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Digital Gateways

  • folder icon closed folder iconUser Manual
    • DGW-100XR User Manual
    • DGW-L1 User Manual
    • DGW-L20X User Manual
    • DGW-L301 User Manual
  • folder icon closed folder iconFunctions Guide
    • SNMP
    • How to configure PRI signalling
    • How to configure MFC/R2 signalling
    • How to configure SS7 signalling
    • How to use Auto Provisioning
    • How to use DID function with OpenVox T1/E1 Gateway
  • folder icon closed folder iconApplication Notes
    • OpenVox TR069 Technote
    • Digital Gateways Connect with Elastix
    • connect E1 gateway with elastix via sip trunk
  • folder icon closed folder iconFAQ
    • SNMP Guide on Digital gateway
    • Digital Gateway connect with 3CX
    • Digital Gateway connect with Issabel
    • Digital Gateway connect with Asterisk16.15.1
    • Digital Gateway connect with Openvox UC
    • Digital Gateway connect with Freepbx 15
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Updated on April 12, 2022

Digital Gateways

  • folder icon closed folder iconUser Manual
    • DGW-100XR User Manual
    • DGW-L1 User Manual
    • DGW-L20X User Manual
    • DGW-L301 User Manual
  • folder icon closed folder iconFunctions Guide
    • SNMP
    • How to configure PRI signalling
    • How to configure MFC/R2 signalling
    • How to configure SS7 signalling
    • How to use Auto Provisioning
    • How to use DID function with OpenVox T1/E1 Gateway
  • folder icon closed folder iconApplication Notes
    • OpenVox TR069 Technote
    • Digital Gateways Connect with Elastix
    • connect E1 gateway with elastix via sip trunk
  • folder icon closed folder iconFAQ
    • SNMP Guide on Digital gateway
    • Digital Gateway connect with 3CX
    • Digital Gateway connect with Issabel
    • Digital Gateway connect with Asterisk16.15.1
    • Digital Gateway connect with Openvox UC
    • Digital Gateway connect with Freepbx 15

Digital Gateways Connect with Elastix

Estimated reading: 2 minutes 700 views

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This document applies to OpenVox DGW Gateway .

OpenVox T1/E1 Gateway is an open source asterisk-based VoIP Gateway solution for operators and call centers. It is a converged media gateway product. It connects traditional telephone system to IP networks and integrates VoIP PBX with the PSTN seamlessly.

Default IP: 172.16.100.1 

                              User Name: admin
                              Password: admin

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If your system topology like the figure described, please enter the gateway default IP address In your browser to login web, and click “NETWORK—>LAN Settings” to set network parameters such as IP.

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Save your changes. Please type in your DNS server in “DNS Server Address”.

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Please select “VOIP—>VoIP Endpoints—>Add New SIP Endpoint” to set SIP trunk. The following figure shows detail information about how to set it.

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About other parameters in SIP, please set by your requirements for there is no need to set them in simple calls.

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Please login your Elastix® server to create a SIP trunk(1001) .In Elastix® server web, please choose “PBX—>Trunks—>Add SIP Trunk” to set like that:

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Click “ROUTING—> Call Routing Rules—> New Call Routing Rule” to set outbound and inbound routing rules like the following:

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Save the inbound call routing rules, please set the outbound rules as introduced. In order to make all calls successfully, please enable and set failover function in advanced routing rule like that:

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Outbound Rules

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Inbound Rules

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Taking advantage of SIP software such as Xlite, eyeBeam to register a SIP extension(301). After all above steps, you can try to make calls and send SMS.

 

 

 

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