Dinstar MTG2000 is a carrier-grade, intelligent Digital VoIP gateway, scalable from 4 to 20 E1/T1 ports. It provides powerful VoIP and FoIP functions (Fax over IP) as well as value-added features such as modem and voice recognition. With high maintainability, easy management, and operational reliability, the MTG2000 is the ideal solution for modern, efficient, and future-oriented communication networks.
The MTG2000 supports a wide range of signaling protocols and enables seamless interconnection between SIP and traditional protocols such as ISDN PRI or SS7. This ensures efficient use of trunking resources while maintaining high voice quality.
With multiple voice codecs, secure signaling encryption, and intelligent voice processing, the Dinstar MTG2000 is ideal for deployment in large enterprises, call centers, service providers, and telecom operators.
Key Features:
– Scalable from 4 to 20 E1/T1 ports
– Supports SIP, ISDN PRI, and SS7
– Carrier-grade voice quality and high reliability
– Redundant power supply
– Supports VoIP, FoIP, modem, and voice recognition
– Intelligent management (Web GUI, SNMP, Syslog)
– High efficiency and low operational costs
– Future-ready architecture
Applications:
– Telecom operators requiring E1/T1 trunking
– Large enterprises and call centers
– Service providers offering SIP trunking services
– Organizations migrating from ISDN/SS7 to IP-based communications
Conclusion: The Dinstar MTG2000 combines powerful VoIP technology, high scalability, and carrier-grade reliability in a compact 1U rack design – the perfect solution for businesses seeking reliability, flexibility, and future-proof communications.
Features
- 4/8/12/16/20 E1s/T1s, RJ48 interface
- Codecs: G.711 a/μ law, G.723.1, G.729A/B, iLBC 13k/15k, AMR
- Dual Power Supplies
- Silence Suppression
- 2 Gigabit Ethernet (GE) ports
- Comfort Noise
- SIP v2.0
- Voice Activity Detection
- SIP-T, RFC3372, RFC3204, RFC3398
- Echo Cancellation (G.168), up to 128ms
- SIP Trunk Work Mode: Peer/Access
- Adaptive Dynamic Buffer
- SIP/IMS Registration: 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 for Modem/POS
- Intelligent Routing Rules
- DTMF Mode: RFC2833 / SIP Info / In-band
- Call Routing based on Time
- Clear Channel / Clear Mode
- Call Routing based on Caller/Called Prefixes
- ISDN PRI: 256 Route Rules for each Direction
- Signaling 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: up to 2000
- PSTN Call Statistics
- PSTN grouped 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
