Description

Dinstar MTG2000-16E1 Carrier-grade Digital VoIP Gateway Dubai

The Dinstar MTG2000-16E1 Dubai is a high-performance, carrier-grade intelligent Digital VoIP Gateway featuring 16 E1/T1 interfaces, engineered to deliver exceptional VoIP and FoIP services while integrating a comprehensive suite of advanced value-added capabilities, including modem functionality, voice recognition, and intelligent processing features. Designed with a strong focus on reliability, simplified maintenance, and flexible management options, it provides enterprises with a scalable, efficient, and future-ready communications backbone. Supporting an extensive array of signaling protocols, the MTG2000-16E1 enables smooth and reliable interworking between SIP platforms and traditional signaling environments such as ISDN PRI and SS7, thereby improving trunking resource efficiency and preserving consistently high voice quality. With support for multiple voice codecs, secure signaling encryption, intelligent routing mechanisms, and enhanced voice optimization technologies, the Dinstar MTG2000-16E1 Dubai stands out as an ideal digital communication gateway for large enterprises, call centers, service providers, and telecom operators that require a robust, secure, and fully integrated voice infrastructure.

Features

Scalable Digital VoIP Gateway for Service Providers

• 16 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

• 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
• Centralized Management System