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Dinstar MTG2000 20*E1/T1 PRI Gateway

The MTG2000 is a brand-new model with a simple VoIP gateway. It is aimed towards telecommunications companies, enterprises, and other similar organizations.

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PRI Gateway MTG2000 Dinstar 20*E1/T1 Keeping the concept of viable, economical, and operable in mind necessitates a lot of coordination and storage. It manages VoIP and FoIP at the carrier level, as well as high-end activities like modem and voice acknowledgment. As a result, it provides clients with a flexible, efficient, and future-oriented correspondence system. MTG2000 understands the interconnection between SIP and traditional indicators like SS7 and PRI and supports a number of flagging conventions. It supports a variety of codec schemes, includes flag encryption and keen voice recognition technology, and improves the efficiency of hauling assets while maintaining speech quality. The storage compartment passage is ideal for a variety of SMEs, call centers’, telecom administrators’, and large-scale projects’ access systems.


  • A 1U dimension and a high-integration structure.
  • Supports up to 20 E1/T1 ports.
  • POS, Modem, FoIP, and VoIP are just a few of the options available.
  • The digital card also supports dialing rules in an operational and flexible manner, allowing users to define dialing rules based on various environmental processes.
  • A number of coding standards are also supported, including iLBC, G.792A/B, G.723.1, and G.711A/U.
  • Interoperability and high compatibility with Alcatel, NEC, and Avaya PBXs.
  • Renowned soft switch suppliers such as ZTE, Cisco, and Huawei would benefit consumers even more.

Mumbai Dinstar MTG2000 20 E1 T1 PRI Gateway | AcmaTel Communications

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Product Specification


Voice Capabilities:
Adaptive Dynamic Buffer Yes
Clear Channel/Clear Mode Yes
Codecs: G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
Comfort Noise Yes
DTMF Mode: RFC2833/Signal/Inband
Echo Cancellation (G.168),with up to 128ms Yes
FAX:T.38 and Pass-through Yes
Silence Suppression Yes
Support Modem/POS Yes
Voice ,Fax Gain Control Yes
Voice Activity Detection Yes
Call Features:
256 Route Rules for each Direction Yes
Call Routing base on Caller/Called Prefixes Yes
Call Routing base on Time Yes
Caller and Called Number Manipulation Yes
Flexible Route Methods PSTN-PSTN, PSTN-IP, IP-PSTN
Intelligent Routing Rule Yes
Compliance CE, FCC
Dimensions(W/D/H) 436*300*44.5mm(1U)
Humidity 10%-90% Non-Condensing
Operating Temperature 0 ℃ ~ 45 ℃
Ping and Tracert Test Yes
Polarity Reversal (GSM) Yes
Power Consumption 45W
Power Supply 100-240VAC, 50-60 Hz
Storage Temperature -20 ℃~80 ℃
Unit Weight 3.8kg
Ethernet Interface
GE1: 10/100/1000 BaseT Adaptive Ethernet
GE0: 10/100/1000 BaseT Adaptive Ethernet
Serial Port :1* RS232, 115200bps
Call History Records via Syslog Yes
Centralized Management System Yes
Data Backup/Restore Yes
Network Capture Yes
NTP Synchronization Yes
PSTN Call Statistics Yes
SIP Trunk Call Statistics Yes
SNMP v2 Yes
Syslog Debug, Info, Error, Warning , Notice
Web GUI Configuration Yes
Physical Interfaces :
DTU Module RJ48(Impedance 120Ω
E1/T1 Ports 4/8/12/16/20 E1/T1
Interface Type 4 E1/T1
Clock Local/Remote Clock Source
E1 Frame Type DF,CRC-4,CRC_ITU
ISDN PRI 23B+D(T1),30B+D(E1),NT or TE ITU-T Q.921, ITU-T Q.931, Q.Sig
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)
Dimension(w/D/H) 225*150*38mm
Operation Humidity 10%-90% no condensation
Temperature (operation): 0 ℃~ 45 ℃ •(storage): -20 ℃~80 ℃
Weight 0.75kg
•Power input:100-240V,50-60Hz output:12VDC,1.0A
SIP Protocol
SIP V2.0 RFC3261 Yes
Configurable SIP Release Code Yes
DNS SRV/A Query YesYes
Dynamic NAT, Static NAT, STUN Yes
NAT Traversal Yes
Outbound Proxy Yes
SDP RFC2327 Yes
Session Timer RFC4028 Yes
SIP Extensions Yes
SIP Master/Slave Server Yes
SIP Registration Yes
SIP Trunk Yes
Software Features:
Access Rule Lists Yes
Caller and Called Number White Lists Yes
Dialing Rules,with up to 2000 Yes
IP Trunk Priority Yes
Local/Transparent Ring Back Tone Yes
Overlapping Dialing Yes
PSTN group by E1 port or E1 Timeslot Yes
RTP and Signaling Encryption (VOS RC4) Yes
SIP Trunk Work Mode :Peer/Access Yes
Voice Codecs Group Yes
•IP Trunk Group Configuration Yes
VoIP Protocol
RFC3262, 3263,3264,3265,3515,2976,3311 Yes
NAT: Dynamic NAT, Rport
RTP/RTCP, RFC2198, 1889 Yes
SIP Trunk Work Mode :Peer/Access Yes
SIP v2.0 (UDP/TCP),RFC3261 SDP,RTP(RFC2833) Yes
SIP-T,RFC3372, RFC3204, RFC3398 Yes
SIP/IMS Registration :with up to 256 SIP Account Yes
SIP/IMS Registration :with up to 256 SIP Accounts Yes

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