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.

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 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

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