Description
Dinstar MTG200 Cost-effective VoIP Trunk Gateway Dubai
The Dinstar MTG200 Dubai Digital VoIP Gateway, offered with 1 or 2 E1/T1 ports, provides an efficient and reliable way to transition your traditional PSTN networks—whether connected through legacy PBXs or E1/T1 service providers—into a modern, feature-rich VoIP environment. With minimal investment, it enables you to retain your existing PSTN infrastructure while unlocking the full benefits of VoIP technology, including flexibility, scalability, and reduced communication costs. Designed as a compact, cost-effective gateway ideal for SMEs and open-source telephony deployments, the MTG200 ensures seamless compatibility with widely used platforms such as Asterisk, Elastix, Trixbox, Freeswitch, and other standard VoIP systems. Its support for ISDN PRI, SS7, and R2 MFC signaling protocols makes integration with legacy PBX setups or PSTN networks simple, smooth, and highly dependable, delivering a practical solution for businesses looking to upgrade without disruption.
Features
Cost-effective VoIP Trunk Gateway for SMEs
- 1/2 ports E1/T1
- Up to 60 simultaneous calls
- Flexible routing
- Multiple SIP trunks
- Fully compatible with Asterisk, FreeSWITCH and mainstream VoIP platforms
Rich Experiences on PSTN Protocols
- ISDN PRI
- ISDN SS7 (optional)
- 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
- 1/2 E1s/T1s, RJ48C interface
- Support Modem/POS
- 2 GE
- DTMF Mode: RFC2833/SIP Info/In-band
- SIP v2.0
- VLAN 802.1p/q
- SIP-T
- ISDN PRI, Q.sig
- SIP/IMS Registration :with up to 256 SIP Accounts
- ISDN SS7
- NAT: Dynamic NAT, Rport
- R2 MFC
- Local/Transparent Ring Back Tone
- Web GUI Configuration
- Overlapping Dialing
- Data Backup/Restore
- Dialing Rules，with up to 2000
- PSTN Call Statistics
- Voice Codecs Group
- SIP Trunk Call Statistics
- Access Rule Lists
- Firmware Upgrade via TFTP/Web
- Radius
- SNMP v1/v2/v3
- Voice Codecs:G.711a/μ law, G.723.1, G.729AB, iLBC,AMR
- Network Capture
- Silence Suppression
- Syslog: Debug, Info, Error, Warning , Notice
- CNG,VAD,Jitter Buffer
- Call History Records via Syslog
- Echo Cancellation (G.168),with up to 128ms
- NTP Synchronization
- T.38 and Pass-through
- Centralized Management System
