Description

Dinstar MTG2000B-12E1 High Availability Digital VoIP Gateway Dubai

Dinstar MTG2000B-12E1 Dubai is a High Availability (HA) carrier-grade digital VoIP gateway equipped with redundant MCUs and redundant power supplies, scalable upto 12 ports E1/T1. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition. With highly maintainable, manageable and operable features, it offers a flexible, high-efficient, future-oriented communication network for users.

Dinstar MTG2000B-12E1 Dubai supports a wide-range of signaling protocols, realizing the interconnection between SIP and traditional signals like ISDN PRI / SS7, utilizing efficiency of trunking resources while ensuring voice quality. With multiple voice codes, secure signal encryption and smart voice recognition technology, MTG2000B is ideal for various applications of large enterprises, call centers, services providers and telecom operators.

Features

Scalable Digital VoIP Gateway for Service Providers

• 12 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

• 12 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