WebRTC on FusionPBX

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digiwolf

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Dec 23, 2021
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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.
  • liberal-dtmf true true.
  • send-message-query-on-register true true.
  • send-presence-on-register true true.
  • wss-binding :7443 true.
3. Go to menu->status->sipstatus then:
  • click fluchcache.
  • click reloadxml.
4. You we NEED well working letsencrypt SSL certificates:

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 freeswitch

6. 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/freeswitch

7. 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

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Logged in

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Calling

1656459388125.png

Receving call

1656459446759.png
 
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