Dinstar MTG5000 E1 VoIP Gateway

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MTG5000 is a carrier-grade Digital VoIP gateway, and it is purposely designed for large enterprise networks, call centers, and Telecom service providers to connect with E1/T1 network interfaces. It is developed with the aspect of powerful call control features and maintenance tools. MTG5000 supports high density calls with a very stable system support. It also provides carrier grade VoIP and FoIP services, as well as value added functions such as fax modem and voice recognition service.

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Availability: In stock SKU: MTG5000 Category:

 

High Capacity Digital VoIP Gateway for Carriers & ITSPs

  • 64  E1/T1 ports
  • Up to 1920 simultaneous calls
  • 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 and Pass-through fax
  • Support modem and POS machines
  • More than 10-year experiences to integrate with a wide range of Legacy PBXs / Service providers’ PSTN networks

Easy Management

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  • Intuitive Web interface
  • Support SNMP
  • Automated provisioning
  • Dinstar Cloud Management System
  • Configuration Backup & Restore
  • Advanced Debug tools
  • 64 E1/T1 Ports
  • 4 Digital Processing Unit (DTU), each support 480 channels
  • Codecs: G.711A/U, G.723.1, G.729A/B and iLBC
  • 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 2000 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
  • 512 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