Description
Dinstar MTG5000 High-Density Digital VoIP Gateway is a next-generation intelligent VoIP gateway that is specifically designed for connecting with E1/T1 network interfaces in large enterprise networks, call centers, and telecom service providers. It is designed with powerful call control features and maintenance tools in mind. MTG5000 supports high-density calls while maintaining a very stable system. It also offers carrier-grade VoIP and FoIP services, as well as value-added features like fax modem and voice recognition. The MT5000 supports a variety of signaling protocols, including SIP and traditional PSTN signaling protocols such as SS7 and PRI. It supports a variety of codec formats, as well as signaling encryption and an ASR interface. MTG500 can be combined with a large call center solution, a telephone system or IP PBX, and an IP/PSTN call service provider.
High Capacity Digital VoIP Gateway for Carriers & ITSPs:- 64 E1/T1 ports high capacity digital VoIP gateway for carriers and ITSPs There can be up to 1920 concurrent calls. Dual Power Sources Routing flexibility Several SIP trunks Completely compatible with popular VoIP platforms
Key features:- Carrier-grade hardware design, 1+1 power supply, and hardware-based HA are key features. Highly scalable and compact design structure, support up to 64 E1 ports(MAX) in 3.5U size • Support flexible dialing rules and operations, allowing users to customize dialing rules according to different \ scall routing environments Support multiple codec formats: G.711A/U, G.723.1, G.729A/B and iLBC Very good compatibility, and be workable with Avaya PBX, NEC and Alcatel, and large soft-switch \from Huawei, Cisco and ZTE, etc.
Features
- There are 64 E1/T1 ports.
- Each DTU (Digital Processing Unit) supports 480 channels.
- G.711A/U, G.723.1, G.729A/B, and iLBC are the codecs used.
- Dual Power Sources
- Suppression of Silence
- 2 GE
- Noise that provides comfort
- Voice Activity Detection in SIP v2.0
- Echo Cancellation (G.168), SIP-T, RFC3372, RFC3204, RFC3398, with up to 128ms SIP Trunk Mode of Work: Peer/Access
- Adaptive The Dynamic Buffer
- Registration for SIP/IMS: up to 2000 SIP Accounts
- Voice and Fax Gain Control NAT: Dynamic NAT, T.38 Rport FAX, and Flexible Pass-Through 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
- Route Rules for ISDN PRI 512 in Each Direction
- ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP, Signal 7/SS7
- Manipulation of the Caller and the Called Number
- R2 MFC
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