Dinstar MTG2000 Carrier-grade Digital VoIP Gateway
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.
- 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 accordingtodifferentworking 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-switchof Huawei,Cisco and ZTE etc
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
1x RS232, 115200bps
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
Software Features Local/Transparent RingBack
ToneOverlapping Dialing
Dialing Rules，with up to2000
PSTN group by E1 port or E1Timeslot
IP Trunk Group Configuration
Voice Codecs Group
Caller and Called Number WhiteLists
Caller and Called Number BlackLists
Access Rule Lists
IP Trunk Prior
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
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
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 SIPAccounts
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 eachDirection
Caller and Called Number Manipulation
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