MTG3000

High Density Digital VoIP Gateway


MTG3000 is a carrier-grade Digital VoIP gateway, scalable from 16 to 63 ports E1/T1 with STM-1 interface. 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 high performance, reliable communication network for users.

MTG3000 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, MTG3000 is ideal for a variety of applications of services providers and telecom operators.

  • E1/T1

  • T.38/T.30

  • PRI

  • SS7

  • NGN/IMS

  • SNMP

High Capacity Digital VoIP Gateway for Carriers & ITSPs

16 to 63 ports E1/T1 in 2U chassis, STM-1 interface

Up to 1890 simultaneous calls

Redundancy Dual MCU units

Dual Power Supplies

Flexible routing 

Multiple SIP trunks

Fully compatible with mainstream VoIP platforms

Rich Experiences on PSTN Protocols

ISDN PRI

ISDN SS7, SS7 links redundancy

R2 MFC

T.38,Pass-through fax, 

Support modem and POS machines

More than 10-year expriences to integrate with a wide range of Legacy PBXs / Service providers' PSTN networks

Easy Management 

  • Intuitive Web interface

    Support SNMP

    Automated provisioning

  • Dinstar Cloud Managemnt System

    Configuration Backup & Restore

    Advanced Debug tools

  • Features
  • Download
  • 1+1 Redundant Main Control Unit (MCU)
    Up to 63 E1s/T1s, STM-1 interface
    4 Digital Processing Unit (DTU), each support 512 channels
    Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC 13k/15k,AMR
    Dual Power Supplies
    Silence Suppression
    2 GE
    Comfort Noise
    SIP v2.0
    Voice Activity Detection
    SIP-T,RFC3372, RFC3204, RFC3398
    Echo Cancellation (G.168),with up to 128ms
    SIP Trunk Work Mode: Peer/Access
    Adaptive Dynamic Buffer
    SIP/IMS Registration :with up to 256 SIP Accounts
    Voice, Fax Gain Control
    NAT: Dynamic NAT, Rport
    FAX:T.38 and Pass-through
    Flexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN
    Support Modem/POS
    Intelligent Routing Rules
    DTMF Mode: RFC2833/SIP Info/In-band
    Call Routing base on Time
    Clear Channel/Clear Mode
    Call Routing base on Caller/Called Prefixes
    ISDN PRI:
    256 Route Rules for each Direction
    Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
    Caller and Called Number Manipulation
    R2 MFC
    Local/Transparent Ring Back Tone
    Web GUI Configuration
    Overlapping Dialing
    Data Backup/Restore
    Dialing Rules, with up to 2000
    PSTN Call Statistics
    PSTN group by E1 port or E1 Timeslot
    SIP Trunk Call Statistics
    IP Trunk Group Configuration
    Firmware Upgrade via TFTP/Web
    Voice Codecs Group
    SNMP v1/v2/v3
    Caller and Called Number White Lists
    Network Capture
    Caller and Called Number Black Lists
    Syslog: Debug, Info, Error, Warning , Notice
    Access Rule Lists
    Call History Records via Syslog
    IP Trunk Priority
    NTP Synchronization
    Radius
    Centralized Management System
  • MTG3000 Trunk Gateway Datasheet
    MTG 3000 VoIP Trunk Gateway User Manual
    MTG Series Quick Installation Guide