Description

Dinstar MTG3000 High Density Digital VoIP Gateway Dubai

The Dinstar MTG3000 Dubai is a robust, carrier-grade Digital VoIP gateway engineered for large-scale deployments, offering scalable configurations that support 16 to 63 E1/T1 ports with an STM-1 interface to accommodate high-density traffic requirements. It delivers reliable, high-performance VoIP and FoIP services while integrating advanced value-added capabilities, including modem transmission support, intelligent voice recognition, and enhanced signaling adaptability. Designed to simplify ongoing maintenance, centralized management, and day-to-day operation, the MTG3000 ensures a stable and efficient communication framework for demanding environments. In addition, it provides comprehensive signaling protocol compatibility, enabling seamless interoperability between SIP platforms and traditional systems such as ISDN PRI and SS7, thereby optimizing trunk utilization without compromising voice fidelity. With an extensive suite of voice codecs, secure signaling encryption, and adaptive processing technologies, the MTG3000 is ideally suited for service providers, enterprise carriers, and telecom operators seeking a flexible, scalable, and high-quality solution for modern communication networks.

Features

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

Technical Highlights

• 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