OpenVox E1 Gateway DGW-L1 - 1 port


Jangan biarkan HACKER  menyusup ke SIP TRUNK yang disediakan oleh Provider, dapat menyebabkan kerugian Rp 200 juta.

Amankan dengan OpenVox Converter SIP ke E1

OpenVox telah terkenal di dunia dengan alat VoIP kualitas Premium, harga terjangkau.  Dengan hardware setara DIGIUM.  Dibangun oleh Team Research berpengalaman dan memiliki pabrik besar

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Jangan biarkan HACKER  menyusup ke SIP TRUNK yang disediakan oleh Provider, dapat menyebabkan kerugian Rp 200 juta.

Amankan dengan OpenVox E1 gateway ( Converter SIP ke E1 ) .

Harga Termurah, dengan kualitas bagus.  Stabil dan jarang rusak.

Segera beli di Sophos technology

OpenVox DGW-L1 T1/E1 Gateway adalah  open source asterisk-based VoIP Gateway untuk  operators dan call centers. E1 Gateway ini menghubungkan  traditional telephone system ke  IP networks dan  mengintegrasikan  VoIP PBX  dengan ISDN . Dengan friendly GUI, users dengan mudah dapat  setup  customized E1 Gateway. secondary development dapat dilengkapi melalui  AMI (Asterisk Management Interface).

The DGW-L1 T1/E1 Gateway supports 1 software-selectable T1/E1/PRI interface dan 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 T1/E1 RJ-48

  • 2 10/100/1000Mbps Ethernet ports

  • 2 USB 2.0 ports 

  • Maximum power: 12W

  • Power supply specification: 12V/AC

  • Operation humidity: 5%~95% non-condensing

  • Operating temperature: 0℃~70℃

  • Storage temperature: -40℃~85℃

System Features

  • Available in 1 port T1/E1, 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.723.1, 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 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


  • 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

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