Jigasi installation strange behaviour

After installing jigasi using the installation guide jigasi does not show up in jitsi-meet. After starting jigasi the sip-communicator.properties file vanishes. See attached screen copy file. The sip-communicator.properties file required changes because of incomplete login incredients for the sip account. The user is a number only while the Sip Server is free3.voipgateway.org. I am not sure if the incredients for the sip-communicator.properties are proper. However the connection works using a sip-phone. Please find attached a screen copy of the last attempt, The sip-communicator.properties and the jigasi.log. Thank you for your help in advance. Toni
jigasi.log (85.4 KB)
sip-communicator.txt (14.4 KB)
screencopy.txt (2.9 KB)

I think having the sip-communicator.properties.bak there break things.
And at the end there is no sip-communicator.properties files, probably due to the exceptions before.

Thank you for the hint! The sip-communicator.properties.bak was moved to my personal home directory and the sip-communicator.properties file now remains. However, jigasi still does not work. Enclosed the log file and the sip-communicator.properties file. May I kindly ask for another hint? Thank you in advance and regards Toni
sipcommunicatorproperty.txt (14.5 KB)
jvblog.txt (600.4 KB)

What exactly doesn’t work? What about jigasi logs?

Hello, No sign appears in jitsi-meet, and when calling one ring tone and immedeately engaged tone. In jigasi log the following severe reports occur:
2022-04-20 15:43:49.938 SCHWERWIEGEND: [13] org.jitsi.impl.neomedia.device.DeviceConfiguration.log() Failed to register custom Renderer org.jitsi.impl.neomedia.jmfext.media.renderer.audio.PulseAudioRenderer with JMF.
2022-04-20 15:46:20.416 SCHWERWIEGEND: [82] org.jitsi.jigasi.AbstractGateway.notifyCallEnded().121 [ctx=16504695793991775351971] Call resource not exists for session.

Find the log file and the jigasi status as hardcopy below.
Thank you!

JigsiStatus.txt (823 Bytes)
jigasilog.txt (90.4 KB)

2022-04-20 16:03:59.669 WARNUNG: [137] org.jitsi.jigasi.SipGatewaySession.run().1576 [ctx=16504706386621682210845] No JVB room name provided in INVITE header

You are not passing the room to join from the sip side in the sip header.

Good evening. Thank you for your comment. However, I have no idea, how to do invoke :frowning: Thank you in advance for your help. Regards Toni

BTW: I just installed jigasi and inserted the credentials. Is there any further advice for configuration?

You can take a look here how it is done with voximplant: Guide for setting up Jigasi with Voximplant
Not sure what you use for sip provider … check whether you can add IVR and pass custom headers there. You can achieve same thing and with asterisk, there are few posts with examples in the forum about it, you can front your sip provider with asterisk doing the IVR for you and the lookup between pins and conference names.

while you learn Asterisk (FIY there are also posts about integration with FreePBX I think, if you know nothing about the subject the learning curve is a bit less I believe) you should already be able to test integration - there is a default room that can be setup to accept the incoming calls lacking headers.