma A142 Data Card replaces the Sangoma S5141 card. Ideal for synchronous serial applications. Whether you use the card to simplify routing and increase security using Sangoma's WANPIPE® software; you need support for an X.25 link, a datascope, an SNA connection, or a broadcast satellite feed; or you have another unique application - Sangoma's Serial Synchronous products communicate mission-critical radar control and surveillance data, essential to modern air and sea communications. This updated card boasts field upgradeable firmware, a choice of PCI or PCI Express interface, and two primary ports that have equal baud rates without the complexity of signalling jumpers. The A142 Synchronous Dual Serial hardware is based on the same advanced engineering design as the award-winning AFT Product Line. Trust the Sangoma® name to deliver optimum data support on standard telecommunications networks. Technical specifications Two full-speed primary ports supporting either V.35/RS422/EIA530 or RS232 serial interfaces. High-speed connections: V.35/RS422/EIA530 to 8 Mbps per port. RS232 to 400 kbps per port. Power: 550 mA at +5 V. PCI Versions are 32 bit (5 V) and 64 bit (3.3 V) compatible. Now available in PCI or PCI Express. Operating temperature range: 0 - 50°C. Software configurable or by machine BIOS. Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Datascope Features All modem control lines are monitored. Either monitoring only or simulation (transmit and receive). Monitoring or simulation of ATM or HDLC at line speeds above 8 Mbps, BSC at line speeds to 128 kbps. Supports Asynch to 256 kbps, and raw unformatted bit streams to 8 Mbps. Time stamps with a resolution of 100 microseconds or better to allow accurate sequencing of events. Each character can be individually time stamped. Serial Interfaces V.35, RS422 or EIA530, or RS232 supported on both primary ports. Clocking: Internally generated or external at line speeds to 8 Mbps. NRZ, NRZi, FM0, FM1, Manchester encoding. All ports are RS485 - capable of supporting multipoint lines. Operating systems Windows® 2000, Windows® XP, Windows® 9x. Linux (all versions, releases and distributions from 1.0 up). FreeBSD. Solaris. Line protocols ATM, Frame Relay, X.25, HDLC, PPP, SS7, BSC Point-to-Point, BSC 3270, SDLC, Transparent bit-stream. Higher level protocols IP/IPX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950 Production quality ISO 9002 Warranty Lifetime warranty on parts and labour. PLUS 30-day “no questions asked” return policy. 908 Sangoma A144 Synchronous Quad Serial Card 355.0000 445.00 Sangoma Sangoma Serial / Data Cards http://www.voipon.co.uk/sangoma-a144-synchronous-quad-serial-card-p-908.html http://www.voipon.co.uk/images/sangoma_a144_serial_card.jpg new Availability: In stock Now newly redesigned for improved efficiency, the Sangoma A144 Data Card replaces the Sangoma S5142 card. Ideal for synchronous serial applications. Whether you use the card to simplify routing and increase security using Sangoma's WANPIPE® software; you need support for an X.25 link, a datascope, an SNA connection, or a broadcast satellite feed; or you have another unique application-Sangoma's Serial Synchronous products communicate mission-critical radar control and surveillance data, essential to modern air and sea communications. This updated card boasts field upgradeable firmware, a choice of PCI or PCI Express interface, and four primary ports that have equal baud rates without the complexity of signalling jumpers. The A144 Synchronous Quad Serial hardware is based on the same advanced engineering design as the award-winning AFT Product Line. Trust the Sangoma® name to deliver optimum data support on standard telecommunications networks. Technical specifications Four full-speed primary ports supporting either V.35/RS422/EIA530 or RS232 serial interfaces. High-speed connections: V.35/RS422/EIA530 to 8 Mbps per port. RS232 to 400 kbps per port. Power: 550 mA at +5 V. PCI Versions are 32 bit (5 V) and 64 bit (3.3 V) compatible. Now available in PCI or PCI Express. Operating temperature range: 0 - 50°C. Software configurable or by machine BIOS. Dimensions: 2U Form factor: 120 mm x 55 mm for use in restricted chassis. Datascope Features All modem control lines are monitored. Either monitoring only or simulation (transmit and receive). Monitoring or simulation of ATM or HDLC at line speeds above 8 Mbps, BSC at line speeds to 128 kbps. Supports Asynch to 256 kbps, and raw unformatted bit streams to 8 Mbps. Time stamps with a resolution of 100 microseconds or better to allow accurate sequencing of events. Each character can be individually time stamped. Serial Interfaces V.35, RS422 or EIA530, or RS232 supported on all four primary ports. Clocking: Internally generated or external at line speeds to 8 Mbps. NRZ, NRZi, FM0, FM1, Manchester encoding. All ports are RS485 - capable of supporting multipoint lines. Operating systems Windows® 2000, Windows® XP, Windows® 9x. Linux (all versions, releases and distributions from 1.0 up). FreeBSD. Solaris. Line protocols ATM, Frame Relay, X.25, HDLC, PPP, SS7, BSC Point-to-Point, BSC 3270, SDLC, Transparent bit-stream. Higher level protocols IP/IPX over Frame Relay/PPP/HDLC/X.25, X.25 over Frame Relay (Annex G), BSC over X.25 (DMT and TCOP), SNA over X.25, PPPoE, PPPoA, IP over ATM. Diagnostic tools WANPIPEMON, SNMP, System logs Certification FCC Part 15 Class A, FCC Part 68, CISPR 22, EN 55022 Class A, CIPSR 24, AFIC-S016, IEC 60950 Production quality ISO 9002 Warranty Lifetime warranty on parts and labour. PLUS 30-day “no questions asked” return policy. 909 Voismart V2GSM 1 Port Card 396.0000 445.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-1-port-card-p-909.html http://www.voipon.co.uk/images/voismart_v2gsm.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 910 Voismart V2GSM 3 Port Card 662.0000 735.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-3-port-card-p-910.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 911 Voismart V2GSM 1 Port GSM-850MHz Board Card 481.0000 535.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-1-port-gsm850mhz-board-card-p-911.html http://www.voipon.co.uk/images/voismart_v2gsm.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 912 Voismart V2GSM 2 Port GSM-850MHz Board Card 645.0000 710.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-2-port-gsm850mhz-board-card-p-912.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 913 Voismart V2GSM 3 Port GSM-850MHz Board Card 747.0000 820.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v2gsm-3-port-gsm850mhz-board-card-p-913.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 914 Voismart V4GSM 4 Port GSM-850MHz Board Card 901.0000 1045.00 VoiSmart Voismart Asterisk GSM Cards http://www.voipon.co.uk/voismart-v4gsm-4-port-gsm850mhz-board-card-p-914.html http://www.voipon.co.uk/images/voismart_vgsm_3-port_sml.jpg new Availability: In Stock VoiSmart vGSM board is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings. No external gateways needed. 100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board. VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards. The antenna is external, and included in the package. You can find latest news and docs on VoiSmart Open Source website Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available. Bus Type: PCI 2.2 compliant IRQ Request levels: Allocated by PCI BIOS IO base addr (hex): Allocated by PCI BIOS Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors Plug & Play: yes External Interface: 4 x SMA/F 50 Ohm antenna connectors Antennas: included Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900 SMS: MT, MO, CB, Text and PDU mode SIM Interface: Supported SIM card: 3V Performance: 4 channels GSM (900/1800/1900) voice interface Where can I download the software? The software is based on Asterisk vISDN , see vISDN website. You can download a recent snapshot from the snapshots page. Installing the package is quite simple. be sure to have asterisk installed (along with headers files) be sure to have kernel headers installed, to be able to compile kernel modules (kernel must be > 2.6.11 ) in visdn src dir, just do ./configure ; make ; make install cp samples/30-visdn.rules to /etc/udev/rules.d (if asterisk is running as non-root, change the above file to set the right devices permissions. You can check an alternative rules file HERE ) copy samples/vgsm.conf and vgsm_operators.conf to you asterisk conf dir, normally /etc/asterisk edit the vgsm.conf according to the SIMs present on the board load the kernel modules: modprobe visdn_vgsm; modprobe visdn_streamport; modprobe visdn_timer_system start asterisk :) with "show vgsm interfaces" you should see each interface status (Example) with "show vgsm interfaces _interface_name_" you should see the detailed info of the interface selected ( Example ) *to dial out, just write into the dialplan: exten => _33.,1,Dial(VGSM/_interface_name_/${EXTEN}) 915 OpenVox B200M Mini PCI ISDN BRI Card 158.0000 196.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b200m-mini-pci-isdn-bri-card-p-915.html http://www.voipon.co.uk/images/b200m_mini_pci.jpg new Availability: In Stock The OpenVox B200M is a Mini PCI type III BRI card supporting 2 BRI S/T interfaces. NT/TE mode can be independently configured on each of the 2 ports. B200M can be used for building open source Asterisk® based systems such as ISDN PBX and VoIP gateway. Target Applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode DTMF detection on all B-channels RoHS compliant Multiparty audio conferences bridge Support mini PCI type III Designed for low-power systems Support AskoziaPBX system, trixbox, Elastix and other asterisk based distributions Support VIA, PC Engines motherboard and AMD geode based motherboard Each port can be configured for TE or NT mode Support Bristuff, ISDN4BSD and mISDN driver. Application ready: Use Asterisk to build your IP-PBX/Voicemail system. Two Year Warranty! At present, we do not provide power supply for NT mode which will be added soon. 916 OpenVox B400M Mini PCI ISDN BRI Card 236.0000 280.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b400m-mini-pci-isdn-bri-card-p-916.html http://www.voipon.co.uk/images/b400m_mini_pci.jpg new Availability: In Stock The OpenVox B400M is a Mini PCI type III BRI card supporting 4 BRI S/T interfaces. NT/TE mode can be independently configured on each of the 4 ports. B400M can be used for building open source Asterisk® based systems such as ISDN PBX and VoIP gateway. Target Applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: Two integrated S/T interfaces ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode DTMF detection on all B-channels RoHS compliant Multiparty audio conferences bridge Support mini PCI type III Designed for low-power systems Support AskoziaPBX system, trixbox, Elastix and other asterisk based distributions Support VIA, PC Engines motherboard and AMD geode based motherboard Each port can be configured for TE or NT mode Support Bristuff, ISDN4BSD and mISDN driver. Application ready: Use Asterisk to build your IP-PBX/Voicemail system. Two Year Warranty! At present, we do not provide power supply for NT mode which will be added soon. 917 OpenVox DE410P PCI ISDN PRI Card with Echo Cancellation 895.0000 995.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-de410p-pci-isdn-pri-card-with-echo-cancellation-p-917.html http://www.voipon.co.uk/images/openvox_de410p.jpg new The OpenVox DE410P is a bundling of our leading D410P product and our new EC100 Octasic DSP-based echo cancellation module. The EC100 provides a certified carrier-grade algorithm that has been labeled a benchmark for echo cancellation for OpenVox. With the improved I/O speed, the card reduces CPU usage and increased card density per server. The OpenVox DE410P is fully compatible with Asterisk applications. The open source driver supports an API interface for custom application development. DE410P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported. About OpenVox EC100 Module The OpenVox EC100 enables users to eliminate echo tails up to 128ms or 1024 taps across all 128 channels in E1 mode or 96 channels in T1/J1 modes. Further, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. Benefits: World recognized & deployed best voice quality Features: 128ms tail/channel (on all channel densities) Octasic Music Protection Adaptive Noise Reduction Automatic Level Control (G.169) Field upgradeable algorithm V.25 / V.8 answer tone (w/ and w/o phase reversal) DTMF as per Q.24 RoHS compliant Certificates: CE and FCC Two Year Warranty! For installation, please note the same procedure needs to be followed as for the installation of the D410P. Please refer to the user manual of D410P. 918 OpenVox DE410E PCI Express ISDN PRI Card with Echo Cancellation 895.0000 995.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-de410e-pci-express-isdn-pri-card-with-echo-cancellation-p-918.html http://www.voipon.co.uk/images/openvox_de410e.jpg new The OpenVox DE410E is a bundling of our leading D410E product and our new EC100 Octasic DSP-based echo cancellation module. The EC100 provides a certified carrier-grade algorithm that has been labeled a benchmark for echo cancellation for OpenVox. With the improved I/O speed, the card reduces CPU usage and increased card density per server. The OpenVox DE410E is fully compatible with Asterisk applications. The open source driver supports an API interface for custom application development. DE410E supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported. About OpenVox EC100 Module The OpenVox EC100 enables users to eliminate echo tails up to 128ms or 1024 taps across all 128 channels in E1 mode or 96 channels in T1/J1 modes. Further, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. Benefits: World recognized & deployed best voice quality Features: 128ms tail/channel (on all channel densities) Octasic Music Protection Adaptive Noise Reduction Automatic Level Control (G.169) Field upgradeable algorithm V.25 / V.8 answer tone (w/ and w/o phase reversal) DTMF as per Q.24 RoHS compliant Certificates: CE and FCC Two Year Warranty! For installation, please note the same procedure needs to be followed as for the installation of the D410E. Please refer to the user manual of D410E. 920 Polycom SoundStation IP6000 425.0000 446.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundstation-ip6000-p-920.html http://www.voipon.co.uk/images/polycom_ip6000_sml.jpg new Availability: In Stock The SoundStation IP 6000 is an advanced IP conference phone that delivers superior performance for small to midsize conference rooms. With advanced features, broad SIP interoperability and remarkable voice quality, the SoundStation IP 6000 is a price/performance breakthrough for SIP-enabled IP environments. For all conference calls, the SoundStation IP 6000 features Polycom HD Voice technology, boosting productivity and reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive conversations. It delivers high-fidelity audio bandwidth from 220 Hz to 14 kHz, capturing both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there. Beyond HD Voice, the SoundStation IP 6000 delivers advanced audio performance that far exceeds previous generations of conference phones. From full-duplex technology that eliminates distracting drop-outs to the latest echo cancellation advancements, only Polycom can deliver a conference phone experience with no compromises. Plus, Automatic Gain Control intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room, making the conversations clearer for all participants. It also features technology that resists interference from mobile phones and other wireless devices, delivering clear communications without distractions. The SoundStation IP 6000 leverages Polycom’s strong history in both conference phone and VoIP technology to deliver the most robust standards-based SIP interoperability in the industry. It shares the same SIP phone software base with Polycom’s award-winning SoundPoint IP products – the most comprehensive, reliable and feature-rich SIP products in the industry, with proven interoperability with a broad array of IP PBX and hosted platform. Robust provisioning, management and security features make Polycom’s family of IP conference phones the only choice for meeting rooms in SIP-based environments. Integrated Power over Ethernet (PoE) simplifies setup, with an AC power kit available for non-PoE environments. Plus, the SoundStation IP 6000 includes a high-resolution backlit display for vital call information and multi-language support. Key Features: Polycom HD Voice – unparalleled clarity to make your conference calls more efficient and productive - calls at up to 14 kHz. 12-foot microphone pickup – combined with Automatic Gain Control for performance far beyond older SoundStation IP conference phones(20% greater). Add up to two optional expansion microphones for even greater coverage. H igh-resolution display – enables robust call information and multi-language support. Call features - Call transfer, hold, divert (forward), pickup, local three-way conferencing, one touch speed dial, re-dial, call waiting, do-not-disturb, and more. Integrated Power over Ethernet (PoE) Acoustic Clarity Technology - for seamless voice conferencing. Robust interoperability - compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration. 921 Polycom SoundStation IP7000 638.0000 694.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-soundstation-ip7000-p-921.html http://www.voipon.co.uk/images/polycom_ip7000_sml.jpg new Availability: In Stock The SoundStation IP 7000 is a breakthrough conference phone that delivers outstanding performance and a robust feature set for SIP-based VoIP platforms. It is the most advanced conference phone ever developed, and is ideal for executive offices, conference rooms, and board rooms. The SoundStation IP 7000 features Polycom HD Voice technology, boosting productivity and reducing listener fatigue by turning ordinary conference calls into crystal-clear interactive conversations. It delivers high-fidelity audio from 160 Hz to 22 kHz, capturing both the deeper lows and higher frequencies of the human voice for conference calls that sound as natural as being there. The SoundStation IP 7000 is the most flexible and expandable conference phone ever developed. Connect two units together for increased loudness and microphone pickup, as well as multiple call control interfaces in the conference room. Connect up to two optional expansion microphones to a single phone to ensure close proximity for everyone in the room. In the future, you will be able to connect the SoundStation IP 7000 to the Polycom HDX high-definition video conferencing system for a complete, integrated voice and video conferencing solution. In the SoundStation IP 7000, Polycom has combined its rich history in voice conferencing and VoIP technology to develop a groundbreaking new conference phone that is the clear choice for SIP-enabled environments. It shares the same SIP phone software with Polycom’s award-winning SoundPoint IP products – the most comprehensive, reliable and feature-rich SIP products in the industry, with proven interoperability with a broad array of IP PBX and hosted platforms. Plus, the SoundStation IP 7000 features a large multi-line high-resolution LCD display with a full XHTML microbrowser, turning your conference phone into a robust applications platform for your conference room. Bundled applications include advanced three-party conference features and LDAP corporate directory integration. Key Features: Polycom HD Voice – unparalleled clarity to make your conference calls more efficient and productive - calls at up to 22 kHz, an industry first, for conference calls that sound as natural as being there. Polycom's patented Acoustic Clarity Technology - delivering the best conference phone experience with no compromises. 20-foot microphone pickup (thats 100% increase on previous generation) - and even more with optional expansion microphones or multi-unit connectivity to reach all corners of the conference room. Automatic Gain Control - intelligently adjusts the microphone sensitivity based on where participants are seated in the conference room. Unparalleled flexibility and expandability. Large, high-resolution LCD display for advanced applications - includes XHTML microbrowser - enables new applications that make conference calling easier and more functional. Integrated Power over Ethernet (PoE) Strong, robust SIP software - leveraging the most advanced SIP endpoint software in the industry, with advanced call handing, security, and provisioning features. Robust interoperability - compatible with a broad array of SIP call platforms to maximize voice quality and feature availability while simplifying management and administration 924 OpenVox DE210P PCI ISDN PRI Card with Echo Cancellation 570.0000 634.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-de210p-pci-isdn-pri-card-with-echo-cancellation-p-924.html http://www.voipon.co.uk/images/openvox_de210p_sml.jpg new Availability: In Stock The DE210P is the D210P product and OpenVox's new EC100 Octasic DSP-based echo cancellation module. The EC100 provides a certified carrier-grade algorithm that has been labeled a benchmark for echo cancellation for OpenVox. With the improved I/O speed, the card reduces CPU usage and increased card density per server. DE210P is fully compatible with Asterisk applications. The open source driver supports an API interface for custom application development. DE210P supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported. About the OpenVox EC100-64 Module The OpenVox EC100-64 enables users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes. Further, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. Key Features: 128ms tail/channel (on all channel densities) Octasic Music Protection Adaptive Noise Reduction Automatic Level Control (G.169) Field upgradeable algorithm V.25 / V.8 answer tone (w/ and w/o phase reversal) DTMF as per Q.24 RoHS compliant Certificates: CE and FCC Two Year Warranty! 925 OpenVox DE210E PCI Express ISDN PRI Card with Echo Cancellation 570.0000 634.00 OpenVox OpenVox Digital (PRI) Cards http://www.voipon.co.uk/openvox-de210e-pci-express-isdn-pri-card-with-echo-cancellation-p-925.html http://www.voipon.co.uk/images/openvox_de210e_sml.jpg new Availability: In Stock The DE210E is the D210E product and OpenVox's new EC100 Octasic DSP-based echo cancellation module. The EC100 provides a certified carrier-grade algorithm that has been labeled a benchmark for echo cancellation for OpenVox. With the improved I/O speed, the card reduces CPU usage and increased card density per server. DE210E is fully compatible with Asterisk applications. The open source driver supports an API interface for custom application development. DE210E supports industry standard telephony and data protocols, including Primary Rate ISDN (both N. American and Standard Euro) protocol families for voice, PPP, Cisco, HDLC, and Frame Relay data modes. Both line-side and trunk-side interfaces are supported. The OpenVox DE210E is the PCI-Express version of the DE210P. About the OpenVox EC100-64 Module The OpenVox EC100-64 enables users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes. Further, this module takes advantage of the Octasic Voice Quality Enhancement to provide superior sound quality on all calls. Key Features: 128ms tail/channel (on all channel densities) Octasic Music Protection Adaptive Noise Reduction Automatic Level Control (G.169) Field upgradeable algorithm PCI-Express V.25 / V.8 answer tone (w/ and w/o phase reversal) DTMF as per Q.24 RoHS compliant Certificates: CE and FCC Two Year Warranty! 926 VXI QD1028-V Lower Cord 16.0000 16.00 VXI Corporation VoIP Accessories http://www.voipon.co.uk/vxi-qd1028v-lower-cord-p-926.html http://www.voipon.co.uk/images/lower_cord.jpg new Availability: In Stock "Y" Training Cord with Quick Disconnect for TuffSet and Passport Headset Systems. 927 EuroTech WINNER Analog to GSM Gateway - 1 GSM channel 130.0000 145.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-winner-analog-to-gsm-gateway-1-gsm-channel-p-927.html http://www.voipon.co.uk/images/eurotech_analog-single_sml.jpg new Availability: In stock EuroTech's Analog to GSM Gateway provides an efficient, cost effective solution for connecting your current telephone system to the mobile GSM network. When making a call from the PSTN line to a cellular phone, the device acts as a router and determines the Least Cost Route (LCR.) The GSM gateway connects to an extension and bypasses the telephone company thus significantly reducing overhead costs of telephone calls. By inserting 1 (or upto 3 with different operators) SIM card(s) into the device, and by being combined with the embedded 'Wismo Quik' Wavecom module, the gateway itself becomes a cell phone! Call costs can be cut by upto 50% thanks to great inclusive minute deals in today's GSM marketplace. The WINNER gateways are therefore an essential tool for cutting overhead costs of telephone bills as they allow direct access to the GSM networks and cuts costs of airtime and interconnections. Key Features Outgoing calls - LCR directs outgoing calls over the normal PSTN network, or GSM Call Forwarder - From mobile to local PBX or PSTN/ VoIP FXS - for trunk connection Easy to use - configurable via DTMF Strong Signal - High Gain external antenna for optimum strength (and 3m cable) Echo Cancellation Reverse polarity - battery reversal, double reversal or break of line CLID, PIN, audio set capability LCD screen - showing network, signal strength, and call status for programming Dual-band - 900/1800 or 850/1900 Mhz Compact - This GSM single line analogue FCT (Fixed Cellular Terminal) is the smallest FCT/GSM Gateway available in the marketplace 928 EuroTech WINNER Analog to GSM Gateway - 2 GSM channels 309.0000 344.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-winner-analog-to-gsm-gateway-2-gsm-channels-p-928.html http://www.voipon.co.uk/images/eurotech_analog-dual_sml.jpg new Availability: In stock EuroTech's Analog to GSM Gateway provides an efficient, cost effective solution for connecting your current telephone system to the mobile GSM network. When making a call from the PSTN line to a cellular phone, the device acts as a router and determines the Least Cost Route (LCR.) The GSM gateway connects to an extension and bypasses the telephone company thus significantly reducing overhead costs of telephone calls. Using the embedded 'Wismo Quik' Wavecom module, and SIM cards, the gateway itself becomes a cell phone! Call costs can be cut by upto 50% thanks to great inclusive minute deals in today's GSM marketplace. The WINNER gateways are therefore an essential tool for cutting overhead costs of telephone bills as they allow direct access to the GSM networks and cuts costs of airtime and interconnections. Key Features 2 Channels - Two simultaneous calls to any two cellular operators Dual channel utilizes 8 SIMs (enabling two simultaneous outgoing calls) using 1-4 SIMs per port Intelligent LCR - LCR with up to 4 cellular networks LCR by prefix or timetable, day of week, hours to call and by number of calls Managable - Windows based powerful management system Programmable - Each SIM has up to 32 prefixes, user may set a specific tariff for each of the 32 prefixes All programming and monitoring may be done via the PC or DTMF and viewed on the LCD Interface - Analogue FXS (trunk) or FXO (extension is optional) interface Integration - Can connect over VOIP gateway Call Back - from mobile to PSTN/ VoIP Call Forwarder - From mobile to local PBX or PSTN/ VoIP Easy to use - configurable via DTMF Strong Signal - High Gain external antenna for optimum strength (and 3m cable) Echo Cancellation Reverse polarity - battery reversal, double reversal or break of line CLID, PIN, audio set capability LCD screen - showing network, signal strength, and call status for programming Dual-band - 900/1800 or 850/1900 Mhz Compact - This GSM single line analogue FCT (Fixed Cellular Terminal) is the smallest FCT/GSM Gateway available in the marketplace 929 EuroTech WINNER Cellufax - Fax to GSM 175.0000 195.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-winner-cellufax-fax-to-gsm-p-929.html http://www.voipon.co.uk/images/eurotech_fax-gsm_sml.jpg new Availability : In stock The Cellufax: the last word in paper-to-paper fax transmission via the GSM network. EuroTech's remote location solution that will bridge the gap left by the absence of landline infrastructure enabling an efficient and cost effective flow of data, whereby a paper fed into one fax machine is carried through the cellular network to a remote fax machine where it exits again as a paper. The phone cable from the fax machine is connected to the WINNER Cellufax analog fax gateway via RJ11. The SIM card is inserted into the WINNER Cellufax After a little configuration, faxes can be sent and received via GSM! No technical knowledge is required to install.   930 EuroTech WINNER BRI to GSM Gateway - 2 GSM channels 350.0000 391.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-winner-bri-to-gsm-gateway-2-gsm-channels-p-930.html http://www.voipon.co.uk/images/eurotech_bri-gsm_sml.jpg new Availability: In stock EuroTech GSM Gateway for ISDN connectivity. The BRI unit comprises 2 GSM channels, 8 SIM cards, enabling it to make two (2) simultaneous calls to eight (8) different cellular operators. Now with special extension - number recall/follow-me. The Dual Cell BRI Gateway connects your desk phones directly to cellular phone networks. This gateway completely bypasses the local landline telephone company. Two desk phones can be connected to each Dual Cell gateway via a Phone switching unit (often a PBX type switch). Each port in the Dual-Cell gateway contains a SIM. A SIM is a smart card that contains information such as the GSM network and phone number billing and verification Id. The Dual Cell can be configured to direct phone calls through the most economical SIM. Main Features: Dual Ports - 2 simultaneous calls Compact - 16.5cm x 10.0cm x 3.0cm Intelligent LCR - can be programmed to route calls according to prefixes, timetable, number of calls, allowing users to take advantage of the lowest tariffs accessible CDR signaling for billing - 12 fields download to files in the following formats: Excel, CSV, Access Up to 4 SIMs per port, and 2 antennas Configurable - Remote access via PC and IP Windows based powerful management system Bi-directional communication for 2 channels at the same time Boot loader for frameware S/W upgrades via PC or IP BTS locking - manual or automatic Dual Band - Wavecom is used at 900/1800 MHz or 850/1900MHz. 931 EuroTech VoIP Master GSM to VoIP Gateway 175.0000 195.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-voip-master-gsm-to-voip-gateway-p-931.html http://www.voipon.co.uk/images/eurotech_voip-gsm_sml.jpg new Availability: In stock The EuroTech VoIP Master cConnects you from anywhere to everywhere, at anytime, and at the lowest possible cost. Its Plug-and-Play, easy and simple to use. It connects your Mobile via VoIP in SIP protocol for free VoIP calls and low cost PSTN calls via a SIP provider (such as ourselves.) Main features Calls from your mobile phone over VOIP Call from your mobile phone to the gateway unit - it will provide you with a dial tone which will connect you via VoIP to another party FREE OF CHARGE. Note: If you call VoIP to PSTN you can use VoIPon's great call rates at only a few pence per minute. ( VoIPon services ) Incoming calls from VOIP to mobile phone- “Follow-Me” feature. For incoming calls from VoIP it can follow you to your mobile phone so you receive any calls anytime, anywhere. Outgoing calls - will be routed via an intelligent LCR (Least Cost Route) that directs cellular calls to the cellular network (via GSM SIM card), and the VoIP calls to the VoIP network, based on the prefix you call. 932 Pika WARP Asterisk Appliance 319.0000 355.00 Pika Technologies Pika IP PBX http://www.voipon.co.uk/pika-warp-asterisk-appliance-p-932.html http://www.voipon.co.uk/images/pika-asterisk-appliance_sml.jpg new Availability: In stock PIKA WARP the Appliance for Asterisk® is ideal for developers looking for a small, low cost computer replacement to deploy Asterisk based applications in the Small Office/Home Office (SOHO) and Small/Medium Enterprises (SME) markets. Completely customizable, it is compatible with VOIP phones as well as analog sets. Unlike your typical computer or appliance, PIKA has covered all your customer’s traditional telephony requirements. Music on Hold (MOH) and Paging can be cumbersome to add to a data centric solution as is power failure transfer (PFT), but all are included in the PIKA appliance. The configuration of the appliance is modular and can include up to 9 ports of a combination of FXO/FXS/BRI plus VOIP stations and trunks. The appliance is designed to address businesses with up to 100 phones Key Features AMCC Power PC 440EP Embedded 533 MHz Processor 1200 mips Supports floating point and MMU (memory management unit) I nternal flash 4 MB NOR memory (uboot) - plus 256 MB NAND(OS + apps) Internal RAM 256 MB External removable 1 GB SD flash memory For additional voice mail prompts / storage For back-up of configuration files and custom settings No hard drive improves reliability 10/100BT Ethernet port One USB host port (v1.1) One FXS port - with every unit 4-port FXS station, 4-port FXO trunk and 4 port/8 channel BRI modules can be used in any combination to a maximum of 8 additional ports Each analog 4-port module also contains a power failure transfer jack (RJ-11) Audio in & Audio out jacks - for music-on-hold and paging functions Backlit LCD display - 40 character (2 x 20) display with scroll button (API controlled) inverts for wall mounting Power LED Reset Button External brick-format universal power supply - with country variant cords Space saving - Surface standalone/stackable or wall mountable to accommodate any space requirements Dynamic thermal management RS232 programming port 933 Digium Switchvox AA60 Asterisk Appliance 935.0000 985.00 Digium Switchvox Digium Switchvox IP PBX http://www.voipon.co.uk/digium-switchvox-aa60-asterisk-appliance-p-933.html http://www.voipon.co.uk/images/switchvox_aa60-sml.jpg new Availability: In stock The Digum Switchvox Asterisk Appliance (AA60) is an Asterisk-based embedded IP PBX designed for small to medium businesses with 2-20 users. This version supports up to 10 concurrent calls and unlimited extensions. The package includes: Switchvox Server Switchvox SOHO Pre-Installed Software Digium Switchvox Silver subscription for 10 users Supports up to 10 concurrent calls Cold spare failover 1 year hardware warranty Below listed are just some of Switchvox's great features: Extensions Unlimited IP Phone/ ATA extensions Auto Attendant - IVRs 3,4 and 5 digit extensions Call control Hold Assisted transfer Blind transfer Call parking DND Send calls Voicemail Flexible Voicemail access Voicemail to Email Automatic Mailbox creation Call Queues Unlimited call queues In queue call routing Custom music on hold per queue Announce position in queue Announce estimated hold time Real time queue status Queue logs Various settings for picking up callers from queue Ring All Round Robin Fewest Calls Least Recently Called Random Conferencing 1 conference room Conference via handset And many more! 934 Flexor CTI Integration software 18.0000 20.00 Camrivox Snom IP Telephones http://www.voipon.co.uk/flexor-cti-integration-software-p-934.html http://www.voipon.co.uk/images/flexor_cti.jpg new Availability: In stock You can seamlessly link and integrate your On-Demand business process applications (such as Salesforce.com) with telephony using Flexor CTI software. This will give you click-to-dial, in-bound screen pop ups and on screen telephony call control resulting in improved customer retention and advanced business processes for your organisation. You will NOT require any additional servers as Flexor CTI software is simply and uniquely deployed via the web to your PC. CRM Telephony integration has traditionally relied upon large investments in infrastructure, hardware and support staff. With Flexor CTI software you will not require this. Instead, using Flexor, you can work within your existing environment making it simple to use; saving you time and hassle. And with Flexor’s unique web enabled deployment mechanism it means it’s simple to set up as well! No capital expenditure needed. No extensive training required. No mind boggling service contracts. Furthermore, by incorporating telephony within your CRM application you can deliver reports from your CRM application instead of having to purchase separate call logging tools that sit outside of your application all of which improves ROI. The Flexor Advantage Unifies CRM & Outlook applications with telephony Retain existing infrastructure – no new servers or infrastructure required Easy to install and maintain – software uniquely web-delivered Embeds telephony with the CRM application to improve efficiency, increase customer satisfaction and generate more orders Low cost (typically 80% cheaper than traditional integration routes) Scalable (from one to tens of thousands of users) Better, faster interaction and service delivery Better reporting - call logging and reporting with no new software required 935 Polycom Productivity Suite for SoundPoint IP Phones 11.0000 11.85 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-productivity-suite-for-soundpoint-ip-phones-p-935.html http://www.voipon.co.uk/images/polycom_productivity.gif new Availability: In stock Visual Conference Management Intuitive, visual control and management of a four-party conference call right from the phone’s display The industry’s first stand-alone visual conference management application on a SIP-based IP phone Visually setup and manage local, ad-hoc conference of up to four parties effortlessly Mitigate the complexity of using complicated * codes to manage conferencing features Eliminate the need for a conference bridge for small ad-hoc conference calls Corporate Directory Access Connect to business colleagues and partners faster Eliminate duplication of directories Save time and resources by managing contacts in one directory Reduce time in finding and dialing contacts Local Call Recording Instant recording of a live call to a standard external USB drive on the SoundPoint IP 650 The industry’s first local call recording application on a SIP-based IP phone Requires no special equipment or proprietary, server-based call recording system Enable the immediate recording of a call with an easy-to-use, visual interface Record all the calls in the highest audio quality for the ultimate playback experience Ideal for legal and regulatory record keeping, archiving, playback, and publishing Voice Quality Monitoring Proactive monitoring and troubleshooting of voice quality issues for faster problem resolution The industry’s first voice quality monitoring application that collects and transmits voice quality metrics directly from the phone in real time Replace many costly and complex solutions that require the use of network probes Improve IT productivity by enabling real-time troubleshooting and problem resolution Third Party Call Control Remotely control the phone and share presence information using a third party application Ideal for integrating the SoundPoint IP phone with unified communications systems and applications 936 EuroTech WINNER Follow Me Call Back FXO/FXS GSM Gateway 165.0000 183.00 EuroTech Communication EuroTech GSM Gateways http://www.voipon.co.uk/eurotech-winner-follow-me-call-back-fxofxs-gsm-gateway-p-936.html http://www.voipon.co.uk/images/eurotech_follow-me_gsm_sml.jpg new Availability: In Stock EuroTech's innovative WINNER Follow Me Call Back GSM Gateway is a gateway which directs calls via the LCR (Least Cost Route) for outgoing calls and ensures that they go out via the most cost effectve route (PSTN or GSM network.) With its FXS and FXO ports, the gateway can be used to connect analog telephone lines to the network, as well as analog phones. Along with other great features such as Follow-Me, Call-Back, Call-Through, SMS configuration and its PC connection - the WINNER Follow Me router is a versatile product with many advantages. Follow-Me The device can be configured to make incoming calls 'Follow' you to your mobile phone, so you can pick up calls anytime, anywhere, without changing your number - and saving the usually expensive cost of forwarding a call via the PSTN to mobile network. The Follow-Me feature routes the call to your mobile via its own SIM card(and the GSM network), meaning that you can take advantage of free minutes offers and many other great deals available on mobile phone tariffs. Call Back The user can call the gateway via cellular phone, and if the Caller ID of the user is an authorized number (configured in the gateway) then a busy tone is sent to the user and the call is disconnected. Then the gateway will immediately call them back using its own tariff and present the user with a dial tone. If the cellular number is not authorized then the call will be redirected to the PSTN or local line. A table of authorized numbers for the Call Back and Call Through can be updated via SMS messages. Key Features Intelligent LCR - LCR with up to 4 cellular networks LCR by prefix or timetable, day of week, hours to call and by number of calls FXO - For connection to the PSTN. FXS - For connection to analog phones. Follow Me - One number follow me from PSTN to Mobile Call Forwarder - From Mobile to local PBX or to PSTN / VOIP Call Back - From Mobile to PSTN / VOIP 2 SIMs - Supports 2 SIM cards for different tariffs. 937 VoIP Number Porting 48 48.00 Incoming Numbers http://www.voipon.co.uk/voip-number-porting-p-937.html http://www.voipon.co.uk/images/0102.gif new What is Number Porting? If you have an existing telephone number provided by BT, you can port the number across to VoIPon thus enjoying the benefits of VoIP without changing your number. Number Porting Benefits Do more with your number! Manage features online such as Call Divert, Voicemail, Caller ID On/Off, and Route your number to any type of VoIP destination! Integrate your current numbers with your VoIP Solution without using analogue-to-VoIP interfaces. Carry your number with you! Your number will always route to your location, regardless of where you are. Next time you move office/home, you don't have to make any arrangements regarding your number. Number Porting Costs Yearly Fee of £21.99 (may vary based on quantity of numbers and/or existing subscriptions) Setup Fee from £48 (+VAT) No further charges for receiving calls 938 PORTech MV-370 - 1 channel GSM/VoIP Gateway 130.0000 130.00 PORTech Communications PORTech VoIP GSM Gateway http://www.voipon.co.uk/portech-mv370-1-channel-gsmvoip-gateway-p-938.html http://www.voipon.co.uk/images/portech_mv-370_sml.jpg new Availability: In Stock The PORTech MV-370 is a single channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-370 supports 1 SIM card so supports one GSM channel at a time. The MV-370 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP ( such as ourselves ) and then onto SIP phones, analog phones, PSTN or a mobile phone. Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-370, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-370 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface. When an IP phone and the MV-370 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly. You can view the larger image for some examples of application of this device. Key Features: VoIP (SIP) - GSM conversion GSM - VoIP conversion Space for 50 mobile toLAN route settings Space for 50 LAN to mobile route settings Voice response for setting and status (dial in from mobile) Standard SIP protocol Full web browser configuration Send and receive SMS Specification Protocols: SIP (RFC2543, RFC3261) TCP/IP: IP/TCP/UDP/RTP/RTCP, CMP/ARP/RARP/SNTP, DHCP/DNS Client , IEEE802.1P/Q, ToS/Diffserv, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE Codec: G.711 u-Law/a-Law, G.723.1 (5.3k), G.723.1 (6.3k), G.729A, G.729A/B, Voice Quality, VAD, CNG, AEC, LEC, Packet loss GSM: BAND: EGSM 900/DCS 1800 MHz, Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR (Half Rate) Codec. Structure of the MV-370 939 PORTech MV-372 - 2 channel GSM/VoIP Gateway 240.0000 240.00 PORTech Communications PORTech VoIP GSM Gateway http://www.voipon.co.uk/portech-mv372-2-channel-gsmvoip-gateway-p-939.html http://www.voipon.co.uk/images/portech_mv-372_sml.jpg new Availability: In Stock The PORTech MV-372 is a dual channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-372 supports 2 SIM cards so supports two GSM channels at a time. It allows the users to make two simultaneous calls from IP phones to GSM or GSM to IP phones. The MV-372 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP ( such as ourselves ) and then onto SIP phones, analog phones, PSTN or a mobile phone. Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-372, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-372 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface. When an IP phone and the MV-372 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly. Key Features: VoIP (SIP) - GSM conversion GSM - VoIP conversion 2 Simultaneous GSM calls (with 2 SIMs inserted) Space for 50 mobile toLAN route settings Space for 50 LAN to mobile route settings Voice response for setting and status (dial in from mobile) Standard SIP protocol Full web browser configuration Send and receive SMS Specification Protocols: SIP (RFC2543, RFC3261) TCP/IP: IP/TCP/UDP/RTP/RTCP, CMP/ARP/RARP/SNTP, DHCP/DNS Client , IEEE802.1P/Q, ToS/Diffserv, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE Codec: G.711 u-Law/a-Law, G.723.1 (5.3k), G.723.1 (6.3k), G.729A, G.729A/B, Voice Quality, VAD, CNG, AEC, LEC, Packet loss GSM: Tri-band : 900/1800/1900 MHz 940 PORTech MV-374 - 4 channel GSM/VoIP Gateway 464.0000 513.00 PORTech Communications PORTech VoIP GSM Gateway http://www.voipon.co.uk/portech-mv374-4-channel-gsmvoip-gateway-p-940.html http://www.voipon.co.uk/images/portech_mv-374_sml.jpg new Availability: In Stock The PORTech MV-374 is a quad channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-374 supports 4 SIM cards so supports four GSM channels at a time. It allows the users to make four simultaneous calls from IP phones to GSM or GSM to IP phones. The MV-374 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP ( such as ourselves ) and then onto SIP phones, analog phones, PSTN or a mobile phone. Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-374, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-374 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface. When an IP phone and the MV-374 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly. Key Features: VoIP (SIP) - GSM conversion GSM - VoIP conversion 4 Simultaneous GSM calls (with 4 SIMs inserted) Space for 50 mobile toLAN route settings Space for 50 LAN to mobile route settings Voice response for setting and status (dial in from mobile) Standard SIP protocol Full web browser configuration Send and receive SMS Specification Protocols: SIP (RFC2543, RFC3261) TCP/IP: IP/TCP/UDP/RTP/RTCP, CMP/ARP/RARP/SNTP, DHCP/DNS Client , IEEE802.1P/Q, ToS/Diffserv, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE Codec: G.711 u-Law/a-Law, G.723.1 (5.3k), G.723.1 (6.3k), G.729A, G.729A/B, Voice Quality, VAD, CNG, AEC, LEC, Packet loss GSM: Tri-band : 900/1800/1900 MHz 941 PORTech MV-378 - 8 channel GSM/VoIP Gateway 864.0000 960.00 PORTech Communications PORTech VoIP GSM Gateway http://www.voipon.co.uk/portech-mv378-8-channel-gsmvoip-gateway-p-941.html http://www.voipon.co.uk/images/portech_mv-378_sml.jpg new Availability: In Stock The PORTech MV-378 is an 8 channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-378 supports 8 SIM cards so supports eight GSM channels at a time. It allows the users to make eight simultaneous calls from IP phones to GSM or GSM to IP phones. The MV-378 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP ( such as ourselves ) and then onto SIP phones, analog phones, PSTN or a mobile phone. Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-378, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-378 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface. When an IP phone and the MV-378 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly. Key Features: VoIP (SIP) - GSM conversion GSM - VoIP conversion 8 Simultaneous GSM calls (with 8 SIMs inserted) Space for 50 mobile toLAN route settings Space for 50 LAN to mobile route settings Voice response for setting and status (dial in from mobile) Standard SIP protocol Full web browser configuration Send and receive SMS Specification Protocols: SIP (RFC2543, RFC3261) TCP/IP: IP/TCP/UDP/RTP/RTCP, CMP/ARP/RARP/SNTP, DHCP/DNS Client , IEEE802.1P/Q, ToS/Diffserv, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE Codec: G.711 u-Law/a-Law, G.723.1 (5.3k), G.723.1 (6.3k), G.729A, G.729A/B, Voice Quality, VAD, CNG, AEC, LEC, Packet loss GSM: Tri-band : 900/1800/1900 MHz 942 PORTech MT-350 - 1 channel FXO/FXS/GSM Fixed Wireless Terminal 115.0000 128.00 PORTech Communications PORTech FWT (FXS,FXO,GSM) http://www.voipon.co.uk/portech-mt350-1-channel-fxofxsgsm-fixed-wireless-terminal-p-942.html http://www.voipon.co.uk/images/portech_mt-350_sml.jpg new Availability: In stock The PORTech MT-350 is a Fixed Wireless Terminal (FWT) which supports FXS, FXO and GSM interface - it can conenct to a PBX, the PSTN or a VoIP gateway. The device can connect to your analogue telephone line and divert calls received via PSTN to your mobile via its SIM card. Usually, call forwarding rates to mobiles via the PSTN are very expensive, meaning that - when out of the office - calls to the office analogue line would either have to be missed, or forwarded to your mobile at very expensive rates. The MT-350 keeps this option open, by using the SIM card you insert, the FWT can transfer all incoming calls via its FXO port to GSM, and route the calls directly to your mobile using very cheap, if not free, mobile-to-mobile call rates or tariffs. Many operators provide free unlimited minutes to mobiles on the same network for a cheap monthly rate, so if a SIM of the same network as your mobile is inserted into the device, these inclusive minutes can be used to route the call to your mobile, at no extra cost. The device also works the other way around, and can route incoming GSM calls to a PSTN number over the analogue line. Also, when a phone is connected via FXS, calls can be routed either by GSM or PSTN. The MT-350 can automatically check if the number dialed is on the network, and if this is the case, then the call will go out over the cheap GSM tariff (providing the user with 2 beeps to their handset to confirm); if not the call will go out over PSTN. Key Features Auto-select most economic route - The MT-350 automatically routes calls out over GSM or PSTN, depending on the destination. Polarity Reversing function Set GSM dial-out code amount Limit of communicating time GSM simulated echo-bell function Follow Me - PSTN to GSM and GSM to PSTN Specification Phone impendance : Below DC 1k&#937;&#65292;AC 600&#937;&#12290;(not include line) Phone Ring Output Voltage : 45v rms Phone Signal (DTMF) Standard : -3dbm&#65374;-24dbm Phone Signal (DTMF)Time &#65306; 50&#65374;100ms DTMF Frequency Rate Error &#65306; ±1&#65285; Feeder Voltage &#65306; 48v AC Adapter &#65306; Input 110V AC, Output 12V DC 1A GSM Specification: GSM Frequency bands : Dual Band EGSM 900 and GSM 1800(GSM Phase 2+) GSM class : Small MS Transmit power : Class 4(2W) for EGSM 900 Class 1(1W) for GSM 1800 SIM card reader : External – connected via interface connector Antenna : 50 Ohm antenna coaxial connector Temperature range : Normal operation : -20&#8451; to +55&#8451; Restricted operation : -25&#8451; to -20&#8451; and +55&#8451; to 70&#8451; Storage : -40&#8451; to +85&#8451; 943 PORTech MT-350S - 1 channel FXO/FXS/GSM Fixed Wireless Terminal (SMS support) 126.0000 140.00 PORTech Communications PORTech FWT (FXS,FXO,GSM) http://www.voipon.co.uk/portech-mt350s-1-channel-fxofxsgsm-fixed-wireless-terminal-sms-support-p-943.html http://www.voipon.co.uk/images/portech_mt-350_sml.jpg new Availability: In stock The PORTech MT-350S is a Fixed Wireless Terminal (FWT) which supports FXS, FXO and GSM interface - it can conenct to a PBX, the PSTN or a VoIP gateway. The MT-350S version is the same as the MT-350 except it also supports the sending and receiving of SMS via teh web-browser interface. The device can connect to your analogue telephone line and divert calls received via PSTN to your mobile via its SIM card. Usually, call forwarding rates to mobiles via the PSTN are very expensive, meaning that - when out of the office - calls to the office analogue line would either have to be missed, or forwarded to your mobile at very expensive rates. The MT-350S keeps this option open, by using the SIM card you insert, the FWT can transfer all incoming calls via its FXO port to GSM, and route the calls directly to your mobile using very cheap, if not free, mobile-to-mobile call rates or tariffs. Many operators provide free unlimited minutes to mobiles on the same network for a cheap monthly rate, so if a SIM of the same network as your mobile is inserted into the device, these inclusive minutes can be used to route the call to your mobile, at no extra cost. The device also works the other way around, and can route incoming GSM calls to a PSTN number over the analogue line. Also, when a phone is connected via FXS, calls can be routed either by GSM or PSTN. The MT-350S can automatically check if the number dialed is on the network, and if this is the case, then the call will go out over the cheap GSM tariff (providing the user with 2 beeps to their handset to confirm); if not the call will go out over PSTN. Key Features Auto-select most economic route - The MT-350S automatically routes calls out over GSM or PSTN, depending on the destination. SMS support - SMS messages can be sent and received via the MT-350S, this features is accessed via the web-browser based interface. Polarity Reversing function Set GSM dial-out code amount Limit of communicating time GSM simulated echo-bell function Follow Me - PSTN to GSM and GSM to PSTN Specification Phone impendance : Below DC 1k&#937;&#65292;AC 600&#937;&#12290;(not include line) Phone Ring Output Voltage : 45v rms Phone Signal (DTMF) Standard : -3dbm&#65374;-24dbm Phone Signal (DTMF)Time &#65306; 50&#65374;100ms DTMF Frequency Rate Error &#65306; ±1&#65285; Feeder Voltage &#65306; 48v AC Adapter &#65306; Input 110V AC, Output 12V DC 1A GSM Specification: GSM Frequency bands : Dual Band EGSM 900 and GSM 1800(GSM Phase 2+) GSM class : Small MS Transmit power : Class 4(2W) for EGSM 900 Class 1(1W) for GSM 1800 SIM card reader : External – connected via interface connector Antenna : 50 Ohm antenna coaxial connector Temperature range : Normal operation : -20&#8451; to +55&#8451; Restricted operation : -25&#8451; to -20&#8451; and +55&#8451; to 70&#8451; Storage : -40&#8451; to +85&#8451; 944 PORTech MT-354 - 4 channel FXO/FXS/GSM Fixed Wireless Terminal 389.0000 434.00 PORTech Communications PORTech FWT (FXS,FXO,GSM) http://www.voipon.co.uk/portech-mt354-4-channel-fxofxsgsm-fixed-wireless-terminal-p-944.html http://www.voipon.co.uk/images/portech_mt-354_sml.jpg new Availability: In stock The PORTech MT-354 is a Fixed Wireless Terminal (FWT) which supports FXS, FXO and GSM interface - it can conenct to a PBX, the PSTN or a VoIP gateway. The MT-354 supports 4 GSM channels at a time, meaning that upto 4 calls can be passing through the device at any one time when 4 SIM cards are inserted. The device can connect to your analogue telephone line and divert calls received via PSTN to your mobile via its SIM card. Usually, call forwarding rates to mobiles via the PSTN are very expensive, meaning that - when out of the office - calls to the office analogue line would either have to be missed, or forwarded to your mobile at very expensive rates. The MT-354 keeps this option open, by using the SIM card you insert, the FWT can transfer all incoming calls via its FXO port to GSM, and route the calls directly to your mobile using very cheap, if not free, mobile-to-mobile call rates or tariffs. Many operators provide free unlimited minutes to mobiles on the same network for a cheap monthly rate, so if a SIM of the same network as your mobile is inserted into the device, these inclusive minutes can be used to route the call to your mobile, at no extra cost. The device also works the other way around, and can route incoming GSM calls to a PSTN number over the analogue line. Also, when a phone is connected via FXS, calls can be routed either by GSM or PSTN. The MT-354 can automatically check if the number dialed is on the network, and if this is the case, then the call will go out over the cheap GSM tariff (providing the user with 2 beeps to their handset to confirm); if not the call will go out over PSTN. Key Features Auto-select most economic route - The MT-354 automatically routes calls out over GSM or PSTN, depending on the destination. 4 Channel - Up to 4 SIM cards allow for four GSM channels to be active at a time. Polarity Reversing function Set GSM dial-out code amount Limit of communicating time GSM simulated echo-bell function Follow Me - PSTN to GSM and GSM to PSTN Specification Phone impendance : Below DC 1k&#937;&#65292;AC 600&#937;&#12290;(not include line) Phone Ring Output Voltage : 45v rms Phone Signal (DTMF) Standard : -3dbm&#65374;-24dbm Phone Signal (DTMF)Time &#65306; 50&#65374;100ms DTMF Frequency Rate Error &#65306; ±1&#65285; Feeder Voltage &#65306; 48v AC Adapter &#65306; Input 110V AC, Output 12V DC 1A GSM Specification: GSM Frequency bands : Dual Band EGSM 900 and GSM 1800(GSM Phase 2+) GSM class : Small MS Transmit power : Class 4(2W) for EGSM 900 Class 1(1W) for GSM 1800 SIM card reader : External – connected via interface connector Antenna : 50 Ohm antenna coaxial connector Temperature range : Normal operation : -20&#8451; to +55&#8451; Restricted operation : -25&#8451; to -20&#8451; and +55&#8451; to 70&#8451; Storage : -40&#8451; to +85&#8451; 945 PORTech MT-358 - 8 channel FXO/FXS/GSM Fixed Wireless Terminal 759.0000 845.00 PORTech Communications PORTech FWT (FXS,FXO,GSM) http://www.voipon.co.uk/portech-mt358-8-channel-fxofxsgsm-fixed-wireless-terminal-p-945.html http://www.voipon.co.uk/images/portech_mt-354_sml.jpg new Availability: In stock The PORTech MT-358 is a Fixed Wireless Terminal (FWT) which supports FXS, FXO and GSM interface - it can conenct to a PBX, the PSTN or a VoIP gateway. The MT-358 supports 8 GSM channels at a time, meaning that upto 8 calls can be passing through the device at any one time when 8 SIM cards are inserted. The device can connect to your analogue telephone line and divert calls received via PSTN to your mobile via its SIM card. Usually, call forwarding rates to mobiles via the PSTN are very expensive, meaning that - when out of the office - calls to the office analogue line would either have to be missed, or forwarded to your mobile at very expensive rates. The MT-358 keeps this option open, by using the SIM card you insert, the FWT can transfer all incoming calls via its FXO port to GSM, and route the calls directly to your mobile using very cheap, if not free, mobile-to-mobile call rates or tariffs. Many operators provide free unlimited minutes to mobiles on the same network for a cheap monthly rate, so if a SIM of the same network as your mobile is inserted into the device, these inclusive minutes can be used to route the call to your mobile, at no extra cost. The device also works the other way around, and can route incoming GSM calls to a PSTN number over the analogue line. Also, when a phone is connected via FXS, calls can be routed either by GSM or PSTN. The MT-358 can automatically check if the number dialed is on the network, and if this is the case, then the call will go out over the cheap GSM tariff (providing the user with 2 beeps to their handset to confirm); if not the call will go out over PSTN. Key Features Auto-select most economic route - The MT-358 automatically routes calls out over GSM or PSTN, depending on the destination. 8 Channel - Up to 8 SIM cards allow for eight GSM channels to be active at a time. Polarity Reversing function Set GSM dial-out code amount Limit of communicating time GSM simulated echo-bell function Follow Me - PSTN to GSM and GSM to PSTN Specification Phone impendance : Below DC 1k&#937;&#65292;AC 600&#937;&#12290;(not include line) Phone Ring Output Voltage : 45v rms Phone Signal (DTMF) Standard : -3dbm&#65374;-24dbm Phone Signal (DTMF)Time &#65306; 50&#65374;100ms DTMF Frequency Rate Error &#65306; ±1&#65285; Feeder Voltage &#65306; 48v AC Adapter &#65306; Input 110V AC, Output 12V DC 1A GSM Specification: GSM Frequency bands : Dual Band EGSM 900 and GSM 1800(GSM Phase 2+) GSM class : Small MS Transmit power : Class 4(2W) for EGSM 900 Class 1(1W) for GSM 1800 SIM card reader : External – connected via interface connector Antenna : 50 Ohm antenna coaxial connector Temperature range : Normal operation : -20&#8451; to +55&#8451; Restricted operation : -25&#8451; to -20&#8451; and +55&#8451; to 70&#8451; Storage : -40&#8451; to +85&#8451; 946 Grandstream GXP 280 IP Telephone 49 49.00 Grandstream Grandstream IP Phones http://www.voipon.co.uk/grandstream-gxp-280-ip-telephone-p-946.html http://www.voipon.co.uk/images/grandstream_gxp-280_sml.jpg new Availability: Mid July The GXP280 is a next generation small business SIP phone that features 1 line appearance, 128x32 graphic LCD, 3 XML programmable softkeys, and dual 10M/100Mbps network ports. It is an affordable choice for a SOHO user needing a feature rich, 1-line IP phone. The GXP280 delivers superior audio quality, comprehensive telephony features, automated provisioning, security protection for privacy and broad interoperability with most 3rd party SIP devices, gateways and leading soft-switch platforms. The product has 1 line appearance with FLASH to handle 2 simultaneous calls, offers support for multiple languages (including English, Spanish, German, French, etc) and headset jacks (both 2.5mm and RJ9 jack). Like the other models in the GXP IP phone series, the product is equipped with high fidelity wideband audio and a full-duplex speakerphone with advanced acoustic echo cancellation. The GXP280 is outfitted with SRTP, is wall mountable, and can also be automatically provisioned for mass deployment. Key Features 128 x 32 pixel graphical LCD with support for multiple languages 1 line appearance with FLASH to handle up to 2 simultaneous calls 3 XML programmable context sensitive soft keys, 3 way conference High fidelity wideband audio, full-duplex speakerphone with advanced acoustic echo cancellation Dual switched auto sensing 10/100 Mbps network ports Automated provisioning for mass deployment, SRTP Detailed Specefication Protocols/Standards: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, PPPoE, TFTP, NTP, STUN, SIMPLE Network Interfaces: Dual switched 10/100Mbps Graphic Display: 128 x 32 pixel LCD Voice codec: Support for G.723.1, G.729 A/B, G.711 u/a-law, G.726, G.722 (wide band), GSM and iLBC, in-band and out-of-band DTMF Telephony features : Hold, Mute, Transfer, Forward, 3-way conference, Downloadable phone book (XML, LDAP to to 200 contacts), XML customisation of screen, call waiting, call log, off-hook auto dial, auto answer, click-to-dial, downloadable ringtones, server redundancy and fail-over support. Headset connection : 2.5mm RJ9 headset jack Wall mountable : Yes QoS : Layer 2 (802.1Q, 801.2p) and Layer 3(ToS, DiffServ, MPLS) QoS. Security : User and admin level passwords, MD5 and MD5- sess based authentication, AES based secure config file, SRTP. Upgrade/Provisioning : Fireware upgrade via TFTP/HTTP, mass deployment using central secure provisioning file. Power : Universal power adapter (Input 100-240V AC 50-60Hz, Output +5V DC, 1.2A, UL Certified) included. Physical : Dimensions: 168mm x 200mm x 89.5mm Unit weight: 0.62kg Package weight: 0.725kg Temperature/humidity: 0-40 degrees C (non-condensing) Compliance : FCC, CE, C-Tick   948 Draytek Vigor2820 ADSL/Cable/3G Router Firewall 130 130.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2820-adslcable3g-router-firewall-p-948.html http://www.voipon.co.uk/images/draytek_2820_sml.jpg new Availability: In stock Vigor 2820 Series ADSL Router Firewall The Vigor 2820 series is DrayTek's latest premium ADSL and broadband router/firewall family, featuring an entirely new stylish case design and new DrayOS 3 operating system. Packed with features, the Vigor 2820 offers truly comprehensive ADSL connetivity and security. Compatible with all variants of ADSL (include ADSLMax and ADSL2+) the Vigor 2820 can also be used for cable-modem and leased line applications thanks to its additional WAN port (Ethernet). A Gigabit Ethernet port on the WAN side provides high speed connectivity for your server (or uplink to a larger Ethernet Switch). Security features include content filtering, web application controls and a new object based firewall management system. Robust & Comprehensive Firewall As with previous DrayTek firewalls, security is taken seriously on the Vigor 2820 Series. The firewall features measures for protection against attacks including DoS (Denial of Service) attacks, IP-based attacks and access by unauthorised remote systems. Wireless, Ethernet and VPN are also protected by various protection systems (see later). A new DrayTek object-based firewall allows even more setup flexibility than ever, enabling you to create combinations of users, rules and restrictions to suit multi-departmental organisations. Content control features of the firewall allow you to set restrictions on web site access, blocking download of certain file types, blocking specific web sites, blocking IM/P2P applications or other potentially harmful or wasteful content. Using the SurfControl site categorisations enable you to block whole categories of web sites (e.g. gambling, pornography etc.), subject to subscription Dual-WAN Load Balancing & Backup As well as the primary ADSL interface, the Vigor 2820 features a secondary WAN port for Internet access. This is an Ethernet interface and can connect to a second ADSL modem, cable modem or any other Ethernet-based Internet feed. The secondary interface can be used either for WAN-Backup or load balancing . WAN-Backup provides contingenry (redunancy) in case of your primary ADSL line or ISP suffering temporary outage). Internet Traffic will be temporarily routed via the secondary Internet access. When normal services is restored to your primary ADSL line, all traffic is switch back to that. If you don't have ADSL, the Ethernet WAN port can instead be used as your primary/only Internet connection (using NAT) so the same router can be used for either ADSL or Ethernet Internet connections. The USB port provides an alternative connection method ( Firmware Upgradable ) for Internet backup by connecting to a compatible USB modem (or cellphone) for access to the high speed 3G cellular networks from UK providers such as Vodafone, Orange, 3 and T-Mobile. If you don't have ADSL at all, the USB/3G access method can be used as your primary/only Internet connection, ideal for temporary locations, mobile applications or where broadband access is not available. Note : For WAN2 you can use either a Ethernet or USB 3G connection but not both at the same time.   Detailed Specification: Physical Interfaces: LAN Ports (Switch) 1 X Gigabit Ethernet (1000Mb/s) Ports 3 X Megabit (100Mb/s) Ports Port-Based VLAN (Inclusive/Exclusive Groups) WAN Ports: ADSL Port Compliant with: ANSI T1.413 Issue2 ITU-T G.992.1 G.dmt (ADSL) ITU-T G.992.2 G.lite ITU-T G.992.3 ADSL2 ITU-T G.992.5 ADSL2+ Annex L (READSL) Annex M Secondary WAN Port : 10/100 Base-TX Ethernet for load balance and WAN failover USB Port for 3G Cellular Modem or Printer VoIP: 2-port FXS Phone Ports (Vigor2820 'V' models only) Load Balance/Failover Features: Outbound Policy-Based Load-Balance WAN Connection Fail-over BoD (Bandwidth on Demand) Vigor 2820 ATM Protocols (DSL): RFC-2684/RFC-1483 Multiple Protocol over AAL5 RFC-2516 PPP over Ethernet RFC-2364 PPP over AAL5 PPPoE pass through LAN/WLAN PPPoE/PPPoA Relay Transparent bridge for MPoA Multiple PVC support for Triple Play Applications (up to 8 simultaneous) Wireless LAN Features ('n' Models Only): 802.11n Compliant (Draft 2.0) Latest 'MIMO' Technology with three aerials (2T3R) Multiple SSID : Create up to 4 virtual wireless LANs (independent or joined) Packet Aggregation and Channel Bonding Compatible with 802.11b and 802.11g Standards Active Client list in Web Interface Wireless LAN Isolation (from VLAN groups and wired Ethernet interfaces) 64/128-bit WEP Encryption WPA/WPA2 Encryption Switchable Hidden SSID Restricted access list for clients (by MAC address) Time Scheduling (WLAN can be disabled at certain times of day) Access Point Discovery WDS (Wireless Distribution system) for WLAN Bridging and Repeating (Firmware Upgradable) 802.1x Radius Authentication Wireless Rate-Control Automatic Power Management 802.11e WMM (Wi-Fi Multimedia) VoIP Features (Vigor2820 'V' Models only): Protocols: SIP, RTP / RTCP 12 SIP Registrar Accounts (for up to 12 VoIP providers) Line port for PSTN Passthrough (integrates POTS line) Auto-fallback to PSTN under power/Internet failure G.168 Line Echo-cancellation Automatic Gain Control Jitter Buffer ( 125ms ) Voice Codecs: G.711 A / µ Law G.723.1 G.726 G.729 A / B VAD / CNG Tone Generation: DTMF , Dial , Busy , Ring Back , Call Progress DTMF Transmission: In Band / Out Band ( RFC-2833 ) / SIP info FAX / Modem Support G.711 Pass-through T.38 for FAX Supplemental Services: Caller ID Call Hold / Retrieve Call Waiting Call Waiting with Caller ID Call Transfer Call Forwarding ( Always , On Busy and On No Answer ) DND (Do not Disturb) Call Barring ( Incoming / Outgoing ) MWI ( Message Waiting Indicator ) ( RFC-3842 ) Hotline (Dial preset number when handset lifted) WAN Protocols (Ethernet): DHCP Client Static IP PPPoE PPTP L2TP * BPA Firewall & Security Features: CSM (Content Security Management): URL Keyword Filtering - Whitelist or Blacklist specific sites or keywords in URLs Surfcontrol Support - Block Web sites by category (subject to subscription) Prevent accessing of web sites by using their direct IP address (thus URLs only) Blocking automatic download of Java applets and ActiveX controls Blocking of web site cookies Block http downloads of file types : Binary Executable : .EXE / .COM / .BAT / .SCR / .PIF Compressed : .ZIP / .SIT / .ARC / .CAB/. ARJ / .RAR Multimedia : .MOV / .MP3 / .MPEG / .MPG / .WMV / .WAV / .RAM / .RA / .RM / .AVI / .AU Time Schedules for enabling/disabling the restrictions Block P2P (Peer-to-Peer) file sharing programs (e.g. Kazza, WinMX etc. ) Block Instant Messaging programs (e.g. IRC, MSN/Yahoo Messenger etc.) Multi-NAT, DMZ Host Port Redirection and Open Port Configuration Policy-Based Firewall MAC Address Filter SPI ( Stateful Packet Inspection ) with new FlowTrack Mechanism DoS / DDoS Protection IP Address Anti-spoofing E-Mail Alert and Logging via Syslog Bind IP to MAC Address Bandwidth Management: QoS Guaranteed Bandwidth for VoIP Class-based Bandwidth Guarantee by User-Defined Traffic Categories DiffServ Code Point Classifying 4-level Priority for each Direction (Inbound / Outbound) Bandwidth Borrowed Temporary (5 minute) Quick Blocking of any LAN Client Bandwidth / Session Limitation Network/Router Management: Web-Based User Interface (HTTP / HTTPS) CLI ( Command Line Interface ) / Telnet / SSH* Administration Access Control Configuration Backup / Restore Built-in Diagnostic Function Firmware Upgrade via TFTP / FTP Logging via Syslog SNMP Management with MIB-II TR-069 TR-104 VPN Facilities: Up to 32 Concurrent VPN Tunnels (incoming or outgoing) Tunnelling Protocols: PPTP, IPSec, L2TP, L2TP over IPSec IPSec Main and Agressive modes Encryption : MPPE and Hardware-Based AES / DES / 3DES Authentication : Hardware-Based MD5 and SHA-1 IKE Authentication : Pre-shared Key and X.509 Digital Signature LAN-to-LAN & Teleworker-to-LAN connectivity DHCP over IPSec NAT-Traversal ( NAT-T ) Dead Peer Detection (DPD) VPN Pass-Through Network Features: DHCP Client / Relay / Server Dynamic DNS NTP Client (Syncrhonise Router Time) Call Scheduling (Enable/Trigger Internet Access by Time) RADIUS Client DNS Cache / Proxy Microsoft™ UPnP Support Routing Protocols: Static Routing RIP V2 Operating Requirements: Rack Mountable (Optional Vigor 2820 mounting bracket required) Wall Mountable Temperature Operating : 0°C ~ 45°C Storage : -25°C ~ 70°C Humidity 10% ~ 90% (non-condensing) Power Consumption: 18 Watt Max. Dimensions: L240.96 * W165.07 * H43.96 ( mm ) Operating Power: DC 15V (via external PSU, supplied) Warranty : Two (2) Years RTB Power Requirements : 220-240VAC   949 Draytek Vigor2820n ADSL/Cable/3G Router Firewall (Bundled with Vigor N61) 168 168.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2820n-adslcable3g-router-firewall-bundled-with-vigor-n61-p-949.html http://www.voipon.co.uk/images/draytek_2820n_sml.jpg new Availability: In stock Vigor 2820 Series ADSL Router Firewall The Vigor 2820 series is DrayTek's latest premium ADSL and broadband router/firewall family, featuring an entirely new stylish case design and new DrayOS 3 operating system. Packed with features, the Vigor 2820 offers truly comprehensive ADSL connetivity and security. Compatible with all variants of ADSL (include ADSLMax and ADSL2+) the Vigor 2820 can also be used for cable-modem and leased line applications thanks to its additional WAN port (Ethernet). A Gigabit Ethernet port on the WAN side provides high speed connectivity for your server (or uplink to a larger Ethernet Switch). Security features include content filtering, web application controls and a new object based firewall management system. This comes bundles with the DrayTek N61 USB wireless LAN adapter (supports 802.11b/g/n Draft 2.0) The DrayTek Vigor N61 is a USB-based wireless LAN adapter supporting the latest 802.11n Draft 2.0 standard, as well as legacy support for 802.11g and 802.11b standards. 802.11n Draft 2.0 provides increased speed and greater resilience than previous wireless standards. Robust & Comprehensive Firewall As with previous DrayTek firewalls, security is taken seriously on the Vigor 2820 Series. The firewall features measures for protection against attacks including DoS (Denial of Service) attacks, IP-based attacks and access by unauthorised remote systems. Wireless, Ethernet and VPN are also protected by various protection systems (see later). A new DrayTek object-based firewall allows even more setup flexibility than ever, enabling you to create combinations of users, rules and restrictions to suit multi-departmental organisations. Content control features of the firewall allow you to set restrictions on web site access, blocking download of certain file types, blocking specific web sites, blocking IM/P2P applications or other potentially harmful or wasteful content. Using the SurfControl site categorisations enable you to block whole categories of web sites (e.g. gambling, pornography etc.), subject to subscription Dual-WAN Load Balancing & Backup As well as the primary ADSL interface, the Vigor 2820 features a secondary WAN port for Internet access. This is an Ethernet interface and can connect to a second ADSL modem, cable modem or any other Ethernet-based Internet feed. The secondary interface can be used either for WAN-Backup or load balancing . WAN-Backup provides contingenry (redunancy) in case of your primary ADSL line or ISP suffering temporary outage). Internet Traffic will be temporarily routed via the secondary Internet access. When normal services is restored to your primary ADSL line, all traffic is switch back to that. If you don't have ADSL, the Ethernet WAN port can instead be used as your primary/only Internet connection (using NAT) so the same router can be used for either ADSL or Ethernet Internet connections. The USB port provides an alternative connection method ( Firmware Upgradable ) for Internet backup by connecting to a compatible USB modem (or cellphone) for access to the high speed 3G cellular networks from UK providers such as Vodafone, Orange, 3 and T-Mobile. If you don't have ADSL at all, the USB/3G access method can be used as your primary/only Internet connection, ideal for temporary locations, mobile applications or where broadband access is not available. Note : For WAN2 you can use either a Ethernet or USB 3G connection but not both at the same time. Wireless LAN ('n' models only) The Vigor2820 Series features the latest 802.11n Draft 2.0 wireless LAN specification and has been certified by the WiFi alliance for cross compatibility and WiFi compliance (including WPA/WPA2 and WMM). 802.11n Draft 2.0 provides a total wireless bandwidth of up to 300Mb/s using new methods such as packet aggregation and channel bonding. Throughput depends on your own environment (factors such as obstructions, number of hosts and distance all make a significant difference), but actual transfer speeds of 100Mb/s are achievable (based on our real world tests). In addition, 802.11n Draft 2.0 provides greater coverage and resilience to interference compared to previous wireless standards thanks to the MIMO technology and the Vigor's triple-antennae diversity arrangement. This offset arrangement of aerials provides offset paths between hosts so that interference can be overcome. Wireless Security is comprehensive too; the Vigor 2820 Series provides several independent levels of security including encryption (up to WPA2), authentication (802.11x) and methods such as MAC address locking and DHCP fixing to restrict access to authorised users only. The Web interface lets you see how many and which clients are currently connected as well as their current bandwidth usage. An 'instant' block lets you disconnect a wireless user temporarily in case of query. The Wireless VLAN facility allows you to isolate wireless clients from each other or from the 'wired' LAN. The Multiple SSID features enables you to have up to four distinct or common virtual wireless access points. For example, you could have one for company usage, with access to your company LAN and another for public access which allows internet surfing only. Setting up wireless security is made easier thanks to the WPS feature (WiFi protected setup) whereby your client PC can get it's security keys by pressing a button on the front of the router.   Detailed Specification: Physical Interfaces: LAN Ports (Switch) 1 X Gigabit Ethernet (1000Mb/s) Ports 3 X Megabit (100Mb/s) Ports Port-Based VLAN (Inclusive/Exclusive Groups) WAN Ports: ADSL Port Compliant with: ANSI T1.413 Issue2 ITU-T G.992.1 G.dmt (ADSL) ITU-T G.992.2 G.lite ITU-T G.992.3 ADSL2 ITU-T G.992.5 ADSL2+ Annex L (READSL) Annex M Secondary WAN Port : 10/100 Base-TX Ethernet for load balance and WAN failover USB Port for 3G Cellular Modem or Printer VoIP: 2-port FXS Phone Ports (Vigor2820 'V' models only) Load Balance/Failover Features: Outbound Policy-Based Load-Balance WAN Connection Fail-over BoD (Bandwidth on Demand) Vigor 2820 ATM Protocols (DSL): RFC-2684/RFC-1483 Multiple Protocol over AAL5 RFC-2516 PPP over Ethernet RFC-2364 PPP over AAL5 PPPoE pass through LAN/WLAN PPPoE/PPPoA Relay Transparent bridge for MPoA Multiple PVC support for Triple Play Applications (up to 8 simultaneous) Wireless LAN Features ('n' Models Only): 802.11n Compliant (Draft 2.0) Latest 'MIMO' Technology with three aerials (2T3R) Multiple SSID : Create up to 4 virtual wireless LANs (independent or joined) Packet Aggregation and Channel Bonding Compatible with 802.11b and 802.11g Standards Active Client list in Web Interface Wireless LAN Isolation (from VLAN groups and wired Ethernet interfaces) 64/128-bit WEP Encryption WPA/WPA2 Encryption Switchable Hidden SSID Restricted access list for clients (by MAC address) Time Scheduling (WLAN can be disabled at certain times of day) Access Point Discovery WDS (Wireless Distribution system) for WLAN Bridging and Repeating (Firmware Upgradable) 802.1x Radius Authentication Wireless Rate-Control Automatic Power Management 802.11e WMM (Wi-Fi Multimedia) VoIP Features (Vigor2820 'V' Models only): Protocols: SIP, RTP / RTCP 12 SIP Registrar Accounts (for up to 12 VoIP providers) Line port for PSTN Passthrough (integrates POTS line) Auto-fallback to PSTN under power/Internet failure G.168 Line Echo-cancellation Automatic Gain Control Jitter Buffer ( 125ms ) Voice Codecs: G.711 A / µ Law G.723.1 G.726 G.729 A / B VAD / CNG Tone Generation: DTMF , Dial , Busy , Ring Back , Call Progress DTMF Transmission: In Band / Out Band ( RFC-2833 ) / SIP info FAX / Modem Support G.711 Pass-through T.38 for FAX Supplemental Services: Caller ID Call Hold / Retrieve Call Waiting Call Waiting with Caller ID Call Transfer Call Forwarding ( Always , On Busy and On No Answer ) DND (Do not Disturb) Call Barring ( Incoming / Outgoing ) MWI ( Message Waiting Indicator ) ( RFC-3842 ) Hotline (Dial preset number when handset lifted) WAN Protocols (Ethernet): DHCP Client Static IP PPPoE PPTP L2TP * BPA Firewall & Security Features: CSM (Content Security Management): URL Keyword Filtering - Whitelist or Blacklist specific sites or keywords in URLs Surfcontrol Support - Block Web sites by category (subject to subscription) Prevent accessing of web sites by using their direct IP address (thus URLs only) Blocking automatic download of Java applets and ActiveX controls Blocking of web site cookies Block http downloads of file types : Binary Executable : .EXE / .COM / .BAT / .SCR / .PIF Compressed : .ZIP / .SIT / .ARC / .CAB/. ARJ / .RAR Multimedia : .MOV / .MP3 / .MPEG / .MPG / .WMV / .WAV / .RAM / .RA / .RM / .AVI / .AU Time Schedules for enabling/disabling the restrictions Block P2P (Peer-to-Peer) file sharing programs (e.g. Kazza, WinMX etc. ) Block Instant Messaging programs (e.g. IRC, MSN/Yahoo Messenger etc.) Multi-NAT, DMZ Host Port Redirection and Open Port Configuration Policy-Based Firewall MAC Address Filter SPI ( Stateful Packet Inspection ) with new FlowTrack Mechanism DoS / DDoS Protection IP Address Anti-spoofing E-Mail Alert and Logging via Syslog Bind IP to MAC Address Bandwidth Management: QoS Guaranteed Bandwidth for VoIP Class-based Bandwidth Guarantee by User-Defined Traffic Categories DiffServ Code Point Classifying 4-level Priority for each Direction (Inbound / Outbound) Bandwidth Borrowed Temporary (5 minute) Quick Blocking of any LAN Client Bandwidth / Session Limitation Network/Router Management: Web-Based User Interface (HTTP / HTTPS) CLI ( Command Line Interface ) / Telnet / SSH* Administration Access Control Configuration Backup / Restore Built-in Diagnostic Function Firmware Upgrade via TFTP / FTP Logging via Syslog SNMP Management with MIB-II TR-069 TR-104 VPN Facilities: Up to 32 Concurrent VPN Tunnels (incoming or outgoing) Tunnelling Protocols: PPTP, IPSec, L2TP, L2TP over IPSec IPSec Main and Agressive modes Encryption : MPPE and Hardware-Based AES / DES / 3DES Authentication : Hardware-Based MD5 and SHA-1 IKE Authentication : Pre-shared Key and X.509 Digital Signature LAN-to-LAN & Teleworker-to-LAN connectivity DHCP over IPSec NAT-Traversal ( NAT-T ) Dead Peer Detection (DPD) VPN Pass-Through Network Features: DHCP Client / Relay / Server Dynamic DNS NTP Client (Syncrhonise Router Time) Call Scheduling (Enable/Trigger Internet Access by Time) RADIUS Client DNS Cache / Proxy Microsoft™ UPnP Support Routing Protocols: Static Routing RIP V2 Operating Requirements: Rack Mountable (Optional Vigor 2820 mounting bracket required) Wall Mountable Temperature Operating : 0°C ~ 45°C Storage : -25°C ~ 70°C Humidity 10% ~ 90% (non-condensing) Power Consumption: 18 Watt Max. Dimensions: L240.96 * W165.07 * H43.96 ( mm ) Operating Power: DC 15V (via external PSU, supplied) Warranty : Two (2) Years RTB Power Requirements : 220-240VAC   950 Draytek Vigor2820Vn ADSL/Cable/3G Router Firewall 176 176.00 Draytek Draytek Routers http://www.voipon.co.uk/draytek-vigor2820vn-adslcable3g-router-firewall-p-950.html http://www.voipon.co.uk/images/draytek_2820vn_sml.jpg new Availability: In stock Vigor 2820 Series ADSL Router Firewall The Vigor 2820 series is DrayTek's latest premium ADSL and broadband router/firewall family, featuring an entirely new stylish case design and new DrayOS 3 operating system. Packed with features, the Vigor 2820 offers truly comprehensive ADSL connetivity and security. Compatible with all variants of ADSL (include ADSLMax and ADSL2+) the Vigor 2820 can also be used for cable-modem and leased line applications thanks to its additional WAN port (Ethernet). A Gigabit Ethernet port on the WAN side provides high speed connectivity for your server (or uplink to a larger Ethernet Switch). Security features include content filtering, web application controls and a new object based firewall management system. Robust & Comprehensive Firewall As with previous DrayTek firewalls, security is taken seriously on the Vigor 2820 Series. The firewall features measures for protection against attacks including DoS (Denial of Service) attacks, IP-based attacks and access by unauthorised remote systems. Wireless, Ethernet and VPN are also protected by various protection systems (see later). A new DrayTek object-based firewall allows even more setup flexibility than ever, enabling you to create combinations of users, rules and restrictions to suit multi-departmental organisations. Content control features of the firewall allow you to set restrictions on web site access, blocking download of certain file types, blocking specific web sites, blocking IM/P2P applications or other potentially harmful or wasteful content. Using the SurfControl site categorisations enable you to block whole categories of web sites (e.g. gambling, pornography etc.), subject to subscription Dual-WAN Load Balancing & Backup As well as the primary ADSL interface, the Vigor 2820 features a secondary WAN port for Internet access. This is an Ethernet interface and can connect to a second ADSL modem, cable modem or any other Ethernet-based Internet feed. The secondary interface can be used either for WAN-Backup or load balancing . WAN-Backup provides contingenry (redunancy) in case of your primary ADSL line or ISP suffering temporary outage). Internet Traffic will be temporarily routed via the secondary Internet access. When normal services is restored to your primary ADSL line, all traffic is switch back to that. If you don't have ADSL, the Ethernet WAN port can instead be used as your primary/only Internet connection (using NAT) so the same router can be used for either ADSL or Ethernet Internet connections. The USB port provides an alternative connection method ( Firmware Upgradable ) for Internet backup by connecting to a compatible USB modem (or cellphone) for access to the high speed 3G cellular networks from UK providers such as Vodafone, Orange, 3 and T-Mobile. If you don't have ADSL at all, the USB/3G access method can be used as your primary/only Internet connection, ideal for temporary locations, mobile applications or where broadband access is not available. Note : For WAN2 you can use either a Ethernet or USB 3G connection but not both at the same time. Wireless LAN ('n' models only) The Vigor2820 Series features the latest 802.11n Draft 2.0 wireless LAN specification and has been certified by the WiFi alliance for cross compatibility and WiFi compliance (including WPA/WPA2 and WMM). 802.11n Draft 2.0 provides a total wireless bandwidth of up to 300Mb/s using new methods such as packet aggregation and channel bonding. Throughput depends on your own environment (factors such as obstructions, number of hosts and distance all make a significant difference), but actual transfer speeds of 100Mb/s are achievable (based on our real world tests). In addition, 802.11n Draft 2.0 provides greater coverage and resilience to interference compared to previous wireless standards thanks to the MIMO technology and the Vigor's triple-antennae diversity arrangement. This offset arrangement of aerials provides offset paths between hosts so that interference can be overcome. Wireless Security is comprehensive too; the Vigor 2820 Series provides several independent levels of security including encryption (up to WPA2), authentication (802.11x) and methods such as MAC address locking and DHCP fixing to restrict access to authorised users only. The Web interface lets you see how many and which clients are currently connected as well as their current bandwidth usage. An 'instant' block lets you disconnect a wireless user temporarily in case of query. The Wireless VLAN facility allows you to isolate wireless clients from each other or from the 'wired' LAN. The Multiple SSID features enables you to have up to four distinct or common virtual wireless access points. For example, you could have one for company usage, with access to your company LAN and another for public access which allows internet surfing only. Setting up wireless security is made easier thanks to the WPS feature (WiFi protected setup) whereby your client PC can get it's security keys by pressing a button on the front of the router. Voice-over-IP Features ('V' Models Only) The Vigor 2820 Series 'V' models build on the established DrayTek VoIP pedigree by adding even more new feaures. Twin analogue phone ports and an FXO analogue line port provide full PSTN and VoIP integration on the same phones, via both the Internet and your regular analogue line. The two phones can be used independently and simultaneously for both incoming and outgoing calls. As well as the two telephone ports, a third port, the FXO (Line) port, connects into your regular analogue line (PSTN/POTS*). This then gives the telephones access to your analogue line to allow you to make calls as well as your VoIP facility (you can select the PSTN line instead of VoIP by dialling #0). Incoming calls are automatically switched through to your telephone(s) (either one or both) so that each phone can be used for both VoIP and POTS calls. Both telephones plugged into your router have access to VoIP and your analogue line. In addition, using the 'Digit-Map' facility you can set rules about particular call destinations using either the POTS line or your SIP/VoIP service. For example, local calls can be routed via your PSTN line (if you have a free calls package for example) whereas international calls can go via your preferred VoIP provider; there is flexibility to have several digit-map rules. Detailed Specification: Physical Interfaces: LAN Ports (Switch) 1 X Gigabit Ethernet (1000Mb/s) Ports 3 X Megabit (100Mb/s) Ports Port-Based VLAN (Inclusive/Exclusive Groups) WAN Ports: ADSL Port Compliant with: ANSI T1.413 Issue2 ITU-T G.992.1 G.dmt (ADSL) ITU-T G.992.2 G.lite ITU-T G.992.3 ADSL2 ITU-T G.992.5 ADSL2+ Annex L (READSL) Annex M Secondary WAN Port : 10/100 Base-TX Ethernet for load balance and WAN failover USB Port for 3G Cellular Modem or Printer VoIP: 2-port FXS Phone Ports (Vigor2820 'V' models only) Load Balance/Failover Features: Outbound Policy-Based Load-Balance WAN Connection Fail-over BoD (Bandwidth on Demand) Vigor 2820 ATM Protocols (DSL): RFC-2684/RFC-1483 Multiple Protocol over AAL5 RFC-2516 PPP over Ethernet RFC-2364 PPP over AAL5 PPPoE pass through LAN/WLAN PPPoE/PPPoA Relay Transparent bridge for MPoA Multiple PVC support for Triple Play Applications (up to 8 simultaneous) Wireless LAN Features ('n' Models Only): 802.11n Compliant (Draft 2.0) Latest 'MIMO' Technology with three aerials (2T3R) Multiple SSID : Create up to 4 virtual wireless LANs (independent or joined) Packet Aggregation and Channel Bonding Compatible with 802.11b and 802.11g Standards Active Client list in Web Interface Wireless LAN Isolation (from VLAN groups and wired Ethernet interfaces) 64/128-bit WEP Encryption WPA/WPA2 Encryption Switchable Hidden SSID Restricted access list for clients (by MAC address) Time Scheduling (WLAN can be disabled at certain times of day) Access Point Discovery WDS (Wireless Distribution system) for WLAN Bridging and Repeating (Firmware Upgradable) 802.1x Radius Authentication Wireless Rate-Control Automatic Power Management 802.11e WMM (Wi-Fi Multimedia) VoIP Features (Vigor2820 'V' Models only): Protocols: SIP, RTP / RTCP 12 SIP Registrar Accounts (for up to 12 VoIP providers) Line port for PSTN Passthrough (integrates POTS line) Auto-fallback to PSTN under power/Internet failure G.168 Line Echo-cancellation Automatic Gain Control Jitter Buffer ( 125ms ) Voice Codecs: G.711 A / µ Law G.723.1 G.726 G.729 A / B VAD / CNG Tone Generation: DTMF , Dial , Busy , Ring Back , Call Progress DTMF Transmission: In Band / Out Band ( RFC-2833 ) / SIP info FAX / Modem Support G.711 Pass-through T.38 for FAX Supplemental Services: Caller ID Call Hold / Retrieve Call Waiting Call Waiting with Caller ID Call Transfer Call Forwarding ( Always , On Busy and On No Answer ) DND (Do not Disturb) Call Barring ( Incoming / Outgoing ) MWI ( Message Waiting Indicator ) ( RFC-3842 ) Hotline (Dial preset number when handset lifted) WAN Protocols (Ethernet): DHCP Client Static IP PPPoE PPTP L2TP * BPA Firewall & Security Features: CSM (Content Security Management): URL Keyword Filtering - Whitelist or Blacklist specific sites or keywords in URLs Surfcontrol Support - Block Web sites by category (subject to subscription) Prevent accessing of web sites by using their direct IP address (thus URLs only) Blocking automatic download of Java applets and ActiveX controls Blocking of web site cookies Block http downloads of file types : Binary Executable : .EXE / .COM / .BAT / .SCR / .PIF Compressed : .ZIP / .SIT / .ARC / .CAB/. ARJ / .RAR Multimedia : .MOV / .MP3 / .MPEG / .MPG / .WMV / .WAV / .RAM / .RA / .RM / .AVI / .AU Time Schedules for enabling/disabling the restrictions Block P2P (Peer-to-Peer) file sharing programs (e.g. Kazza, WinMX etc. ) Block Instant Messaging programs (e.g. IRC, MSN/Yahoo Messenger etc.) Multi-NAT, DMZ Host Port Redirection and Open Port Configuration Policy-Based Firewall MAC Address Filter SPI ( Stateful Packet Inspection ) with new FlowTrack Mechanism DoS / DDoS Protection IP Address Anti-spoofing E-Mail Alert and Logging via Syslog Bind IP to MAC Address Bandwidth Management: QoS Guaranteed Bandwidth for VoIP Class-based Bandwidth Guarantee by User-Defined Traffic Categories DiffServ Code Point Classifying 4-level Priority for each Direction (Inbound / Outbound) Bandwidth Borrowed Temporary (5 minute) Quick Blocking of any LAN Client Bandwidth / Session Limitation Network/Router Management: Web-Based User Interface (HTTP / HTTPS) CLI ( Command Line Interface ) / Telnet / SSH* Administration Access Control Configuration Backup / Restore Built-in Diagnostic Function Firmware Upgrade via TFTP / FTP Logging via Syslog SNMP Management with MIB-II TR-069 TR-104 VPN Facilities: Up to 32 Concurrent VPN Tunnels (incoming or outgoing) Tunnelling Protocols: PPTP, IPSec, L2TP, L2TP over IPSec IPSec Main and Agressive modes Encryption : MPPE and Hardware-Based AES / DES / 3DES Authentication : Hardware-Based MD5 and SHA-1 IKE Authentication : Pre-shared Key and X.509 Digital Signature LAN-to-LAN & Teleworker-to-LAN connectivity DHCP over IPSec NAT-Traversal ( NAT-T ) Dead Peer Detection (DPD) VPN Pass-Through Network Features: DHCP Client / Relay / Server Dynamic DNS NTP Client (Syncrhonise Router Time) Call Scheduling (Enable/Trigger Internet Access by Time) RADIUS Client DNS Cache / Proxy Microsoft™ UPnP Support Routing Protocols: Static Routing RIP V2 Operating Requirements: Rack Mountable (Optional Vigor 2820 mounting bracket required) Wall Mountable Temperature Operating : 0°C ~ 45°C Storage : -25°C ~ 70°C Humidity 10% ~ 90% (non-condensing) Power Consumption: 18 Watt Max. Dimensions: L240.96 * W165.07 * H43.96 ( mm ) Operating Power: DC 15V (via external PSU, supplied) Warranty : Two (2) Years RTB Power Requirements : 220-240VAC   951 Atcom IP04 4-port Asterisk IP PBX - 4 FXS 289 289.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-4-fxs-p-951.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 4 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 952 Atcom IP04 4-port Asterisk IP PBX - 3 FXO 264 264.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-3-fxo-p-952.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 3 x FXO modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 953 Atcom IP04 4-port Asterisk IP PBX - 2 FXO 239 239.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-2-fxo-p-953.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 2 x FXO modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 954 Atcom IP04 4-port Asterisk IP PBX - 1 FXO 214 214.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-1-fxo-p-954.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 1 x FXO module The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 955 Atcom IP04 4-port Asterisk IP PBX - 3 FXS 264 264.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-3-fxs-p-955.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 3 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 956 Atcom IP04 4-port Asterisk IP PBX - 2 FXS 239 239.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-2-fxs-p-956.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 2 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 957 Atcom IP04 4-port Asterisk IP PBX - 1 FXS 214 214.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-1-fxs-p-957.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 1 x FXS module The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 958 Atcom IP04 4-port Asterisk IP PBX - 3 FXO 1 FXS 289 289.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-3-fxo-1-fxs-p-958.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 3 x FXO and 1 FXS  modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 959 Atcom IP04 4-port Asterisk IP PBX - 2 FXO 2 FXS 289 289.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-2-fxo-2-fxs-p-959.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 2 x FXO and 2 FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 960 Atcom IP04 4-port Asterisk IP PBX - 1 FXO 3 FXS 289 289.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-1-fxo-3-fxs-p-960.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 1 x FXO and 3 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 961 Atcom IP04 4-port Asterisk IP PBX - 1 FXO 2 FXS 264 264.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-1-fxo-2-fxs-p-961.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 1 x FXO and 2 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 962 Atcom IP04 4-port Asterisk IP PBX - 2 FXO 1 FXS 264 264.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-2-fxo-1-fxs-p-962.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 2 x FXO and 1 FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 963 Atcom IP04 4-port Asterisk IP PBX - 1 FXO 1 FXS 239 239.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-1-fxo-1-fxs-p-963.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with 1 x FXO and 1 x FXS modules The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 964 Atcom IP04 4-port Asterisk IP PBX - IP only 189 189.00 Atcom Atcom IP PBX http://www.voipon.co.uk/atcom-ip04-4port-asterisk-ip-pbx-ip-only-p-964.html http://www.voipon.co.uk/images/atcom_ip04_pbx_sml.jpg new Availability: In stock The IP04 (IP-04) is a complete Asterisk powered IP-PBX with up to four FXS/FXO modules, It is an embedded system with built-in SIP/IAX proxy server and NAT functions. It provides a solid, uniform platform for VoIP communications as well as traditional PSTN communications. Targeted for small and medium businesses with an easy to use graphical user interface, IP04 provides an amazing value cost-saving solution on their telecommunication/data needs. With IP04, company with branch offices in different locations/countries can be easily combined together to work like a virtual single office through the internet. The IP04 comes with no modules, it is IP only. The IP04 is a stable, tried and tested open source IP PBX - not only is the software open-source, so is the hardware. More information can be found here . System : uClinux/Astfin Hardware: CPU: 400MHz Blackfin 532 Chip Four analog (FXO/FXS) module interfaces. NAND flash 256 M SDRAM 64M Features: Built-in configurable Asterisk IP PBX Astfin GUI High performance OSLEC (Open Source Line Echo Canceller) Voice mail Call forward, call waiting, call transfer Call conference Call queues SIP trunking, IAX trunking PSTN analog trunk (up to four PSTN trunks) Flexible dial plan Configurable IVR menu. Replaceable MMC/SD memory. Interface: 1 x RJ45 port 1 x Power port 1 x RS232 port 4 x RJ11 port 1 x MMC/SD slot 4 x FXS/FXO module slots 965 Polycom Power Supply for Soundpoint IP501 (2200-17568-015) 9 9.00 Polycom Polycom IP Telephones http://www.voipon.co.uk/polycom-power-supply-for-soundpoint-ip501-220017568015-p-965.html http://www.voipon.co.uk/images/polycom_psu-sml.jpg new Availability: In Stock Polycom Power Supply for Soundpoint IP501 (2200-17568-015) Universal Power Supply for Polycom Soundpoint IP501 VoIP phones. 966 PoE Cable for Polycom IP300 / IP500 20 20.00 Polycom VoIP Accessories http://www.voipon.co.uk/poe-cable-for-polycom-ip300-ip500-p-966.html http://www.voipon.co.uk/images/polycom_poe_cable-sml.jpg new Availability: In Stock PoE Cable for Polycom IP300 /IP500   Part number 2200-11014-601  967 OpenVox A1200P0012 - 12 FXO 270.0000 488.00 OpenVox OpenVox A1200P http://www.voipon.co.uk/openvox-a1200p0012-12-fxo-p-967.html http://www.voipon.co.uk/images/openvox_a1200p0012_sml.jpg new Availability: In stock The A1200P0012 is an OpenVox A1200P bundled with 12 FXO modules. You can also add FXO-100 or FXS-100 modules to use with this product according to your requirements. The image above is A1200P0012 with 12 FXO modules installed. The user can simply use A1200P with the provided Zaptel device driver (download on the left of the page) and Asterisk software, without the need to modify a single line of software code. Via professional firmware design and a new device driver, the A1200P greatly decreases the CPU payload, and also increases the stability of whole system. The 12 port card uses less CPU cycle than the 4 port card. The A1200P supports up to a total of 12 FXO and/or FXS connections with FXO-100 and/or FXS-100 modules. The FXO-100 (FXO) module allows the A1200P card to terminate analog telephone lines (POTS). The FXS-100 (FXS) module allows the A1200P card to terminate analog telephones. Key Features and benefits: Low CPU payload: below 25% CPU usage on server with 8 x A1200P fully loaded with 96 ports after driver installed, on a Celereon D 2.53Ghz. Scalable: Just add additional cards to extend the system. Modular design: Can upgrade at any time by adding/removing FXO or FXS modules up to a total of 12 per card. (A1200P0012 comes with 12 FXO modules - so is fully populated) Both 3.3v or 5v PCI slots can be used with this card. Module Pin to Pin compatible with X100M and S100M. You can use X100M/S100M modules on this card, or use FXO-100/ FXS-100 Modules from OpenVox. Excellent value: High quality at a low price Bundle: Includes 3 RJ11 to RJ45 splitters which plug into each of the 3 ports on the card to provide 12 ports for use on a fully expanded A1200P Application ready: Use Asterisk to build your IP-PBX system. RoHS Compliant. Certificates: CE, FCC Two year warranty 969 AC Adaptor for Aastra 142 DECT Station 16 16.00 Aastra Aastra SIP Dect/WLAN IP Phone http://www.voipon.co.uk/ac-adaptor-for-aastra-142-dect-station-p-969.html http://www.voipon.co.uk/images/no_image_available.jpg new Availability: In stock AC Adaptor for Aastra 142 DECT Station 970 Atcom AG-188 IAX2/SIP Analog Telephone Adaptor 40.0000 45.00 Atcom Atcom IAX Analog Adaptor http://www.voipon.co.uk/atcom-ag188-iax2sip-analog-telephone-adaptor-p-970.html http://www.voipon.co.uk/images/atcom_ag188_sml.jpg new Availability: In Stock The AG-188 is an analog telephone adapter from Atcom with one FXS port supporting both SIP and IAX protocols. The AG-188 adapts multi voice control protocols and voice compression codec to directly convert analogue voice into IP packets for internet transport, thus effectively using the existing bandwidth to provide a PSTN quality voice service. AG-188 series one port voice gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. The AG-188 is small and can be widely used in SOHO, small office and enterprise branches. Key features: Support two sip servers running at the same time. Redundancy sip server capable. NAT, Firewall. DHCP client and server. Support PPPoE, (used for ADSL, cable modem connecting). Support major G7.xxx CODEC. VAD,CNG. G.165 compliant 16ms echo cancellation E.164 dial plan and customized dial rules Hotline. Call Forward, Call Transfer, 3-way conference calls Call ID display DND(Do Not Disturb),Black List,Limit List Data Features: Static/Dynamic WAN-IP-Addressing, PPPoE Management: Web, telnet and keypad management. Adjustable user password and super password Upgrade firmware through HTTP, FTP or TFTP. Telnet remote management. Upload/download setting file Auto-provision. Safe mode provides reliability Phone book, maximum 100 entries. Interface Two RJ45 ports, one for WAN, one LAN. One RJ11 port, acts as lifeline function through lifeline accessory. Power port Data networking: MAC Address TCP:Transmission Control Protocol DHCP:Dynamic Host Configuration Protocol PPPoE, PPP Protocol over Ethernet SNTP - Simple Network Time Protocol STUN - Simple Traversal of User Datagram ... MD5 Message-Digest Algorithm DNS - Domain Name Server RTP: Real-time Transport Protocol RTCP:Real-time Control Protocol Telnet:Internet's remote login protocol HTTP:Hyper Text Transfer protocol FTP:File Transfer protocol TFTP:Trivial File Transfer Protocol Call control /VoIP Features SIP RFC3261,RFC 2543 Tone generation and Local DTMF re-generation according with ITU-T G.711(A-law or u-law) G729 AGC(Auto Gain Control) G.168/165 compliant 16ms echo cancellation AEC(Auto Echo Cancellation) VAD (Voice Activity Detection) CNG(Comfort Noise Generation Electric requirements Voltage: 9V ~ 24V Power adapter: output DC 12V/450 mA Operating requirement Operation temperature: 0 to 40 C ( 32 to 104 F) Storage temperature: -30 to 65 C (-22 to 149 F) Humidity: 10 to 90% no dew Regulatory compliance CE, FCC part 15 971 Snom High Foot Stand for Snom 300 18 18.00 Snom VoIP Accessories http://www.voipon.co.uk/snom-high-foot-stand-for-snom-300-p-971.html http://www.voipon.co.uk/images/snom_300_footstand_sml.jpg new Availability: In stock   The foot stand for the Snom 300 gives the phone a new look and lifts the phone up upright, providing its users a clear view of the phone. With the foot stand you do not need to lean over to use the phone as the phone lifts up to a more comfortable height to make it easier for you to make and receive calls. The foot stand fits into the phone at the rear end just like the standard stand but in different slots.   972 Snom High Foot Stand for Snom 320, 360, 370 29 29.00 Snom VoIP Accessories http://www.voipon.co.uk/snom-high-foot-stand-for-snom-320-360-370-p-972.html http://www.voipon.co.uk/images/snom_360_footstand_sml.jpg new Availability: In stock The foot stand for the Snom 320, 360 and 370 gives the phone a new look and lifts the phone upright, providing its users a clear view of the phone. With the foot stand you do not need to lean over to use the phone as the phone lifts up to a more comfortable height to make it easier for you to make and receive calls. The foot stand fits into the phone at the rear end just like the standard stand but in different slots.   973 Sangoma A500H FlexBRI PCI Card 146.0000 190.00 Sangoma Sangoma Hybrid Cards http://www.voipon.co.uk/sangoma-a500h-flexbri-pci-card-p-973.html http://www.voipon.co.uk/images/sangoma_flex_bri.jpg new Availability: Due Autumn/Fall 2008 Connect your Analog Lines into your BRI system using only one interface card! Ideal for small or home offices, Sangoma's industry-leading hybrid FlexBRI solution will seamlessly integrate your analog fax machine with your BRI phone system, using only one PCI or PCI express slot and will fit in even the smallest 1U servers. A single PCI or PCI Express slot hosts the connection for up to 4 ports of BRI and 2 ports of Analog FXO or FXS and ensures common synchronous clocking for all channels with absolutely no signaling issues. The card is 100% software configurable. Like all cards in the award-winning AFT Product line, Octasic™ Telco-grade, hardware echo cancellation is available. Technical specifications 4 ports of BRI are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers—simply invert the colour-coded module. 2 ports of Analog FXS or FXO are supported. Supports Asterisk®, Yate™, FreeSwitch™, CallWeaver™, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single PCI or PCI Express interface. Autosense compatibility with 5V and 3.3V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables, and two narrow RJ11/4 – RJ11/6 analog cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3V or 5V. Temperature range: 0° – 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Operating systems Linux (all versions, releases and distributions from 1.0 up). Solaris. Diagnostic tools WANPIPEMON, SNMP, System logs. Certification FCC Part 15 Class A, FCC Part 68, CE, TBR 1 Production quality ISO 9002 Warranty Lifetime parts and labour with product registration. PLUS 30-day “no questions asked” return policy. 974 Sangoma A500H FlexBRI PCI Express Card 171.0000 225.00 Sangoma Sangoma Hybrid Cards http://www.voipon.co.uk/sangoma-a500h-flexbri-pci-express-card-p-974.html http://www.voipon.co.uk/images/sangoma_flex_bri.jpg new Availability: Due Autumn/Fall 2008 Connect your Analog Lines into your BRI system using only one interface card! Ideal for small or home offices, Sangoma's industry-leading hybrid FlexBRI solution will seamlessly integrate your analog fax machine with your BRI phone system, using only one PCI or PCI express slot and will fit in even the smallest 1U servers. A single PCI or PCI Express slot hosts the connection for up to 4 ports of BRI and 2 ports of Analog FXO or FXS and ensures common synchronous clocking for all channels with absolutely no signaling issues. The card is 100% software configurable. Like all cards in the award-winning AFT Product line, Octasic™ Telco-grade, hardware echo cancellation is available. Technical specifications 4 ports of BRI are supported. Mix TE and NT modes, as required. Changing modes requires no jumpers—simply invert the colour-coded module. 2 ports of Analog FXS or FXO are supported. Supports Asterisk®, Yate™, FreeSwitch™, CallWeaver™, PBX/IVR projects, as well as other Open Source and proprietary PBX, Switch, IVR or VoIP gateway applications. Single PCI or PCI Express interface. Autosense compatibility with 5V and 3.3V PCI busses. Fully PCI 2.2 and PCI Express compliant, compatible with all commercially available motherboards, proper sharing of PCI interrupts. Dimensions: 2U Form factor: 187mm x 55mm for use in restricted chassis. Short 2U compatible mounting clips included for installation in 2U rackmount servers and high quality, tested 2m 8-pin RJ45 port splitter cables, and two narrow RJ11/4 – RJ11/6 analog cables included. 32 bit bus master DMA data exchanges across PCI interface at 132 Mbytes/sec for minimum host processor intervention. Intelligent hardware: Downloadable FPGA programming with multiple operating modes. Add new features related to voice and/or data when they become available. Power: 800mA peak, operational 300mA max at +3.3V or 5V. Temperature range: 0° – 50°C. Optimized DMA stream and hardware-level HDLC handling unload the host CPU. Raw bitstream interfaces can be used to support arbitrary non-standard line protocols, such as non-byte aligned monosynch or bisynch. WANPIPE® supports certified, field tested and reliable Frame Relay, PPP, HDLC and X.25. Operating systems Linux (all versions, releases and distributions from 1.0 up). Solaris. Diagnostic tools WANPIPEMON, SNMP, System logs. Certification FCC Part 15 Class A, FCC Part 68, CE, TBR 1 Production quality ISO 9002 Warranty Lifetime parts and labour with product registration. PLUS 30-day “no questions asked” return policy.   975 Openvox B100M 1 Port ISDN BRI Mini PCI Card 99.0000 110.00 OpenVox OpenVox Digital (BRI) Cards http://www.voipon.co.uk/openvox-b100m-1-port-isdn-bri-mini-pci-card-p-975.html http://www.voipon.co.uk/images/openvox_b100m_bri_card.gif new Availability: In Stock OpenVox B100M is a mini PCI type III BRI card supporting one BRI S/T interface. NT/TE mode can be independently configured on the port. B100M can be used for building open source Asterisk® based systems such as ISDN PBX and VoIP gateway. Target Applications: High performance ISDN PC cards ISDN PABX for BRI VoIP gateways ISDN LAN routers for BRI ISDN least cost routers for BRI ISDN test equipment for BRI Features: One integrated S/T interface ITU-T I.430 and TBR 3 certified and S/T ISDN supporting in TE and NT mode DTMF detection on all B-channels RoHS compliant Multiparty audio conferences bridge Support mini PCI type III Designed for low-power systems Support AskoziaPBX system, trixbox, Elastix and other asterisk based distributions Support VIA, PC Engines motherboard and AMD geode based motherboard The port can be configured for TE or NT mode Support Bristuff, ISDN4BSD and mISDN driver Application ready: Use Asterisk to build your IP-PBX/Voicemail system Two Year Warranty! 978 Doro Pro Sound HS1110 Headset 0 0.00 Doro IP Telephones http://www.voipon.co.uk/doro-pro-sound-hs1110-headset-p-978.html http://www.voipon.co.uk/images/doro_pro_sound_hs1110m.jpg new Availability: In Stock Professional lightweight headset Over the head wear style Kevlar* reinforced cables Cable quick disconnector. In use indicator on boom. Monaural headset Fits both ears Adjustable headband Kevlar* reinforced cables Dual stainless steel microphone boom DNRS (Doro noise reduction system) In use indicator on boom** Cable quick disconnector Cables for both cordless/corded phones included Gold plated connectors Weight (g) 979 Doro IP830c IP Telephone 134.0000 147.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip830c-ip-telephone-p-979.html http://www.voipon.co.uk/images/doro_ip830c.jpg new Availability: In Stock Doro IP830c VoIP Telephone with Large Display VoIP telephone with large display Excellent business phone with tiltable, backlit graphic display for better menu navigation as well as access to advanced call control features through your web browser. Network features n x 10/100Mbps WAN port 1 n x 10/100Mbps LAN port 1 PoE / Class / 1 VLAN DHCP client NTP/SNTP Configuration features Configuration via web Configuration via menu Https and Certificates Remote upgrade and config Prog. Configuration server Upgrade via TFTP Upgrade via HTTP Voice features SIP 2.0 RTP/RTCP Codec G.711 A-Law Codec G.711 µ-Law Codec GSM Codec G.723.1 Codec G.72 9A/B/AB •/-/- G.722 (wideband) Voice Activity Detection Comfort Noice Generation Silence suppression Jitter buffer Echo cancellation DTMF (inband/outband/SIP-INFO) • / • / • Telephone features LED indication Display 128 x 64 pixels Choice of off-hook tones Number of Lines/SIP Registrations 12 Phonebook 100 Redial 100 Function Keys 12 Desktop/Wall mountable D/W Languages selection Handsfree Headset outlet RJ22 Call features Call forwarding Call hold Call transfer (monitored) Call transfer (blind) 3-way Conference CLIP / CLIR • / • Ringer Nr of ring volume controls Nr of ring tone frequency controls Caller ID Caller ID memory 100 Time and date of each memory Message waiting Display name from the network Display name from phonebook 980 Doro IP840c IP Telephone 168.0000 179.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip840c-ip-telephone-p-980.html http://www.voipon.co.uk/images/doro_ip840c.jpg new Availability: In stock Doro IP840c Advanced VoIP Telephone with High Definition Display Advanced VoIP telephone with high definition display Doro ip840c features a 250 position phonebook and a high definition display for enhanced presentation of caller information, news ticker information, public phone directories and much more. Network features n x 10/100Mbps WAN port 1 n x 10/100Mbps LAN port 1 PoE / Class / 1 VLAN DHCP client NTP/SNTP Configuration features Configuration via web Configuration via menu Https and Certificates Remote upgrade and config Prog. Configuration server Upgrade via TFTP Upgrade via HTTP Voice features SIP 2.0 RTP/RTCP Codec G.711 A-Law Codec G.711 µ-Law Codec GSM Codec G.723.1 Codec G.729A/B/AB G.722 (wideband) Voice Activity Detection Comfort Noice Generation DTMF (inband/outband/SIP-INFO) Telephone features LED indication Display 240 x 128 pixels Choice of off-hook tones Number of Lines/SIP Registrations 12 Phonebook 250 Redial 100 Function Keys 12 Desktop/Wall mountable D/W Languages selection Handsfree Headset outlet RJ22 Call features Call forwarding Call hold Call transfer (monitored) Call transfer (blind) 3-way Conference CLIP / CLIR Ringer Nr of ring volume controls Nr of ring tone frequency controls Caller ID Caller ID memory 100 Time and date of each memory Message waiting Display name from the network Display name from phonebook Silence suppression Jitter buffer Echo cancellation 981 Doro IP820c IP Telephone 106.0000 115.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip820c-ip-telephone-p-981.html http://www.voipon.co.uk/images/doro_ip820c.jpg new Availability: In Stock Doro IP820c VoIP Telephone (12 SIP Registrations) Doro ip820c provides 12 SIP registrations, full-duplex speakerphone, headset outlet, 12 programmable function keys and a 100 position phonebook. Network features n x 10/100Mbps WAN port 1 n x 10/100Mbps LAN port 1 PoE / Class / 1 VLAN DHCP client NTP/SNTP Configuration features Configuration via web Configuration via menu Https and Certificates Remote upgrade and config Prog. Configuration server Upgrade via TFTP Upgrade via HTTP Voice features SIP 2.0 RTP/RTCP Codec G.711 A-Law Codec G.71 1 µ-Law Codec GSM Codec G.723.1 Codec G.729A/B/AB •/-/- G.722 (wideband) Voice Activity Detection Comfort Noice Generation Silence suppression Jitter buffer Echo cancellation DTMF (inband/outband/SIP-INFO) • / • / • Telephone features LED indication Display 2x24 Choice of off-hook tones Number of Lines/SIP Registrations 12 Phonebook 100 Redial 100 Function Keys 12 Desktop/Wall mountable D/W Languages selection Handsfree Headset outlet RJ22 Call features Call forwarding Call hold Call transfer (monitored) Call transfer (blind) 3-way Conference CLIP / CLIR • / • Ringer Nr of ring volume controls Nr of ring tone frequency controls Caller ID Caller ID memory 100 Time and date of each memory Message waiting Display name from the network Display name from phonebook 982 Doro IP810c IP Telephone 67.0000 72.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip810c-ip-telephone-p-982.html http://www.voipon.co.uk/images/doro_ip810c.jpg new Availability: In stock Doro IP810c VoIP Business Phone Doro ip810c offers lots of functionality for being an entry level IP telephone. Four SIP registrations, wideband audio codec and 6 programmable function keys are just a few. Network features n x 10/100Mbps WAN port 1 n x 10/100Mbps LAN port 1 PoE / Class • / 1 VLAN DHCP client NTP/SNTP Configuration features Configuration via web Configuration via menu Https and Certificates Remote upgrade and config Prog. Configuration server Upgrade via TFTP Upgrade via HTTP Voice features SIP 2.0 RTP/RTCP Codec G.711 A-Law Codec G.711 µ-Law Codec GSM Codec G.723.1 Codec G.729A/B/AB •/-/- G.722 (wideband) Voice Activity Detection Comfort Noice Generation Silence suppression Jitter buffer Echo cancellation DTMF (inband/outband/SIP-INFO) • / • / • Telephone features LED indication Display 2x16 Choice of off-hook tones Number of Lines/SIP Registrations 4 Phonebook 100 Redial 100 Function Keys 6 Desktop/Wall mountable D/W Languages selection Handsfree Headset outlet RJ22 Call features Call forwarding Call hold Call transfer (monitored) Call transfer (blind) 3-way Conference CLIP / CLIR • / • Ringer Nr of ring volume controls Nr of ring tone frequency controls Caller ID Caller ID memory 100 Time and date of each memory Message waiting Display name from the network Display name from phonebook 983 Doro IP700 Wifi IP Telephone 0 0.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip700-wifi-ip-telephone-p-983.html http://www.voipon.co.uk/images/doro_ip_700.jpg new Availability: In Stock The slim and lightweight ip700wifi is a stylishly designed wireless phone for low-cost VoIP telephony. It allows businesses and households to achieve significant savings by making and receiving calls at a fraction of the cost of GSM and other local and long distance phone services. Ideal wherever access to wireless internet is available, ip700wifi is perfect for the home and office. It can also be used at public venues across the globe such as hotels and resorts, airports, train stations, restaurants, internet cafés, school campuses, commercial centres and other Wi-Fi hotspots. Smart business solution ip700wifi is an excellent solution for small and medium-sized businesses. Internal calls are handled with standard PBX functions including Call Forward, Call Waiting and Speed Dial. In addition, companies with offices in more than one location can easily set up communications to appear as if originating from the same geographical site. For example, employees working abroad can receive and place calls as if they were actually situated at the head office, completely transparent to the customer. Configuration is handled effortlessly through a WEB interface. SIP for today and tomorrow Based on SIP 2.0, ip700wifi represents the very latest in internet communication technology. With the SIP standard recognised as the leading protocol for the delivery of data, multimedia and voice over IP, ip700wifi is a safe investment guaranteed to provide a long-term, low-cost and universal solution. And with support for multiple SIP accounts, users can carry both their business extension and private numbers in the same phone. This allows them to place calls from, or answer calls made to either number as if still still at the office or at home – even when half way around the world. Secure and convenient ip700wifi with Ad-Hoc mode is fully compliant with IEEE 802.11b for fast, reliable and encrypted communication in WLAN / Wi-Fi networks. Auto logon WLAN on pre-set priority, One-touch Site Survey and multiple WLAN configuration storage have all been incorporated to enhance the phone’s functionality. In addition, access point call handover is seamless within the same SSID. The result is a phone that provides not only low-cost VoIP telephony, but also the freedom and convenience of wireless communication. All the features one expects In addition to its handsomely slim and lightweight design, ip700wifi offers a full range of features including backlit, colour display, vibrating alert, adjustable ringer, over 15 selectable languages and a 100 position phonebook. Top sound quality is provided through Comfort Noise Generation, Voice Activity Detection and Echo Cancellation. Furthermore, ip700wifi is exceptionally simple to set up. Even the software is easily updated via Wi-Fi. Features User Interface: Slim lightweight design backlit colour display 100 positions phonebook speed dialing adjustable ringer vibrating alert clock call timer headset outlet language selection. Caller ID call log call forward call waiting redial seamless AP call hand-over within same SSID. Audio quality: G.711a t G.711u t G.729a comfort noise generation voice activity detection adjustable jitter buffer. WLAN / Wi-Fi / VoIP: IEEE 802.11b/g one touch site survey auto logon WLAN on pre-set priority security encryption multiple WLAN and SIP (RFC3261 compliant) user configuration storage STUN RTP TCP/IP static IP DHCP ICMP DNS HTTP. Installation: Software update via Wi-Fi configuration via WEB interface static IP address / DHCP NTP 984 Doro IP84EKP Keypad / Expansion Module 74.85 74.85 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip84ekp-keypad-expansion-module-p-984.html http://www.voipon.co.uk/images/doro_ip85ekp.jpg new Availability: In Stock Extra keypad for ip820c/ip830c/840c Get more out of your doro ip820c, ip830c or ip840c telephone with our accessory ip85ekp expansion module. This extra keypad gives you 42 additional keys with highly flexible programming options, all easily configured via the built-in WEB interface. Expand your possibilities 42 programmable keys LED status indicators Easily programmed 986 Doro IP880dect IP Telephone 0 0.00 Doro Doro IP Telephones http://www.voipon.co.uk/doro-ip880dect-ip-telephone-p-986.html http://www.voipon.co.uk/images/doro_ip880_dect.jpg new Availability: In Stock Doro IP880dect An expandable complement to wired VoIP systems, ip880dect provides greater user freedom with DECT wireless technology. Each ip880dect gateway can handle up to 8 SIP accounts and handsets, 2 SIP providers (e.g. external long distance service plus own SIP enabled PBX for internal/incoming calls) as well as regular PSTN service.  Perfect for small businesses or individual departments within larger companies. Main features: LAN interface connection: 10BASE-T (RJ45 8/8) PSTN connection: RJ11 Double antenna for diversity 3 concurrent VoIP calls (SIP 2.0 – RFC3261) Call forward options (all, busy and no answer) Voice mail support (provider dependant) Call timer Call transfer, call hold DECT Call transfer (Attend and Blind), call hold SIP Music on hold support (only via PBX) Call divert Call park, call pick-up (only via PBX) Call waiting / switching between calls Redial Caller ID with name from phonebook VID ring tone (IP and PSTN) Call log: incoming, outgoing and missed calls Call return (from call log) Call swap (one handset toggles between two calls) Internal call Gateway phonebook Speed dial Automatic on hook when returning to charger DTMF sending in call Emergency call routing to PSTN CID; FSK and DTMF. Message waiting. CODECSs: G.711, G729A/B SIP features: RFC3261 compliant Digest/basic authentication DNS SRV (RFC3263), redundant server support Offer/Answer (RFC3264) Message waiting indication reception (RFC3842) Subscription for MWI (RFC3265) In-Band DTMF/Out of band DTMF (RFC2833) STUN Client (NAT Traversal) rport (RFC3581) REFER (RFC3515) 8 SIP registrations with dual SIP server/registrar SIP configuration via http-page or from handset Handset Features Lithium battery 650mAH Talk time 15h Standby time 150h Headset jack: 2,5mm 64K CSTN backlit Colour LCD display 5 levels battery indicator Key lock Speakerphone Microphone mute Ringer off selection Message indication Phonebook Alarm Out of range warning 10  Selectable ringer melodies Adjustable ringer volume Adjustable handset, speaker and headset volume Auto answer Automatic encryption (DECT) Call reject option Date/Time from the network Key beep on/off Multi Language support: DE, EN, ES, FR, IT, NL, PL, PT and SE Redial Silent selection at incoming call