Description

Dinstar MTG2000-12E1 Carrier-grade Digital VoIP Gateway Dubai

The Dinstar MTG2000-12E1 Dubai is a robust, high-performance, carrier-grade intelligent Digital VoIP Gateway featuring 12 E1/T1 interfaces, engineered to deliver superior VoIP and FoIP services while integrating a broad spectrum of advanced value-added functions, including modem capabilities, voice recognition, and enhanced signaling intelligence. Designed with exceptional maintainability, operability, and centralized manageability, it provides a flexible, efficient, and future-proof communication framework ideal for evolving network infrastructures. Supporting an extensive range of signaling protocols, the MTG2000-12E1 ensures seamless interoperability between SIP-based platforms and legacy telephony systems such as ISDN PRI and SS7, thereby maximizing trunking efficiency and maintaining consistently high voice quality across diverse environments. With multi-codec support, secure signaling encryption, intelligent voice processing, and scalable architecture, the MTG2000-12E1 stands out as an optimal solution for large enterprises, call centers, telecom service providers, and operators requiring a dependable, feature-rich, and scalable digital communication gateway.

Features

Scalable Digital VoIP Gateway for Service Providers

• 12 E1/T1 in 1U chassis
• Up to 600 simultaneous calls
• Redundancy Dual MCU 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
• Centralized Management System