Continuous ringing even when answered/hangup

[raulbmjr]

I've done setting up the WEBRTC according to your instructions and successfully tested it by registering an extension using another server (FreePBX version 15). I can successfully do an inbound and outbound call to an outside number, but the problem is whenever I tried to call another extension on the same FreePBX server (or myself) the ringing does not stop even when it was successfully being answered or hangup.

I cannot see any issues in the FreePBX logs as it terminates the session successfully

The tone in the webrtc is not terminating even it was already canceled, answered or hangup

Appreciate your advice on this. Thank you

[InnovateAsterisk]

Please send me the console log.

I think this is the same issue that someone else reported. They send two progress 180 events, and this caused the media to play twice. So on answering a call, it would only stop the one.

The latest release fixes that.

[raulbmjr]

Thank you. I installed what it is on your site at https://github.com/InnovateAsterisk/Browser-Phone last week and I am not sure what is the latest release. Is it already out since I only saw one version on github. Can you provide me if the update is already available? Thank you

[InnovateAsterisk]

The update was made 3 days ago: https://github.com/InnovateAsterisk/Browser-Phone/commit/32c1cabced4a91b7300960bf24e95be060bba794

Try this version first.

[raulbmjr]

freepbx-logs.txt

Please see attached for the FreePBX Logs. Note that I changed the real IP addresses and extension numbers on the attached for security reasons. Thank you

[InnovateAsterisk]

Sorry, I meant the browsers console log. (Developers Tools)

[raulbmjr]

console.txt

Attached is the console log. I already replaced the phone.js with the bug fixed but still the same issue was encountered

Appreciate your help on this.

[InnovateAsterisk]

You have the same condition:

sip-0.20.0.min.js:2 Wed Jan 05 2022 05:38:58 GMT+0800 (Philippine Standard Time) | sip.Inviter | Inviter.onProgress phone.js:2179 Audio: media/Tone_EarlyMedia-US.mp3 sip-0.20.0.min.js:2 Wed Jan 05 2022 05:38:58 GMT+0800 (Philippine Standard Time) | sip.Inviter | Inviter.onProgress phone.js:2179 Audio: media/Tone_EarlyMedia-US.mp3 sip-0.20.0.min.js:2 Wed Jan 05 2022 05:39:07 GMT+0800 (Philippine Standard Time) | sip.Inviter | Inviter.onAccept

You server is sending 180 progress twice. The patch released a few days ago does fix this.

Please make sure you are clearing out your browser cache. It should work correctly. https://github.com/InnovateAsterisk/Browser-Phone/blob/df6bf122334bf701d18a22b677d25bab344de21b/Phone/phone.js#L2171

[raulbmjr]

Thank you. It worked! :)