MTG3000

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

Get a Quote

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 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

Intuitive Web interface
Support SNMP
Automated provisioning
Dinstar Cloud Management System
Configuration Backup & Restore
Advanced Debug tools

Product Features

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

All Parameters Product Comparison