Description
Dinstar MTG3000T High Density Trancoding Gateway Dubai
The Dinstar MTG3000T Dubai is a powerful, versatile, and high-performance transcoding gateway designed to manage demanding voice networks, supporting up to 1,568 concurrent transcoding sessions with exceptional efficiency. It seamlessly converts multiple channels of voice data across a wide spectrum of commonly used codecs such as G.711A/U, G.723.1, G.729A/B, iLBC, G.726, and AMR, enabling smooth and uninterrupted IP-to-IP communication between diverse systems. By effectively bridging codec and capability differences within complex telecommunications environments, the MTG3000T delivers reliable, stable, and consistent performance, making it a preferred solution for organizations that require scalable, robust, and flexible voice transcoding operations.
Features
High Capacity Transcoding Gateway
- Transcoding from IP to IP
- Up to 2048 VoIP Sessions
- Dual Power Supplies
- Scalable by 4 DTUs Boards
- Multiple SIP Trunks
- Fully Compatible with Mainstream VoIP Platforms
Rich Experiences on PSTN Protocols
- 2U Size
- T.38 and Pass-through fax
- Support modem and POS machines
- Flexible dialling rules, thus adapting to different requirements of different environments.
- More than 10-year experiences to integrate with a wide range of Legacy PBXs/Service providers’ PSTN networks
Easy ManagementÂ
- Intuitive Web interface
- Support SNMP
- Automated provisioning
- Dinstar Cloud Management System
- Configuration Backup & Restore
- Advanced Debug tools
Technical Highlights
- Up to 4 Digital Processing Unit (DTU)
- SIP Trunk Groups
- 2 GE ports
- 256 SIP trunks
- Redundant Power Supply
- Outbound Proxy supported
- G.711—G.711: 2048 sessions
- Maximum 256 SIP accounts
- G.711—G.729: 1568 sessions
- Cloud based management system
- G.711—G.723: 1344 sessions
- Web GUI management
- G.711—G.726: 2048 sessions
- SNMP
- G.711—iLBC: 960 sessions
- Firmware Upgrade via TFTP/FTP/HTTP
- G.711— AMR: 832 sessions
- Support configuration backup/restore
- G.723—G.729: 896 sessions
- Local maintenance via console
- SIP, SIP-T
- Call trace/syslog
- SIP Trunk Work Mode: Peer/Access
- Call test
- SIP/IMS Registration: with up to 256 SIP Accounts
- Network capture
- NAT: Dynamic NAT, Rport
- Signaling hunter
- Caller/Called Number Black lists
- Voice codecs: G.711a/μ law,G.723.1,G.729A/B, iLBC, AMR
- Caller/Called Number White lists
- FAX: T.38 and Pass-through
- IP access rule list
- Support Modem/POS

