In this tutorial we are going to enable WebRTC on FusionPBX to use with an external webphone, in my case i use Saraphone.
1. First step login on your FusionPBX server and go to Menu->Advanced->Sip Profiles.
2. Click on "internal", then modify this.
5. Then restart FreeSWITCH:
6. To check if its working we run the following code on the server terminal:
7. On a different Webserver install Saraphone.
8. Then edit saraphone.html to preset WSS proxy address and port, and the SIP domain.
Initial screen
Logged in
Calling
Receving call
1. First step login on your FusionPBX server and go to Menu->Advanced->Sip Profiles.
2. Click on "internal", then modify this.
- liberal-dtmf true true.
- send-message-query-on-register true true.
- send-presence-on-register true true.
- wss-binding :7443 true.
- click fluchcache.
- click reloadxml.
cd /usr/src/fusionpbx-install.sh/debian/resources/
./letsencrypt.sh
Flollow the steps, on finish
cat /etc/dehydrated/certs/your-domain/fullchain.pem /etc/dehydrated/certs/your-domain/privkey.pem > /etc/freeswitch/tls/wss.pem
5. Then restart FreeSWITCH:
systemctl restart freeswitch6. To check if its working we run the following code on the server terminal:
netstat -pln |grep 7443 and you may see an output like this:tcp 0 0 yourip:7443 0.0.0.0:* LISTEN 129975/freeswitch7. On a different Webserver install Saraphone.
- As root go into HTML directory of your webserver, and then:
git clone https://github.com/gmaruzz/saraphone.git
chown -R www-data:www-data saraphone
8. Then edit saraphone.html to preset WSS proxy address and port, and the SIP domain.
- Search for tag id="domain" and replace the tag value="yourdomain" with your tenant domain.
- Search for tag id="proxy" and replace the tag value="yourdomain" with your server domain.
Initial screen
Logged in
Calling
Receving call
