Dinstar MTG2000
Digital VoIP Gateway
Supplier in Dubai
At Logixer, we supply the Dinstar MTG2000, a carrier-grade intelligent Digital VoIP Gateway scalable from 4 to 20 E1/T1 ports, delivering high-performance VoIP and FoIP services along with advanced features such as modem support and voice recognition. Designed for flexibility, efficiency, and future-ready communications, it supports a wide range of signaling protocols to ensure seamless interconnection between SIP and traditional signals like ISDN PRI and SS7, optimizing trunking resources while maintaining exceptional voice quality. With multiple voice codecs, secure signal encryption, and smart voice recognition technology, the MTG2000 is the ideal solution for large enterprises, call centers, telecom operators, and service providers in Dubai and across the UAE.
Contact Now
E1/T1
NGN/IMS
PRI
SNMP
SS7
T.38/T.30
Scalable Digital VoIP Gateway for Service Providers
- 4 to 20 ports 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
Easy to Use & Manage
Intuitive Web Interface
Supports SNMP
Automated Provisioning
Dinstar Cloud Management System
Configuration Backup & Restore
Advanced Debug Tools
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
Features
Scalable from 4 to 20 E1/T1 ports, the Dinstar MTG2000 offers carrier-grade VoIP and FoIP services with advanced signaling, encryption, and voice recognition.
4/8/12/16/20 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
IP Trunk Priority
NTP Synchronization
Radius
Centralized Management System
Access Rule Lists
Call History Records via Syslog
Syslog: Debug, Info, Error, Warning , Notice
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 Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
256 Route Rules for each Direction
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
4/8/12/16/20 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
IP Trunk Priority
NTP Synchronization
Radius
Centralized Management System
Access Rule Lists
Call History Records via Syslog
Syslog: Debug, Info, Error, Warning , Notice
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 Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
256 Route Rules for each Direction
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
Reliable Dinstar MTG2000 Digital VoIP Gateway Supplier in Dubai
Upgrade your communication network with the trusted Dinstar MTG2000 Digital VoIP Gateway Supplier in Dubai – Contact Logixer today for the best solutions!