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MTG200
Cost-effective VoIP Trunk Gateway
MTG200 series Digital VoIP Gateways with 1/2 ports E1/T1 simply migrate your legacy PSTN networks (legacy PBX or E1/T1 service providers) , to VoIP network. Only small investment, you can enjoy the real benefits of VoIP, and retain your PSTN connectivity. It is a compact box designed for SMEs and open-source market, fully compatible with Asterisk / Elastix / Trixbox / Freeswitch and mainstream VoIP platform. With support of ISDN PRI / SS7 / R2 MFC, integrating with your legacy PBX or PSTN network is also so easy.
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  • E1/T1
  • T.38/T.30
  • PRI
  • SS7
  • R2
  • NGN/IMS
Detailed parameters
  • 1/2 E1s/T1s, RJ48 interface
  • Dual Power Supplies
  • 2 GE
  • SIP v2.0
  • SIP-T,RFC3372, RFC3204, RFC3398
  • SIP Trunk Work Mode: Peer/Access
  • SIP/IMS Registration :with up to 256 SIP Accounts
  • NAT: Dynamic NAT, Rport
  • 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
  • Local/Transparent Ring Back Tone
  • Overlapping Dialing
  • PSTN group by E1 port or E1 Timeslot
  • IP Trunk Group Configuration
  • Voice Codecs Group
  • Caller and Called Number White Lists
  • Access Rule Lists
  • IP Trunk Priority
  • Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC 13k/15k,AMR
  • Silence Suppression
  • ISDN PRI, Q.sig
  • Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
  • R2 MFC
  • Web GUI Configuration
  • Data Backup/Restore
  • PSTN Call Statistics
  • SIP Trunk Call Statistics
  • Firmware Upgrade via TFTP/Web
  • SNMP v1/v2/v3
  • Network Capture
  • Syslog: Debug, Info, Error, Warning , Notice
  • Call History Records via Syslog
  • NTP Synchronization
  • Centralized Management System
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