Description

Dinstar MTG2000B-16E1 High Availability Digital VoIP Gateway Dubai

The Dinstar MTG2000B-16E1 Dubai is a High Availability (HA), carrier-grade digital VoIP gateway engineered with redundant MCUs, dual power supplies, and scalable architecture supporting up to 16 E1/T1 ports for mission-critical operations. It delivers robust VoIP and FoIP services along with enhanced value-added capabilities, including modem transmission support, intelligent voice recognition, and optimized media handling to meet diverse deployment requirements. Built with strong emphasis on maintainability, manageability, and operational efficiency, the MTG2000B-16E1 enables businesses to implement flexible, future-ready communication networks that ensure consistent, high-quality performance. Its extensive support for signaling protocols facilitates seamless interoperability between SIP-based IP networks and legacy systems such as ISDN PRI and SS7, thereby improving trunking utilization while preserving excellent call quality. With compatibility for multiple high-fidelity voice codecs, secure encryption for signaling and media streams, and advanced voice processing technologies, this gateway is an ideal choice for large enterprises, call centers, service providers, and telecom operators seeking reliable, scalable, and high-performance communication infrastructure.

Features

Scalable Digital VoIP Gateway for Service Providers

• 16 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

• 16 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