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Dinstar MTG2000 16 E1/T1 Trunk Gateway (MGT2000-16E1)

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24/07/2024 £3,336.50 exc. VAT
£4,003.80 inc. VAT
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Key Features
  • Carrier grade hardware design, 1+1 power supply
  • High-integrated structure, up to 20 E1 ports in 1U size
  • Support flexible dialing rules and operation

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SKU:MGT2000-16E1
Brand: Dinstar
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Dinstar MTG2000 16 E1/T1 Trunk Gateway (MGT2000-16E1)

The Dinstar MTG2000 is a carrier-grade intelligent Digital VoIP gateway, scalable from 4 to 20 ports E1/T1. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition. With highly maintainable, manageable and operable features, it offers a flexible, high-efficient, future-oriented communication network for users.

MTG2000 supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG2000 is ideal for various applications of large enterprises, call centers, services providers and telecom operators.

Dinstar MTG2000 Key Features

  • Carrier  grade  hardware  design,  1+1  power  supply
  • High-integrated  structure,  up  to  20  E1  ports  in  1U  size
  • Support  flexible  dialing  rules  and  operations,  allowing  users  to  customize  dialing  rules  according  to  different working  environments
  • Support  multiple  coding  standards:  G.711A/U,  G.723.1,  G.729A/B  and  iLBC
  • High  compatibility,  interoperable  with  PBX  of  Avaya,  NEC  and  Alcatel,  and  also  leading  soft-switch  of Huawei,Cisco  and  ZTE  etc
Dinstar MTG2000 - Technical Specifications

Physical Interfaces

  • E1/T1 Ports 4/8/12/16/20 E1/T1
  • DTU Module 4 E1/T1
  • Interface Type RJ48(Impedance 120Ω)
  • Ethernet Interface GE1: 10/100/1000 BaseT Adaptive Ethernet GE0: 10/100/1000 BaseT Adaptive Ethernet
  • Serial Port 1* RS232, 115200bps

Voice Capabilities

  • Codecs: G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
  • Silence Suppression Comfort Noise
  • Voice Activity Detection
  • Echo Cancellation (G.168),with up to 128ms Adaptive Dynamic Buffer
  • Voice, Fax Gain Control FAX:T.38 and Pass-through Support Modem/POS
  • DTMF Mode: RFC2833/Signal/In-band Clear Channel/Clear Mode

Environmental

  • 1+1 Redundancy Power Supply Power Supply: 100-240VAC, 50-60 Hz Power Consumption:45W
  • Operating Temperature:0 ℃  ~ 45 ℃ Storage Temperature: -20 ℃ ~80 ℃ Humidity:10%-90% Non-Condensing Dimensions(W/D/H): 436*300*44.5mm(1U) Unit Weight: 3.8kg
  • Compliance: CE, FCC

PSTN

  • ISDN PRI 23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
  • Signal 7/SS7 ITU-T, ANSI,ITU-CHINA MTP1/MTP2/MTP3, TUP/ISUP
  • E1 Frame Type: DF,CRC-4,CRC_ITU
  • T1 Frame Type: 4-Frame Multi-frame (F4,FT), 2-Frame  Multi-frame (F12, D3/4), Extended Super-frame (F24, ESF) , Remote Switch Mode (F72, SLC96) Line Codes: E1:NRZ,CMI,AMI,HDB3 T1:NRZ,CMI,AMI,B8ZS
  • Clock: Local/Remote Clock Source

Maintenance

  • Web GUI Configuration Data Backup/Restore PSTN Call Statistics SIP Trunk Call Statistics
  • Firmware Upgrade via TFTP/FTP/Web Network Capture
  • SNMP v2
  • Syslog: Debug, Info, Error, Warning , Notice Call History Records via Syslog, NTP Synchronization, Centralized Management System                   

Software Features

  • Local/Transparent Ring Back Tone Overlapping
  • Dialing Rules up to 2000
  • PSTN group by E1 port or E1 Timeslot IP Trunk Group Configuration
  • Voice Codecs Group
  • Caller and Called Number White Lists Caller and Called Number Black Lists Access Rule Lists
  • IP Trunk Priority

VoIP Protocol

  • SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833), RFC3262, 3263,3264,3265,3515,2976,3311
  • SIP TLS/SRTP
  • RTP/RTCP, RFC2198, 1889
  • SIP-T,RFC3372, RFC3204, RFC3398
  • SIP Trunk Work Mode : Peer/Access
  • SIP/IMS Registration :
  • With up to 2000 SIP Accounts NAT: Dynamic NAT, Rport

Call Features

  • Flexible Route Methods
  • PSTN-PSTN, PSTN-IP, IP-PSTN
  • Intelligent Routing Rules Call Routing base on Time
  • Call Routing base on Caller/Called Prefixes 256 Route Rules for each Direction
  • Caller and Called Number Manipulation

 

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