We’ve been chasing an intermittent “no ringback tone” bug for a few days. Outbound calls go completely silent while ringing, but the moment the other end picks up, audio is perfectly normal. Staff say it’s happening on some calls but not others, and getting worse. FreePBX
17.0.28, current Sangoma Phone Desktop (4.2.0 on Windows).
First I ruled out the PBX side properly. Turned on pjsip set logger on and pulled traces against calls staff confirmed were silent. Signaling was clean every time: carrier sends 183 Session Progress with real early-media SDP within about a second of the INVITE, Asterisk relays
it to the extension immediately, no drops or delays. Checked this against three separate confirmed-silent calls. All clean.
That pointed me at RTP delivery instead, maybe an ICE/NAT issue since these are WebRTC-registered softphone extensions. So I turned on rtp set debug on and waited for the next incident. Got one: extension called a mobile number, staff confirmed silence, then normal audio once
connected. This time I could see the actual packets. Carrier’s early-media RTP arrived at Asterisk and got relayed to the extension’s real address continuously, 546 packets back to back, zero gaps, tagged “via ICE” confirming a valid candidate pair was already up. Ran the whole
length of the ring. Staff still heard nothing. Then it got answered and audio worked instantly, over what’s basically the same media path.
So I’ve got packet-level proof the audio leaves the server and reaches the workstation. Not signaling, not NAT, not ICE, not RTP loss. Whatever’s wrong is entirely inside the client, in the handoff between “ringing” and “answered” states.
Anyone on a current build seeing this too? Happy to share the raw debug capture if anyone from Sangoma wants it, it’s a pretty clean reproduction: audio proven delivered, client proven silent, then working the instant call state flips.