Description
Dinstar MTG3000 supports a wide range of signaling protocols, allowing for the interconnection of SIP and traditional signals such as ISDN PRI / SS7 while maximizing trunking resource efficiency and voice quality. With multiple voice codes, secure signal encryption, and smart voice recognition technology, the MTG3000 is ideal for a wide range of service providers and telecom operators’ applications. MTG3000 is a carrier-grade VoIP gateway with high reliability and performance for telecom operators and ITSPs. MTG3000 uses an STM-1 interface with high integration and large capacity, focusing on the concept of maintainable, manageable, and operable. It offers carrier-grade VoIP and FoIP services, as well as value-added features like modem and voice recognition. As a result, it builds a flexible, high-efficiency, and future-oriented communication network for users.MTG3000 supports a variety of signaling protocols, allowing for the interconnection of SIP and traditional signals such as SS7 and PRI. It supports multiple codec methods, provides signal encryption technology and smart voice recognition technology, and improves trucking resource utilization efficiency while maintaining voice quality. The trunk gateway is ideal for ITSP, telecom operators, and large-scale enterprise networks. MTG3000 supports a variety of signaling protocols, allowing for the interconnection of SIP and traditional signals such as SS7 and PRI. It supports multiple codec methods, provides signal encryption technology and smart voice recognition technology, and improves trucking resource utilization efficiency while maintaining voice quality. The trunk gateway is ideal for ITSP, telecom operator, and large-scale enterprise networks.
Features
- Each DTU (Digital Processing Unit) supports 512 channels.
- G.711a/law, G.723.1, G.729A/B, iLBC 13k/15k, AMR codecs
- Dual Power Sources
- Suppression of Silence
- 2 GE
- Noise that provides comfort
- Voice Activity Detection in SIP v2.0
- SIP-T,RFC3372,RFC3204,RFC3398 Echo Cancellation (G.168),up to 128ms SIP Trunk Mode of Work: Peer/Access
- Adaptive The Dynamic Buffer
- Registration for SIP/IMS: with up to 256 SIP Accounts
- Voice and Fax Gain Dynamic NAT control, Rport FAX: T.38, and Pass-through\sFlexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN Modem/POS Support
- Rules for Intelligent Routing
- DTMF Protocol: RFC2833/SIP Info/In-band
- Time-based call routing
- Clear Mode/Clear Channel
- Routing calls based on caller/called prefixes
- ISDN PRI: Each Direction Has 256 Route Rules
- ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP, Signal 7/SS7
- Manipulation of the Caller and the Called Number
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