Dinstar MTG2000B
Digital VoIP Gateway
Supplier in Dubai
At Logixer, we supply the Dinstar MTG2000B High Availability (HA) carrier-grade digital VoIP gateway in Dubai, UAE, designed for mission-critical communications with redundant MCUs and power supplies, scalable from 4 to 16 E1/T1 ports. Delivering exceptional VoIP and FoIP services along with advanced features like modem support and voice recognition, the MTG2000B supports a wide range of signaling protocols for seamless interconnection between SIP and traditional signals such as ISDN PRI and SS7, optimizing trunking resources while maintaining superior voice quality. With multiple voice codecs, secure signal encryption, and smart voice recognition technology, it is ideal for large enterprises, call centers, service providers, and telecom operators seeking high reliability, easy maintenance, and future-ready communication solutions.
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E1/T1
NGN/IMS
PRI
SNMP
SS7
T.38/T.30
Scalable Digital VoIP Gateway for Service Providers
- 4 to 16 ports 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
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
High Availability VoIP gateway with redundant MCUs and power, scalable 4–16 E1/T1 ports, multi-protocol support, and superior voice quality.
4/8/12/16/ 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
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/ 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 MTG2000B Digital VoIP Gateway Supplier in Dubai
Upgrade your communication network today with the Dinstar MTG2000B from Logixer – contact us now for expert supply and support in Dubai!