Description

Dinstar MTG2000B-12E1 High Availability Digital VoIP Gateway Dubai

The Dinstar MTG2000B-12E1 Dubai is a robust, high-availability (HA) carrier-grade digital VoIP gateway engineered with redundant MCUs and dual power supplies to ensure continuous service uptime while supporting scalable expansion of up to 12 E1/T1 ports. Designed to deliver dependable VoIP and FoIP performance, it integrates advanced capabilities such as modem passthrough, intelligent voice-processing technologies, and enhanced voice recognition to improve operational effectiveness. Its architecture emphasizes maintainability, centralized manageability, and long-term efficiency, enabling organizations to build a flexible and future-oriented communication infrastructure. With comprehensive support for multiple signaling protocols, the MTG2000B-12E1 facilitates seamless interoperability between SIP networks and legacy telephony systems including ISDN PRI and SS7, thereby optimizing trunk utilization without compromising audio clarity. Equipped with a wide range of voice codecs, secure encryption for signaling, and high-precision voice recognition features, this gateway is ideally positioned for deployments within large enterprises, contact centers, telecom operators, and service providers seeking reliable, scalable, and secure voice connectivity.

Features

Scalable Digital VoIP Gateway for Service Providers

• 12 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

• 12 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