Dinstar MTG3000
Digital VoIP Gateway
Supplier in Dubai
High Density Digital VoIP Gateway
At Logixer, we supply the Dinstar MTG3000 High Density Digital VoIP Gateway in Dubai, UAE, a carrier-grade solution scalable from 16 to 63 E1/T1 ports with an STM-1 interface, designed for large-scale and mission-critical communications. Delivering high-performance VoIP and FoIP services along with advanced features like modem support and smart voice recognition, the MTG3000 ensures reliable, maintainable, and secure communication for service providers and telecom operators. Supporting a wide range of signaling protocols, it enables seamless interconnection between SIP and traditional networks such as ISDN PRI and SS7, optimizing trunking efficiency while maintaining superior voice quality. With multiple voice codecs, robust signal encryption, and carrier-grade reliability, the MTG3000 is the ideal choice for scalable, secure, and future-ready communication networks in Dubai and across the UAE.
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E1/T1
NGN/IMS
PRI
SNMP
SS7
T.38/T.30
High Capacity Digital VoIP Gateway for Carriers & ITSPs
- 16 to 63 ports E1/T1 in 2U chassis, STM-1 interface
- Up to 1890 simultaneous calls
- Redundancy Dual MCU units
- 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-density VoIP gateway scalable from 16–63 E1/T1 ports with STM-1 interface, supporting SIP, ISDN PRI, SS7, secure encryption, and smart voice recognition.
1+1 Redundant Main Control Unit (MCU)
Up to 63 E1s/T1s, STM-1 interface
Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC
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
4 Digital Processing Unit (DTU), each support 512 channels
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
1+1 Redundant Main Control Unit (MCU)
Up to 63 E1s/T1s, STM-1 interface
Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC
4 Digital Processing Unit (DTU), each support 512 channels
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 MTG3000 Digital VoIP Gateway Supplier in Dubai
Upgrade your network with the Dinstar MTG3000 from Logixer – contact us today for reliable, high-performance VoIP solutions in Dubai!