Fusionpbx-WEBRTC

WebRTC Dialer for FusionPBX. A fully featured browser based WebRTC SIP phone for FusionPBX.

*NOTE this is a custom made WebRTC client for Fusionpbx and i have used Browser-Phone which is a repo of Mr. Conrad de Wet. I have just made changes on the Browser-Phone code and created few php files to make it work with FusionPBX.

Description

This web application for FusionPBX. When installed you won't have to enter the extension username / password or do any other configuration on FusionPBX for this to work. You just have to assign a user to the extension and then open the webphone. The application will automatically fetch your domain / username / password and callerID and it will register automatically.

Features

SIP Audio Calling
SIP Video Calling
XMPP Messaging
Call Transfer (Both Blind & Attended)
3rd Party Conference Call
Call Detail Records
Call Recording (Audio & Video)
Screen Share during Video Call
Scratchpad Share during Video Call
Video/Audio File Share during Video Call
SIP (text/plain) Messaging
SIP Message Accept Notification (not delivery)
Buddy (Contact) Management
Useful debug messages sent to console.
Works on: Chrome (all features work), Edge (same as Chrome), Opera (same as Chrome), Firefox (Most features work), Safari (Most feature work)
Asterisk SFU - Including talker notification and Caller ID
Dark Mode & Light Mode - System Setting Detects

XMPP Features v0.2.x

User Login & Auth (Use SIP credentials)
Buddy List (Roster) Saved on Server
Buddy vCard
Buddy Picture Upload
Message Typing Indication
Message Delivery & Read Notification
Offline Message History (If supported by server)
Tested to work with Openfire

Requirements.

You need to have a Valid SSL Certificate. You can get one from Certbot which will be free for 3 months, or you can get your own SSL certificate and install it on the FusionPBX server.
You need to have a valid Domain and you need to make sure the Domain's DNS A record points to the FusionPBX server.
Required Freeswitch Version 1.8+
Optional : Openfire XMPP Server.

Installation Guide.

cd /usr/src/
git clone https://github.com/amitiyer/Fusionpbx-WEBRTC.git
cd Fusionpbx-WEBRTC
cp -R Browser-Phone /var/www/fusionpbx/
cp -R core/phone  /var/www/fusionpbx/core/

Certificates.

As mentioned above, you can get a certbot certificate or have your own. Now you need to copy the certificates to freeswitch directory.

Copy your certificates like below.

cat fullchain.pem > /etc/freeswitch/tls/all.pem
cat privkey.pem >> /etc/freeswitch/tls/all.pem

Then restart Freeswitch.

service freeswitch restart

Configuring FusionPBX.

Login to your FusionPBX Admin go to Advance -> SIP Profiles and edit Internal, Make sure the value for ws-binding is :5066 and wss-binding is :7443 and both are Enabled.

Confirm if your freeswitch has =ws and =wss on your internal profile.

Restart freeswitch if necessary.

Create a user from Accounts -> Users. Create or edit the extension on which you want to enable WebRTC and assign the user on the extension.

Adding Phone on the Menu

Edit the default menu from Advance -> Menu Manager. Add a new menu. You can keep the title whatever you want. Link : /core/phone/index.php Target : Internal Icon : anything you want. Parent Menu Blank Groups : superadmin,admin,users,agents Protected : True Order : 101

Once done. log out and login with the user on which you have enabled the WebRTC and you will get a phone option on FusionPBX menu. Click on it and you will be able to make calls.

amitiyer / fusionpbx-webrtc Goto Github PK