Hello. I am using Freepbx 14.0.16 with WebRTC module 22.214.171.124
webRTC users can make and receive calls. The one issue is that when a call is placed there is no “ringing sound” before the call is answered. Just dead air. After sip debugging I found that sip event 183 session progress is being received by the PBX. I believe this is the event that would trigger ringing prior to call pickup (There is also a sip event 180 ringing but I think 183 is a more advanced message?) Anyways the sip event 183 is never passed to the webRTC client.
For the WebRTC client there is a sip event when the call is placed but no event from then on until the call is answered.
All normal sip phones connected to the PBX operate properly and on those sip devices you do hear ringing so I do believe otherwise all is good. Possibly a WebRTC issue? I can see no settings to tweak the WebRTC extensions. I did try progressinband for the outgoing sip trunk to the provider but this made no difference. I also tried forcing a ring sound for outbound calls on the sip trunk with r added to the dial string but no difference there either.
Here is the WebRTC sip config if that helps. Thanks for any assistance/ideas.
Thanks for the reply. I should have been more clear. I did a sip debug on the WebRTC peer and when the sip even 183 is received by the pbx its never forwarded to the WebRTC client. On the sip debug for the WebRTC peer you see the call request go out, I see the request go out the trunk, event 183 comes back up the trunk, its never sent to the webRTC client (can see that via the sip debug), then when the call is answered the sip handshake completes and RTP is setup. Things proceed as normal. DTMF works etc. To entertain your idea I did log into the UCP while on the same network as the Freepbx server with no change in behavior. If anyone out there does not have this issue can you share your webRTC module version and freepbx version? Any other ideas? Thanks!
Anyone else successfully have ringing on outgoing calls placed from WebRTC using Freepbx 14.0.16 with WebRTC module 126.96.36.199? Just trying to figure out of this is a bug since sip message 183 is not being forwarded to the WebRTC endpoint from the phone system as the call progresses. Sip 183 comes in the sip trunk from the carrier but debugging on the voip server it is never sent back out. Thanks.
I can confirm we do not have an outgoing ringing tone either for my users using UCP. Just dead silence until the other side picks up or voicemail answers. Even tested on the local lan as the FreePBX box same results. We are using
Wow so the behavior is even in 15.X. Thats a bit surprising as it seems like an obvious issue. Just makes me wonder if Im missing something but Ive researched the heck out of this and feel Ive hit a roadblock. I suppose at this point I could open a bug ticket but I heard from somewhere that the webRTC module is completely unsupported so not sure if its appropriate to do so. I may try anyways if I dont see feedback here. Thanks for reporting longqvo!