keywords: ip pbx voip gateway gsm gateway

OpenVox DGW-L301 Series E1/T1/PRI VoIP Gateway

A cost-effective version of 1 E1/T1/PRI gateway;

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

Supports up to 30 concurrent calls.

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  • Specifications

    OpenVox DGW-L301 E1/T1/PRI Gateway is an open-source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. This kind of gateway connects traditional telephone systems to IP networks and seamlessly integrates VoIP PBX with the PSTN. With a friendly GUI, users may easily set up their customized gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).

    The DGW-L301 E1/T1/PRI gateway supports 1 software-selectable E1/T1/PRI 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 IPPBX
    • 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 T1/E1 RJ45
    • 2 10/100 Mbps Ethernet ports
    • Maximum Power Consumption: 3W
    • Power supply specification: 12V/1A
    • Operation humidity: 10%~90% non-condensing
    • Operating temperature: 0℃~50℃
    • Storage temperature: -20℃~70℃
  • Features

    System Features

    • Available in 1 port E1/T1, energy efficiency concurrent processing, up to 30
    • 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 for custom dialplans
    • Firmware update by HTTP
    • Support call statistics
    • Support TR069
    • Support auto provision
    • Support channel status show dynamically 
    • Support backup/upload configuration file
    • Multiple detailed log output 
    • Support Chinese language
    • Automatically reboot 
    • Good compatibility, support Asterisk, Elastix, Freeswitch and Small and medium IPPBX platform 
    • Available for OEM
    • 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, This gateway registers with the endpoint
    • SIP accounts can be registered to multiple servers
    • Combine different SIP Trunks into group
    • SIP(RFC3261) compliance
    • DTMF: RFC2833, SIP INFO, INBAND
    • Support T.38 /Pass-through Fax

    Routing

    • 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
    • IPv4, UDP/TCP, DHCP, TFTP, SCP
    • HTTP/HTTPS/SSH
    • Support DDNS
    • Support ping & traceroute command on the web
    • Support network capture on the web
  • Demo

    Online Demo

    Username:admin

    Password:admin