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Dinstar MTG200 Cost-effective VoIP Trunk Gateway

The MTG200 series Digital VoIP Gateways with 1/2 ports E1/T1 allow you to easily migrate your legacy PSTN networks (legacy PBX or E1/T1 service providers) to a VoIP network. With a small investment, you can enjoy the true benefits of VoIP while maintaining your PSTN connectivity. It is a small box designed for SMEs and the open-source market, and it is fully compatible with Asterisk, Elastix, Trixbox, Freeswitch, and other popular VoIP platforms. Integrating with your legacy PBX or PSTN network is also simple with ISDN PRI / SS7 / R2 MFC support.

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Dinstar MTG200 Trunk Gateway enables direct call routing between fixed-line ISDN and cost-effective IP networks in order to capitalize on low-cost VoIP telephony. Businesses of all sizes can benefit from low-cost Voice over IP calls and achieve significant cost savings by integrating MTG200 Trunk Gateway with existing PBX / PABX telephone systems. This eliminates the high upfront costs associated with communications infrastructure changes associated with VoIP migration.MTG200 provides PBX/IP-PBX users with a powerful telephony cost-saving device and advanced flexible routing capabilities between ISDN PRI and SIP, as well as the ability to be placed between the PBX and the PSTN, saving the cost of an additional PBX E1/T1 port. It is ideal for a variety of access networks used by SMEs, call centers, telecom operators, and large enterprises.

Key Features: Up to 60 simultaneous calls with 2* E1/T1 ports, Support for echo cancellation, DJB, CNG, VAD, and QoS  Use of the PBX’s existing E1/T1 ISDN PRI interface, Use of the IP-existing PBX’s VoIP interface, Maintain existing dialing  habits and business communication patterns, Built-in Web-based management, SNMP, command line interface (CLI)


  • 1/2 E1/T1 connectors, RJ48C interface
  • Modem/POS 2 GE Support
  • DTMF Protocol: RFC2833/SIP Info/In-band
  • VLAN 802.1p/q SIP v2.0
  • ISDN PRI SIP-T, Q.sig
  • Registration for SIP/IMS: with up to 256 SIP Accounts
  • ISDN SS7 NAT: Rport R2 MFC, Dynamic NAT
  • Local/Visible Ring Back Tone
  • Configuration of the Web GUI
  • Overlapping Dialing Data Backup/Restore Dialing Rules, with up to 2000 PSTN connections Statistics on Phone Calls
  • Codecs for Voice Statistics for Group SIP Trunk Calls
  • Lists of Access Rules
  • TFTP/Web Radius Firmware Upgrade
  • SNMP versions 1, 2, and 3
  • G.711a/law, G.723.1, G.729AB, iLBC, AMR Network Capture are examples of voice codecs.
  • Suppression of Silence
  • Syslog: Debug, Information, Error, Warning, and Notice
  • CNG, VAD, and Jitter Buffer
  • Call history is recorded using Syslog Echo Cancellation (G.168) and up to 128ms NTP synchronization.
  • T.38 and the Centralized Management System with Pass-Through


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