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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
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