Description

Dinstar MTG1000 Low Density E1/T1 Digital VoIP Gateway Dubai

The Dinstar MTG1000 Dubai E1/T1 Digital VoIP Gateway is a compact, high-performance, and cost-effective trunk gateway engineered to seamlessly bridge PSTN and IP networks for organizations seeking reliable convergence between traditional telephony and modern VoIP systems. Available with 1 or 2 E1/T1 ports, the device incorporates a robust hardware design that ensures comprehensive PSTN access while supporting advanced SIP-to-SIP interworking to facilitate smooth, uninterrupted communication between diverse network elements. Powered by a dedicated DSP processor and an efficient system architecture, the Dinstar MTG1000 Dubai delivers exceptional performance when converting PCM voice signals into IP packets, consistently maintaining operational stability and high voice quality even under maximum load conditions. Building on Dinstar’s extensive expertise with ISDN PRI, SS7, and R2 MFC signaling technologies, the MTG1000 is fully interoperable with major VoIP platforms, PBXs, and digital trunk interfaces, making it a flexible, scalable, and dependable solution for enterprises looking to enhance their digital communication infrastructure.

Features

Cost-effective VoIP Trunk Gateway

• 1/2 ports E1/T1 in 1U chassis
• Dual Power Supplies
• Up to 60 simultaneous calls
• 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

• 1/2 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, Q.sig
• 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