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Administrator Manual

The click to dial plugin can be used after installing webrtc_phone extension. This extension is available in the TER.

Installation

Go to the extension manager, search webrtc_phone extension and then install it.

Configuration

The extension have 5 configuration parameters:

  1. Web socket server address:
    This is the address of the SIP server with websocket capability, the example for asterisk server is ws://sip.example.com:8088/ws
  2. SIP URI address:
    This address contains user SIP URI information for the webclient with format sip:sip_user@sip_server. The example for asterisk with freepbx GUI, for extension 1000 the URI address will looks like sip:991000@sip.example.com
  3. Display name:
    The name for the web client that will be displayed during call.
  4. SIP password:
    Password for authentication to the SIP server.
  5. SIP debug:
    Enable or disable SIP debug on the javascript console.

The following picture shows the example configration:

Extension Configuration

Webrtc_phone extension configuration

Asterisk(freepbx) configuration

With freepbx, webrtc can be enabled by changing the option in the WebRTC Phone section on the extension settings. Change the option Enable WebRTC User Control Panel Phone to yes as shown below:

Freepbx Settings

Enabling the webrtc support on asterisk.