DINSTAR MTG3000 High Density Digital VoIP Gateway
DINSTAR MTG3000 High Density Digital VoIP Gateway
DINSTAR MTG3000 High Density Digital VoIP Gateway

DINSTAR MTG3000 High Density Digital VoIP Gateway: Empowering Reliable and Scalable Carrier-Grade Communications

Experience the power of DINSTAR MTG3000, a high-density digital VoIP gateway designed for carriers and telecom operators. Offering scalability from 16 to 63 ports E1/T1 with an STM-1 interface, this carrier-grade solution ensures superior voice and fax over IP (FoIP) services. With built-in value-added features like modem support and voice recognition, MTG3000 delivers a reliable, high-performance communication network.

Designed with a focus on maintainability, manageability, and operability, the MTG3000 supports a wide range of signaling protocols, facilitating seamless interconnections between SIP and traditional signals such as ISDN PRI/SS7. By optimizing trunking resources, voice quality remains uncompromised. The gateway supports multiple voice codecs, secure signal encryption, and advanced voice recognition technology, making it ideal for various applications within the service provider and telecom operator domains.

Product Features:

  • Scalable architecture supporting 16 to 63 ports E1/T1 with STM-1 interface
  • Carrier-grade VoIP and FoIP services
  • Value-added functions including modem and voice recognition
  • Wide range of signaling protocol support for SIP and traditional signals
  • Efficient utilization of trunking resources without compromising voice quality
  • Multiple voice codecs for enhanced flexibility
  • Secure signal encryption for heightened communication security
  • Advanced voice recognition technology for streamlined operations

Key Features
    • High Density Port Configuration
      16 to 63 ports E1/T1 in 2U chassis, STM-1 interface
    • Scalable Simultaneous Calls
      Up to 1890 simultaneous calls
    • Redundant Dual MCU Units
      Redundancy with dual MCU units
    • Dual Power Supply
      Dual power supplies for enhanced reliability
    • Flexible Routing Options
      Supports flexible routing configurations
    • Multiple SIP Trunks
      Capability to handle multiple SIP trunks
    • Compatibility with Mainstream VoIP Platforms
      Fully compatible with popular VoIP platforms
  • E1/T1
    E1/T1
  • T.38/T.30
    T.38/T.30
  • PRI
    PRI
  • SS7
    SS7
  • NGN/IMS
    NGN/IMS
  • SNMP
    SNMP
  • HD Audio

Rich Experiences on PSTN Protocols

  • ISDN PRI Support: Supports ISDN PRI signaling protocol
  • ISDN SS7 with Redundancy: Supports ISDN SS7 signaling protocol with redundant SS7 links
  • R2 MFC Compatibility: Compatible with R2 MFC signaling protocol
  • T.38 and Fax Pass-through: Supports T.38 protocol for reliable fax transmission and pass-through fax functionality
  • Modem and POS Machine Support: Enables connectivity and compatibility with modems and POS machines
  • Seamless Integration with Legacy PBXs and PSTN Networks: Over 10 years of experience in integrating with a wide range of Legacy PBX systems and service providers' PSTN networks

Streamlined Administration: Ease of Management

Empowering Connections, Enhancing Business
  • Intuitive Web Interface: User-friendly web interface for easy management and configuration
  • SNMP Support: Enables monitoring and management using SNMP (Simple Network Management Protocol)
  • Automated Provisioning: Streamlines the provisioning process with automated configuration and deployment
  • Dinstar Cloud Management System: Integrated with the Dinstar Cloud Management System for centralized management and monitoring
  • Configuration Backup & Restore: Allows easy backup and restoration of configurations for hassle-free maintenance
  • Advanced Debug Tools: Provides advanced debugging capabilities for efficient troubleshooting and diagnostics

Features

Enhance Communication with Advanced Features
    • 1+1 Redundant Main Control Unit (MCU): Redundant main control unit for enhanced system reliability
    • 4 Digital Processing Units (DTU): Four DTUs, each supporting 512 channels
    • Dual Power Supplies: Dual power supply units for increased power redundancy
    • 2 GE Interfaces: Two Gigabit Ethernet interfaces for high-speed connectivity
    • SIP v2.0 Support: Support for SIP version 2.0
    • SIP-T, RFC3372, RFC3204, RFC3398: Support for SIP-T protocol and relevant RFCs
    • SIP Trunk Work Mode: Configurable work mode for SIP trunks: Peer/Access
    • SIP/IMS Registration: Support for up to 256 SIP accounts for SIP/IMS registration
    • NAT: Dynamic NAT, Rport: Support for dynamic NAT and Rport for NAT traversal
    • Flexible Route Methods: Flexible routing methods including PSTN-PSTN, PSTN-IP, IP-PSTN
    • Intelligent Routing Rules: Intelligent routing rules for optimized call routing
    • Call Routing based on Time: Call routing based on time conditions
    • Call Routing based on Caller/Called Prefixes: Call routing based on caller/called number prefixes
    • 256 Route Rules for each Direction: Up to 256 route rules for each direction
    • Caller and Called Number Manipulation: Manipulation of caller and called numbers for customized routing
    • Local/Transparent Ring Back Tone: Local or transparent ring back tone support
    • Overlapping Dialing: Support for overlapping dialing scenarios
    • Dialing Rules: Support for up to 2000 dialing rules
    • PSTN Grouping: Grouping of PSTN lines by E1 port or E1 timeslot
    • IP Trunk Group Configuration: Configuration of IP trunk groups
    • Voice Codecs Group: Grouping of voice codecs for efficient codec management
    • Caller and Called Number White Lists: Whitelisting of caller and called numbers
    • Caller and Called Number Black Lists: Blacklisting of caller and called numbers
    • Access Rule Lists: Lists for defining access rules
    • IP Trunk Priority: Configurable priority for IP trunks
    • Radius Support: Support for RADIUS protocol
    • Up to 63 E1s/T1s, STM-1 Interface: Support for up to 63 E1s/T1s with an STM-1 interface
    • Voice Codecs: Support for G.711a/μ law, G.723.1, G.729A/B, iLBC 13k/15k, AMR codecs
    • Silence Suppression: Support for silence suppression
    • Comfort Noise: Comfort noise generation for improved voice quality
    • Voice Activity Detection: Voice activity detection for efficient bandwidth utilization
    • Echo Cancellation (G.168): Echo cancellation with up to 128ms tail length (G.168)
    • Adaptive Dynamic Buffer: Adaptive dynamic buffer for optimized voice packet handling
    • Voice and Fax Gain Control: Control of voice and fax gain levels
    • FAX Support: Support for T.38 protocol and pass-through fax
    • Modem/POS Support: Support for modem and POS connectivity
    • DTMF Mode: Configurable DTMF mode (RFC2833/SIP Info/In-band)
    • Clear Channel/Clear Mode: Support for clear channel and clear mode operation
    • ISDN PRI Support: Support for ISDN PRI signaling protocol
    • Signal 7/SS7 Support: Support for Signal 7/SS7 protocols including ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
    • R2 MFC Compatibility: Compatibility with R2 MFC signaling protocol
    • Web GUI Configuration: Configuration management via web-based GUI
    • Data Backup/Restore: Backup and restore of configuration data
    • PSTN Call Statistics: Call statistics for PSTN calls
    • SIP Trunk Call Statistics: Call statistics for SIP trunk calls
    • Firmware Upgrade via TFTP/Web: Firmware upgrade using TFTP or web-based methods
    • SNMP v1/v2/v3: Support for SNMP versions 1, 2, and 3
    • Network Capture: Network capture for detailed packet analysis
    • Syslog Support: Support for syslog messages at various levels (Debug, Info, Error, Warning, Notice)
    • Call History Records via Syslog: Call history records accessible via syslog
    • NTP Synchronization: NTP synchronization for accurate timekeeping
    • Centralized Management System: Integration with a centralized management system for unified management and monitoring
Clearing Doubts, Enhancing Experiences

Frequently Asked Questions

What does the HD Voice of this IP phone offer?

The HD Voice feature offers superior sound quality, providing clear and crisp audio for all your calls. It aims to significantly reduce background noise and improve speech clarity, ensuring a more effective communication experience.

Can I use multiple accounts with this IP phone?

Yes, this IP phone supports 2 extension accounts. This multi-line capability allows for increased flexibility, enabling the handling of multiple calls simultaneously.

Can I expand the number of ports on the DINSTAR MTG3000?

Yes, the DINSTAR MTG3000 is scalable and can be expanded from 16 to 63 ports E1/T1.

What signaling protocols are supported by the MTG3000?

The MTG3000 supports a wide range of signaling protocols, including ISDN PRI, ISDN SS7, R2 MFC, and SIP v2.0.

Can the MTG3000 handle simultaneous calls?

Yes, the MTG3000 can support up to 1890 simultaneous calls.

Does the MTG3000 support redundancy?

Yes, the MTG3000 features redundancy with dual Main Control Unit (MCU) and dual power supplies for enhanced reliability.

What are the supported voice codecs on the MTG3000?

The MTG3000 supports a variety of voice codecs, including G.711a/μ law, G.723.1, G.729A/B, iLBC 13k/15k, and AMR.

Can the MTG3000 handle fax transmission?

Yes, the MTG3000 supports fax transmission with T.38 protocol and pass-through fax functionality.

Is the MTG3000 compatible with legacy PBX systems?

Yes, the MTG3000 has over 10 years of experience in integrating with a wide range of legacy PBX systems and service providers' PSTN networks.

Does the MTG3000 support remote management?

Yes, the MTG3000 can be remotely managed through an intuitive web interface and also supports SNMP for monitoring and management.

Can I configure call routing based on specific criteria?

Yes, the MTG3000 allows intelligent call routing based on time, caller/called number prefixes, and configurable routing rules.

What security features are available on the MTG3000?

The MTG3000 supports secure signal encryption, caller and called number whitelists/blacklists, and access rule lists for enhanced security.

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    114 Lavender Street, #07-51, CT Hub 2, Singapore 338729
  • Phone Number:
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  • Email:
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