Dinstar MTG1000

MTG1000 series E1/T1 Digital VoIP Gateways with 1/2 ports E1/T1 is a compact and cost-effective trunk gateway designed to interconnect between PSTN and IP networks. With powerful hardware design, MTG1000 series has comprehensive PSTN access capabilities as well as SIP to SIP interworking features that enables the interconnection between all these elements.

MTG1000 series trunk gateway with high-efficient design and strong DSP processor ensures high-performance of the interconversion of PCM voice signal and IP packets, even when the gateways are fully loaded. MTG1000 is interoperable with mainstream VoIP platforms, and is compatible with PSTN network with digital trunk interfaces based on our years' experiences on ISDN PRI / SS7 / R2 MFC.

  • 1/2 ports E1/T1 in 1U chassis

  • Dual Power Supplies

  • Up to 60 simultaneous calls

  • Flexible routing

  • Multiple SIP trunks

  • Fully compatible with mainstream VoIP platforms

Technical Specifications

Overview & Key Capabilities

The MTG1000 (Low-Density E1/T1 Digital VoIP Gateway) is a compact, cost-effective trunk gateway designed for efficient interconnection between traditional PSTN and VoIP/IP networks. It comes in a 1U chassis and offers configurations with 1 or 2 E1/T1 ports, supporting high call-throughput and flexible routing.

With a robust DSP design, it ensures high-performance PCM-to-IP conversion, even under full load conditions.

Protocol Support & Routing Flexibility

  • PSTN Protocols: Supports ISDN PRI, ISDN SS7 with link redundancy, and R2 MFC.

  • Fax and Modem: Supports T.38 fax, pass-through fax, and modem/POS communication.

  • Call Routing Options: Offers versatile routing paths: PSTN-PSTN, PSTN-IP, and IP-PSTN.

Voice Codecs & Quality Enhancements

  • Supported Codecs: G.711 (A/μ law), G.723.1, G.729A/B, iLBC (13k/15k), and and AMR.

  • Voice Quality Features: Includes silence suppression, comfort noise generation (CNG), voice activity detection (VAD), jitter buffering, and echo cancellation (G.168) with up to 128 ms tail length.

SIP & Network Features

  • SIP Protocols: Supports SIP v2.0, SIP-T, and multiple related RFCs (e.g., RFC3372, RFC3204, RFC3398).

  • SIP Capacity: Up to 256 SIP accounts, as well as multiple SIP trunks.

  • NAT Traversal: Features Dynamic NAT and Rport for seamless SIP connectivity.

  • Routing Rules: Includes intelligent routing based on time schedules, caller/called number prefixes, priority trunking, overlapping dialing, and up to 256 rules per direction.

Management & Monitoring

  • Administration Tools: Equipped with a web-based GUI, SNMP v1/v2/v3 support, automated provisioning, and centralized cloud-based management.

  • Diagnostics & Logging: Supports data backup/restore, firmware upgrades via TFTP/Web, network capture, syslog (debug/info/warning/error), call history logging, and NTP synchronization.

Call Capacity & Hardware Design

  • Handles up to 60 simultaneous calls, similar to the MTG1000.

  • Built with dual power supplies (1+1 redundancy) for high availability and reliability.

  • Features two Gigabit Ethernet (GE) ports for network and management connectivity.

Expanded Variants & Specs (Optional)

Certain sources, especially from manufacturer listings, indicate expanded configurations for MTG1000B models:

  • Port Options: Some variants support 1, 2, or even up to 4 E1/T1 ports, with up to 120 concurrent calls.

  • Security & Encryption: Includes TLS/SRTP, RTP/signaling encryption (e.g., VOS RC4).

  • PBX Compatibility: Works with Avaya, NEC, Alcatel PBXs and Huawei, Cisco, ZTE soft-switches

  • Interface & Dimensions:

    • RJ48 E1/T1 ports

    • GE0/GE1 Ethernet

    • RS232 serial port (115200 bps)

    • Power: Input 100–240 VAC, 50–60 Hz

    • Power usage: ~15 W

    • Dimensions: 436 × 300 × 44.5 mm (1U), Weight ~2.0 kg

Feature Comparison Snapshot

Category Key                          Highlights

Hardware                                                             1–2 E1/T1 ports (RJ48), dual power supplies, 2×GE Ethernet

Call Capacity                                                      Up to 60 simultaneous calls (some models: up to 120)

Protocols Supported                                       ISDN PRI, SS7 (redundant), R2 MFC

Voice & Fax Features                                       Voice codecs, echo cancellation, fax/modem support

SIP & Routing                                                      SIP v2.0, SIP-T, up to 256 accounts, flexible routing

Management Tools                                          Web GUI, SNMP, cloud management, syslog, firmware/backup

Advanced Options                           TLS/SRTP, encryption, broad port scalability, PBX/soft-switch compatibility