Showing posts with label FreePBX.. Show all posts
Showing posts with label FreePBX.. Show all posts

FreePBX WebRTC Phone Setup.

Posted: Thursday, March 6, 2014 by Unknown 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.

How To Solve The Problem Of The "Unavailable / Busy"message Recording ?!

Posted: Tuesday, May 14, 2013 by Unknown in Labels: , ,
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When using the *97 to access your voice-mail you can go to the advanced settings and record your "Unavailable / Busy" message but none of this messages will be played back to the caller when recording "Unavailable / Busy"message  using this way because *NOW has a bug that will prevent those messages from being played back to the caller if you are unavailable or busy see Ticket #1387 .you can solve this problem easily using the * feature code.

-What The * Feature Code Is Used For ?

1-To leave a voice mail message to any extension by directly dialing  the voice-mail of the user.for example if you want to leave a voice-mail message to extension 1001,you will dial * followed by the extension number like this *1001.

2-To access your voice-mail / the other users voice-mail by dialing the * followed by the extension number of the destination you want to access its voice-mail.for example if you want to access the voicemail of extention 1001 you will dial * followed by the extension number but you must interrupt the initial menu you will receive by pressing * and after a successful authentication you will log in and now you can record your "Unavailable / Busy"message. 

How To Solve The Problem Of The "ON-Demand" Recording?!

Posted: Monday, May 13, 2013 by Unknown in Labels: , ,
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On-Demand Call Recording Not Working?!Let's Fix It!:

By default,when enabling (Inbound / Outbound / Internal / External) call recording from the user extension configuration page , it will work without any problem but when enabling On-Demand recording it will fail so you need additional configuration steps. 

  1-Go To " FreePBX Advanced Settings "...Change The " Asterisk Dial Options ", " Asterisk Outbound Trunk Dial Options " As Follows :

Asterisk Dial Options : tTwWr
Asterisk Outbound Trunk Dial Options : TtwW

   -Setting the t allow the call recipient to transfer the call (*2 or ##).
   -Setting the T allow the call initiator to transfer the call (*2 or ##).
   -Setting the w allows the call recipient to toggle recording (*1).
   -Setting the W allows the call initiator to toggle call recording (*1).

2-Use " VI " To Edit The " features_featuremap_additional.conf " And  Add The " automon=*1 " :

vi /etc/asterisk/features_featuremap_additional.conf
automon=*1

3-Use " VI " To Edit The " features_general_custom.conf " And  Add The " featuredigittimeout=1500 " , " courtesytone=beep " :

vi  /etc/asterisk/features_general_custom.conf
featuredigittimeout=1500
courtesytone=beep

Note:You Must Reboot Your *NOW After Making The Changes To Take Effect.