FreePBX WebRTC Phone Setup.

Posted: Thursday, March 6, 2014 by Mahmoud Ramadan Ali in Labels:
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-WebRTC stands for “Web Real-Time Communications,” a technology focused on embedding real-time communications, such as voice, directly within web browsers.
 
-FreePBX system administrators can download the WebRTC module from within the FreePBX Module Admin, or check out this video that shows off some of the new features enabled by the WebRTC module.

Note:The WebRTC module
still in beta that's why its NOT available yet in the FreePBX modules repository but the module can be installed on systems running Asterisk 11.5 (or higher) and FreePBX 2.11.0.11. For browsers, only Chrome is supported.

Download the WebRTC module from the " GIT " repository and set the proper permissions. 
-Go to the "modules" directory and use the "git" command to install the WebRTC module and give the "asterisk" user & group the full control permission.
[root@localhost modules]#cd /var/www/html/admin/modules
[root@localhost modules]#git clone https://github.com/FreePBX/webrtc.git webrtc
Initialized empty Git repository in /var/www/html/admin/modules/webrtc/.git/
remote: Reusing existing pack: 309, done.
remote: Total 309 (delta 0), reused 0 (delta 0)
Receiving objects: 100% (309/309), 4.76 MiB | 67 KiB/s, done.
Resolving deltas: 100% (108/108), done.       
[root@localhost modules]#chown -R asterisk:asterisk webrtc 
  
Install the module from the FreePBX module admin.
-In the FreePBX module admin you should see the WebRTC module as "WebRTC Phone" available for installation.

Enable the "WebRTC" module in the "Extensions" page.
-Now when creating/editing Extensions, a new option is available to enable WebRTC.
-In FreePBX go to "Applications" then "Extensions" ensure at least one extension has WebRTC enabled and has a voicemail password.

Log into the ARI portal to use your WebRTC phone.
-when an extension logs into the ARI, there is a WebRTC phone option available in the left hand menu. 
-Use this extension to log into the ARI User Panel and click on the WebRTC phone option.In the softphone text display you should see the words "Registered with Sip Server".At this point you can dial like normal from this phone.