Description
Dinstar MTG2000B-4E1 High Availability Digital VoIP Gateway Dubai
The Dinstar MTG2000B-4E1 Dubai is a High Availability (HA), carrier-grade digital VoIP gateway engineered with redundant MCUs, dual power supplies, and a scalable framework supporting up to 4 E1/T1 ports, ensuring continuous and resilient operations across diverse environments. It provides dependable carrier-grade VoIP and FoIP services while incorporating advanced value-added capabilities such as integrated modem support, intelligent voice recognition, and enhanced signaling interoperability. Built for streamlined maintenance, simplified management, and stable long-term performance, the gateway offers users a flexible, efficient, and forward-looking communications infrastructure suitable for evolving business requirements. With comprehensive signaling protocol support, it enables effortless interworking between SIP and traditional telephony networks including ISDN PRI and SS7, thereby maximizing trunking resource utilization and delivering consistently high voice quality. Combined with support for multiple voice codecs, secure encryption mechanisms, and robust voice processing features, the MTG2000B-4E1 is ideal for large enterprises, call centers, service providers, and telecom operators that require a versatile, scalable, and reliable communication platform.
Features
Scalable Digital VoIP Gateway for Service Providers
- 4 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
- 4 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
