keywords: ip pbx voip gateway gsm gateway

DTU-301 (Digital Trunk Unit)

Supports 1 software-selectable T1/E1/PRI interface;

Supports up to 30 concurrent calls.

Request a Quote
  • Specifications

    DTU-301 (Digital Trunk Unit) is an open-source asterisk-based VoIP Gateway Module solution for operators and call centers. It is a converged media gateway product, which could be used with OpenVox UCP Series products. This kind of gateway connects traditional telephone system to IP networks and integrates VoIP PBX with the PRI/SS7/R2 seamlessly. With friendly GUI, customers may easily setup their customized gateway. Also secondary development can be completed through API. 

    The DTU-301 supports 1 software-selectable T1/E1 interface and supports up to 30 concurrent calls.


    Target Applications

    • Connect legacy PBX systems to low-cost VoIP services
    • Connect legacy PBX systems to remote sites over private VoIP links
    • Connect IP PBX systems to legacy TDM services
    • Phased transition from legacy PBX to IP PBX
    • Connect virtualized systems to legacy TDM services
    • Transcoding by connecting systems using varying codecs
    • Lync connectivity to SIP or legacy TDM providers and SIP or Legacy PBX


    Technical Specifications

    • 1 E1/T1 RJ45
    • 2 10/100M Ethernet port (one on the front panel and one on the back panel)
    • Power Consumption: 3W Maximum.
    • Operation temperature: 0°C to 50°C
    • Storage temperature: -20°C to 70°C
    • Operation environment humidity: 10%-90% No condensation
  • Features

    System Features

    • Available in 1 port T1/E1, energy efficiency concurrent processing, up to 30 channels
    • Signalling: PRI/R2/SS7
    • Support up to 24 countries’ standard R2 signalling
    • Support new R2 variant
    • Simple and convenient configuration via Web GUI
    • Codecs support: G.711A, G.711U, G.729A, G.722, GSM
    • Support protocols:SIP、IAX、TCP、UDP、RTP、SSH、HTTP、HTTPS
    • Support NTP time synchronization and client time synchronization
    • Support SSH access for background management, Asterisk CLI command operation
    • Open API interface (AMI)
    • Support ports group management
    • Support custom dialplans
    • G.168 Echo Cancellation
    • Firmware update by HTTP
    • Support call statistics
    • Support auto provision
    • Support channel status show dynamically
    • Support backup/upload configuration file
    • Multiple detailed log output
    • Support Chinese language
    • Automatically reboot
    • Compatibility with all kinds of SIP servers, such as Asterisk, 3CX, Freeswitch and other IPPBX platforms
    • Available for OEM/ODM
    • 3-month "No Question Asked" Return Policy, and Two-year Warranty

    SIP Features

    • Support add, modify & delete SIP Accounts
    • SIP registration with Domain
    • Support multiple SIP registrations:Anonymous, Endpoint registers with this gateway and this gateway registers with the endpoint
    • SIP accounts can be registered to multiple servers
    • Combine different SIP Trunks into group
    • SIP(RFC3261) compliance
    • Support T.38 /Pass-through Fax


    • Flexible routing settings
    • Support 512 routing
    • Support caller/callee manipulation and filtering
    • Trunk group support, Trunk priority management
    • Support add, modify & delete routing
    • E1/T1 port grouping
    • Support Failover

    Network Features

    • Network type: Static IP and DHCP
    • Support DDNS
    • Support ping & traceroute command on the web
    • Support network capture on the web
  • Demo

    Online Demo