Description

Dinstar MTG2000B High Availability Digital VoIP Gateway Dubai

The Dinstar MTG2000B Dubai is a high-availability (HA), carrier-grade digital VoIP gateway engineered with redundant MCUs, dual power supplies, and a scalable architecture supporting 4 to 16 E1/T1 ports to ensure robust performance and operational continuity. It delivers dependable VoIP and FoIP services while incorporating advanced value-added capabilities such as modern modem compatibility, enhanced voice recognition, and intelligent system controls that improve overall communication efficiency. Built for superior maintainability, simplified management, and smooth long-term operation, the MTG2000B provides organizations with a highly flexible and future-ready communication framework. With comprehensive support for a wide range of signaling protocols, it seamlessly bridges SIP with legacy systems including ISDN PRI and SS7, thereby optimizing trunking capacity and preserving excellent voice fidelity. Equipped with multiple high-quality voice codecs, secure signaling encryption, and sophisticated voice recognition features, the MTG2000B stands out as an ideal choice for large enterprises, call centers, service providers, and telecom operators that require reliable, scalable, and high-performance communication infrastructure.

Features

Scalable Digital VoIP Gateway for Service Providers

• 4 to 16 ports E1/T1 in 1U chassis
• Up to 480 simultaneous calls
• Redundant MCUs (Main Control Unit)
• 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

Technical Highlights

• 4/8/12/16/ E1s/T1s, RJ48 interface
• 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