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Dinstar MTG5000

Digital VoIP Gateway

Supplier in Dubai

High Density Digital VoIP Gateway

At Logixer, we supply the Dinstar MTG5000 High Density Digital VoIP Gateway in Dubai, UAE, a carrier-grade solution specifically designed for large enterprise networks, call centers, and telecom service providers to interconnect with E1/T1 network interfaces. Engineered for high-performance call control and system stability, the MTG5000 supports high-density calls while providing reliable carrier-grade VoIP and FoIP services. It also includes advanced features such as fax modem support and smart voice recognition, making it an ideal choice for organizations seeking scalable, secure, and efficient communication solutions.

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MTG5000 Digital VoIP Gateway Supplier in Dubai

MTG5000 Digital VoIP Gateway Supplier in Dubai

E1/T1
MTG5000 Digital VoIP Gateway Supplier in Dubai
NGN/IMS
MTG5000 Digital VoIP Gateway Supplier in Dubai
PRI

MTG5000 Digital VoIP Gateway Supplier in Dubai

SNMP

MTG5000 Digital VoIP Gateway Supplier in Dubai
SS7
MTG5000 Digital VoIP Gateway Supplier in Dubai
T.38/T.30

High Capacity Digital VoIP Gateway for Carriers & ITSPs

 


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  •     64  E1/T1 ports
  •      Up to 1920 simultaneous calls 
  •     Redundancy Dual MCU units
  •    Dual Power Supplies
  •     Flexible routing 
  •    Multiple SIP trunks
  •   Fully compatible with mainstream VoIP platforms
MTG5000 Digital VoIP Gateway Supplier in Dubai

Easy to Use & Manage  

 Intuitive Web Interface

 Supports SNMP

 Automated Provisioning

 Dinstar Cloud Management System

 Configuration Backup & Restore

 Advanced Debug Tools 

MTG5000 Digital VoIP Gateway Supplier in Dubai
MTG5000 Digital VoIP Gateway Supplier in Dubai

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 supporting E1/T1 interfaces, carrier-grade VoIP/FoIP services, fax modem support, and smart voice recognition for enterprises.  

  64 E1/T1 Ports

 4 Digital Processing Unit (DTU), each support 480 channels

 Codecs: G.711A/U, G.723.1, G.729A/B and 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

 SIP Trunk Work Mode: Peer/Access

 Adaptive Dynamic Buffer

 SIP/IMS Registration: with up to 2000 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

 Centralized Management System

  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

 512 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

 Call History Records via Syslog

 IP Trunk Priority

 Radius

 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

 DSNMP v1/v2/v3

 Caller and Called Number White Lists

 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

 NTP Synchronization

  64 E1/T1 Ports

 4 Digital Processing Unit (DTU), each support 480 channels

 Codecs: G.711A/U, G.723.1, G.729A/B and 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

 SIP Trunk Work Mode: Peer/Access

 Adaptive Dynamic Buffer

 SIP/IMS Registration: with up to 2000 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

 Centralized Management System

  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

 512 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

 Call History Records via Syslog

 IP Trunk Priority

 Radius

 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

 DSNMP v1/v2/v3

 Caller and Called Number White Lists

 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

 NTP Synchronization

Reliable Dinstar MTG5000 Digital VoIP Gateway Supplier in Dubai

 Enhance your communication network with the Dinstar MTG5000 from Logixer – contact us today for reliable high-density VoIP solutions in Dubai!

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