Hello
I am trying to use the webrtc phone in the UCP. I’ve read quite a number of posts & wiki articles, so far no luck. I am NOT running a commercial licence, I run a Freepbx 16.0.26, all modules uptodate,
(UCP Phone - PBX GUI - Documentation for instance)
=> I’ve created a letsencrypt certificate, and connect with to Freepbx admin & UCP, certificate is ticked as default
=> extension is a regular pjsip ext, with: enable apvf, enable ice, ename rtp mux to yes, transport set to 0.0.0.0-wss (this transport has been configure), dtls=yes, certifcate = my certificate,
=> Settings / SIP Settings / General: webrtc/ stun server set,
=> Settings / SIP Settings / chan_pjsip: TSL Settings - certificate manager: mycert, ssl tlsv1_2, transport wss, ws, yes
=> Admin / User mgt: UCP / enable phone: Yes
In the dashboard, the UCP daemon shows as not running, not sure if this is an issue
I’ve even rebooted the server !
But when I log into UCP, I can all widget BUT the phone - does not show !
ok - I’ve made some progress… sadly… I have issues with color and did not realize there were two tabs when adding widgets, and did not see the side bar wisgets - the phone is now added. It is blinking with a message connecting to socket… will search for that now
Are you saying that the web GUI breaches the Web Content Accessibility Guidelines at the fundamental level A? That could make it illegal in some countries, or in some contexts in some countries.
Not quite - I do have issues with colors (slightly colorblind), but I can see when adding a widget that there are two “categories” / tabs, deskbar widget and side bar widget, but for some reason, it took me a while to realize that they were indeed categories, and that I could click on site bar widget tab - to me (and I may be the only one !), that wasnt obvious, but given that this designs exists for a while, that is probably just me !
And last… as there are not so many threads talking about webrtc configuration, and that I mentionned no audio… My current setup is Freepbx server hosted on GoogleCloud, ie, no direct public IP, but CGNAT, and the remote UCP is also natted, I had to configure a STUN & TURN server to have audio (in the WebRTC Settings sections of asterisk sip settings)
