Description
Dinstar MTG2000 Carrier-grade Digital VoIP Gateway Dubai
Dinstar MTG2000 Dubai is a carrier-grade, intelligent Digital VoIP Gateway designed to deliver robust performance, offering scalability from 4 to 20 E1/T1 ports to support diverse deployment requirements. Engineered for reliability, it delivers high-quality VoIP and FoIP services along with advanced value-added capabilities, including modem support and enhanced voice recognition functions. Built with strong maintainability, manageability, and operational efficiency, the Dinstar MTG2000 Dubai provides a flexible and future-driven communication framework suitable for evolving network environments. It supports a broad range of signaling protocols and seamlessly bridges SIP with traditional systems such as ISDN PRI and SS7, optimizing trunking resources while preserving excellent voice quality. With comprehensive codec support, secure signal encryption, and intelligent voice processing technologies, the MTG2000 serves as an ideal solution for large enterprises, call centers, service providers, and telecom operators seeking high-performance and scalable communication infrastructure.
Features
Scalable Digital VoIP Gateway for Service Providers
- 4 to 20 ports E1/T1 in 1U chassis
- Up to 600 simultaneous calls
- Redundancy Dual MCU 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
- 4/8/12/16/20 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
- Centralized Management System
