Description

Dinstar MTG2000-8E1 Carrier-grade Digital VoIP Gateway Dubai

The Dinstar MTG2000-8E1 Dubai is a high-performance, carrier-grade intelligent Digital VoIP Gateway featuring 8 E1/T1 interfaces, engineered to deliver exceptional VoIP and FoIP services while incorporating a range of advanced value-added capabilities, including modem functionality, voice recognition, and enhanced signaling support. Designed with superior maintainability, manageability, and operational efficiency, the MTG2000-8E1 provides a flexible, scalable, and future-ready communication framework tailored to meet the evolving demands of modern enterprises. With broad compatibility across diverse signaling protocols, it ensures smooth and reliable interconnection between SIP networks and traditional telephony systems such as ISDN PRI and SS7, enabling organizations to optimize trunk resource allocation without compromising voice clarity or service reliability. Equipped with multiple high-quality voice codecs, secure signaling encryption, intelligent call handling, and robust voice processing, the Dinstar MTG2000-8E1 Dubai stands out as an ideal choice for large enterprises, call centers, service providers, and telecom operators seeking a dependable, scalable, and high-performance communication gateway solution.

Features

Scalable Digital VoIP Gateway for Service Providers

• 8 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

• 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
• Centralized Management System