Description

Dinstar MTG2000B-8E1 High Availability Digital VoIP Gateway Dubai

The Dinstar MTG2000B-8E1 Dubai is a High Availability (HA), carrier-grade digital VoIP gateway engineered with redundant MCUs and dual power supplies, delivering stable performance and scalability for up to 8 E1/T1 ports within demanding communication environments. It ensures dependable VoIP and FoIP services while incorporating enhanced capabilities such as modem transmission, advanced voice recognition, and comprehensive resource optimization to support complex deployments. Designed for exceptional maintainability and streamlined management, it enables organizations to build a robust, high-performance, and future-oriented communication infrastructure. With extensive signaling protocol support, the MTG2000B-8E1 enables seamless interoperability between SIP networks and legacy systems including ISDN PRI and SS7, ensuring efficient trunk utilization and consistently high voice quality. Coupled with multiple voice codecs, secure signaling encryption, and intelligent voice-processing technologies, this gateway is an ideal solution for large enterprises, call centers, telecom operators, and service providers seeking reliable, scalable, and feature-rich digital voice connectivity.

Features

Scalable Digital VoIP Gateway for Service Providers

• 8 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

• 8 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