.. ==================================================
.. FOR YOUR INFORMATION
.. --------------------------------------------------
.. -*- coding: utf-8 -*- with BOM.

.. include:: ../Includes.txt


.. _admin-manual:

Administrator Manual
====================

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

.. index::
        single: Installation

Installation
------------

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


.. index::
        single: Configuration


Configuration
-------------

The extension have 5 configuration parameters:

#. 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**
#. 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**
#. Display name:
	The name for the web client that will be displayed during call.
#. SIP password:
	Password for authentication to the SIP server.
#. SIP debug:
	Enable or disable SIP debug on the javascript console.


The following picture shows the example configration:

.. figure:: ../Images/AdministratorManual/BeSettings.png
	:width: 200px
	:alt: Extension Configuration

	Webrtc_phone extension configuration


.. index::
        single: Asterisk 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:

.. figure:: ../Images/AdministratorManual/AstSettings.png
	:width: 200px
	:alt: Freepbx Settings

	Enabling the webrtc support on asterisk.