DINSTAR MTG1000 Low Density E1/T1 Digital VoIP Gateway
DINSTAR MTG1000 Low Density E1/T1 Digital VoIP Gateway

DINSTAR MTG1000 Low-Density E1/T1 Digital VoIP Gateway: Seamless Interconnection Between PSTN and IP Networks

The DINSTAR MTG1000 Low-Density E1/T1 Digital VoIP Gateway is a cutting-edge solution designed to facilitate smooth integration between traditional PSTN networks and IP-based communication systems. With its compact and cost-effective design, this gateway offers 1/2 ports for E1/T1 connections, providing comprehensive PSTN access capabilities and enabling seamless SIP-to-SIP interworking.


Featuring a powerful hardware design, the MTG1000 series ensures optimal performance in handling the interconversion of PCM voice signals and IP packets, even when the gateways are operating at full capacity. Equipped with a robust DSP processor, this gateway guarantees high efficiency and reliability, making it an ideal choice for various telecommunication projects.


With extensive interoperability with leading VoIP platforms and compatibility with digital trunk interfaces based on ISDN PRI/SS7/R2 MFC, the MTG1000 series offers exceptional flexibility and versatility. Its reliable performance and compatibility make it a valuable asset for diverse telecommunication environments.

Key Features
    • High Port Density
      1/2 ports E1/T1 in 1U chassis
    • Dual Power Supplies
      Provides redundant power source for enhanced reliability
    • Scalable Call Capacity
      Supports up to 60 simultaneous calls
    • Flexible Routing
      Offers versatile routing options for efficient call management
    • Multiple SIP Trunks
      Enables the use of multiple SIP trunks for increased connectivity
    • Compatibility with Leading VoIP Platforms
      Fully compatible with mainstream VoIP platforms for seamless integration
  • 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 Compatibility:  Enables seamless integration with ISDN PRI networks
  • ISDN SS7 with Redundancy:  Supports ISDN SS7 protocol with redundant links for enhanced reliability
  • R2 MFC Support:  Facilitates communication using R2 MFC signaling protocol
  • T.38 and Pass-through Fax:  Supports T.38 protocol for reliable fax transmission and pass-through fax for compatibility with various fax machines
  • Modem and POS Machine Support:  Compatible with modems and Point-of-Sale (POS) machines for efficient data communication
  • Legacy PBX and PSTN Network Integration:  Leverages over 10 years of experience to seamlessly integrate with a wide range of Legacy PBXs and service providers' PSTN networks

Streamlined Administration: Ease of Management

Empowering Connections, Enhancing Business
  • Intuitive Web Interface:  User-friendly web interface for easy configuration and management
  • SNMP Support:  Enables monitoring and management of the gateway using SNMP protocol
  • Automated Provisioning:  Streamlines the provisioning process with automated configuration deployment
  • Dinstar Cloud Management System:  Centralized cloud-based management system for convenient gateway administration
  • Configuration Backup & Restore:  Allows easy backup and restoration of gateway configurations for seamless maintenance
  • Advanced Debug Tools:  Provides advanced debugging capabilities for efficient troubleshooting and diagnostics

Features

Enhance Communication with Advanced Features
    • E1s/T1s with RJ48 Interface:  1/2 E1s/T1s connectivity with RJ48 interface
    • Dual Power Supplies:  Redundant power supply for enhanced reliability
    • 2 GE Ports:  Two Gigabit Ethernet ports for high-speed connectivity
    • SIP v2.0:  Support for SIP protocol version 2.0
    • SIP-T, RFC3372, RFC3204, RFC3398:  Support for SIP-T, RFC3372, RFC3204, and RFC3398 protocols
    • SIP Trunk Work Mode: Peer/Access:  Flexible SIP trunk work mode options: Peer or Access
    • SIP/IMS Registration: with up to 256 SIP Accounts:  Support for SIP/IMS registration with up to 256 SIP accounts
    • NAT: Dynamic NAT, Rport:  Support for dynamic NAT and Rport for NAT traversal
    • Flexible Route Methods: PSTN-PSTN, PSTN-IP, IP-PSTN:  Flexible routing options: PSTN-PSTN, PSTN-IP, and IP-PSTN
    • Intelligent Routing Rules:  Advanced routing rules for intelligent call routing
    • Call Routing based on Time:  Customizable call routing based on time schedules
    • Call Routing based on Caller/Called Prefixes:  Configurable call routing based on caller/called number prefixes
    • 256 Route Rules for each Direction:  Support for up to 256 route rules for each direction
    • Caller and Called Number Manipulation:  Ability to manipulate caller and called number information
    • Local/Transparent Ring Back Tone:  Local or transparent ring back tone configuration
    • Overlapping Dialing:  Support for overlapping dialing scenarios
    • Dialing Rules, with up to 2000:  Configurable dialing rules with support for up to 2000 entries
    • PSTN Group by E1 Port or E1 Timeslot:  Ability to group PSTN lines by E1 port or timeslot
    • IP Trunk Group Configuration:  Configuration of IP trunk groups for efficient management
    • Voice Codecs Group:  Ability to group voice codecs for efficient configuration
    • Caller and Called Number White Lists:  Configuration of white lists for caller and called numbers
    • Caller and Called Number Black Lists:  Configuration of black lists for caller and called numbers
    • Access Rule Lists:  Configuration of access rule lists for enhanced security
    • IP Trunk Priority:  Setting of IP trunk priorities for efficient call routing
    • Radius:  Support for Radius protocol for authentication and accounting
    • Codecs: G.711a/μ law, G.723.1, G.729A/B, iLBC 13k/15k, AMR:  Support for various voice codecs including G.711a/μ law, G.723.1, G.729A/B, iLBC 13k/15k, and AMR
    • Silence Suppression:  Enables suppression of silence during voice communication
    • Comfort Noise:  Provides comfort noise generation for enhanced voice quality
    • Voice Activity Detection:  Detects voice activity for efficient bandwidth utilization
    • Echo Cancellation (G.168), with up to 128ms:  Built-in echo cancellation with up to 128ms tail length
    • Adaptive Dynamic Buffer:  Adapts dynamic buffer size for optimal voice transmission
    • Voice, Fax Gain Control:  Controls voice and fax gain levels for optimal audio quality
    • FAX: T.38 and Pass-through:  Supports T.38 protocol for fax transmission and pass-through mode
    • Support Modem/POS:  Compatibility with modems and Point-of-Sale (POS) machines
    • DTMF Mode: RFC2833/SIP Info/In-band:  Support for DTMF signaling modes including RFC2833, SIP Info, and In-band
    • Clear Channel/Clear Mode:  Clear channel or clear mode configuration for voice communication
    • ISDN PRI, Q.sig:  Support for ISDN PRI and Q.sig signaling protocols
    • Signal 7/SS7: ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP:  Support for Signal 7/SS7 protocols including ITU-T, ANSI, ITU-CHINA, MTP1/MTP2/MTP3, TUP/ISUP
    • R2 MFC:  Support for R2 MFC signaling protocol
    • Web GUI Configuration:  Web-based graphical user interface for easy configuration
    • Data Backup/Restore:  Ability to backup and restore configuration data
    • PSTN Call Statistics:  Statistics on PSTN call usage and activity
    • SIP Trunk Call Statistics:  Statistics on SIP trunk call usage and activity
    • Firmware Upgrade via TFTP/Web:  Ability to upgrade firmware using TFTP or web interface
    • SNMP v1/v2/v3:  Support for SNMP versions 1, 2, and 3
    • Network Capture:  Capturing and analyzing network traffic for troubleshooting
    • Syslog: Debug, Info, Error, Warning, Notice:  Syslog support for logging debug, info, error, warning, and notice messages
    • Call History Records via Syslog:  Recording call history records via Syslog
    • NTP Synchronization:  Network Time Protocol (NTP) synchronization for accurate timekeeping
    • Centralized Management System:  Centralized system for efficient management of multiple gateways
Clearing Doubts, Enhancing Experiences

Frequently Asked Questions

Q: What is the number of E1s/T1s supported by the gateway?

A: The gateway supports 1/2 E1s/T1s with an RJ48 interface.

Q: Does the gateway have redundant power supplies?

A: Yes, the gateway is equipped with dual power supplies for enhanced reliability.

Q: How many SIP accounts can be registered with the gateway?

A: The gateway supports registration of up to 256 SIP accounts.

Q: What types of routing methods are available?

A: The gateway offers flexible routing methods, including PSTN-PSTN, PSTN-IP, and IP-PSTN.

Q: Can the gateway handle fax transmissions?

A: Yes, the gateway supports T.38 protocol for reliable fax transmissions as well as pass-through fax mode.

Q: Is it possible to integrate the gateway with legacy PBXs?

A: Yes, the gateway has over 10 years of experience integrating with a wide range of legacy PBXs.

Q: What voice codecs are supported by the gateway?

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

Q: Does the gateway support NAT traversal?

A: Yes, the gateway supports dynamic NAT and Rport for seamless NAT traversal.

Q: Can the gateway handle simultaneous calls?

A: Yes, the gateway can handle up to 60 simultaneous calls.

Q: Is SNMP supported for monitoring and management?

A: Yes, the gateway supports SNMP for efficient monitoring and management.

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