Recently updated Jigasi (and JVB + Jitsi-meet), incoming calls no longer get Answered

I will check, but as I remember there were no changes in the sip bundle for years, which makes it strange :slight_smile: can you creat a tcpdump of such a call and send it to me, or better clear jigasi logs in var/log/jitsi restart jigasi, make a problem call and send to me all the files from it, there is a pcap file in there. You can send it to damencho At jitsi dot org.

So the 1.0-235 is ed8693b and 1.1-38-g8f3c241-1 is 8f3c241.

One thing you can try setting this property .SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true, does it change anything?

These are all the latest sip changes and there are no significant changes … which leaves me puzzled about this :frowning:

Hello @damencho

I am new to this but i installed the all things with latest version and facing the same issue with latest jigasi 1.1-38-g8f3c241-1. Incoming call come but it get failed.

I am using freeswitch for the calling part.

@voxter What is the best way to downgrade it because i have installed from package using apt-get install jigasi.


What is the failure? Can you upload logs? @vishalmpai

I did apt-get remove jigasi, then:

apt-get install jigasi=1.0-235

Thats all it took.

I will try to get an A/B comparison of logs and pcaps to share with you Damien. I’ll also try the config setting you mentioned. You should try that setting as well vishalmpai.

Certainly strange!

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@damencho here is the log of jigasi and sngrep of call. Jigasi_log.txt (6.1 KB)

As @voxter posted that invite is been resend by jigasi again back to server in place further processing.

Hello @voxter

Thanks for your help i downgraded the jigasi and it’s working fine now.

@voxter are you also using freeswitch.

@vishalmpai I suspect this does not even come to our sip bundle, as UserAgent is freeswitch it may indicate a problem in jsip we use. We haven’t seen this when we were using jitsi with freeswitch and we haven’t seen this with asterisk and many other providers we were using and are still using that sip provider.

Can you send a tcpdump and the /var/log/jitsi folder content after clearing and restarting jigasi, without that there is nothing I can do.

ok will provide you that. Shall upload here or send to you via email if yes please share the email id.

send it to damencho at jitsi dot org.


Did you find a solution to your problem? I wonder if this is not the same issue as the one I had reported here Jigasi stopped working after upgrading. Not using freeswitch but the call is coming from asterisk in my case.


Hi, I would like to add that this happens to me as well.

Incoming call with Asterisk 13.19 (chan_sip), Jitsi/Jigasi running on Ubuntu 18. Jitsi Meet and Jigasi installed from the repo.

Like the others, downgrading jigasi does not exhibit the problem. Let me know what other information you need to track the cause of the problem.

Same problem with me, using asterisk 16.3

Works fine with jigasi=1.0-235

Confirming this is reproducible for me with Asterisk on Debian buster as well. Downgrading to Jigasi 1.0 fixes the problem.

i confirm that with jigasi downgraded inbound sip call from Asterisk and from freeswitch are working again

Setting .SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true totally fixes this issue for me.

Same issue with asterisk 13.19.2, setting SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true doesn’t change anything.


Sorry for butting in like this here, but the OP states that this issue only affects incoming calls and that outgoing calls work fine. I have a recent version of jigasi, but my outgoing calls are not working according to this post: Jitsi-meet and jigasi SIP to Asterisk server . I also see that my incoming calls are not working even if I set org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME to my current room (I can see the “background icon ring” in the conference room, but the call isn’t established). .SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true does not seem to change anything in my case.

Hello all,

just to confirm that the current Jigasi 1.1-38-g8f3c241-1 available on repo (apt-get) is not answering incoming calls from B2BUA (Asterisk) to Jitsi/Jigasi.
Using SIP tools like sngrep I was able to see that Asterisk correctly sends the SIP Invites to Jitsi, but there is no ACK back from Jitsi to Asterisk, hence the call is not answered.

Downgrading to jigasi 1.0.235 solved the issue.

Environment used:
Jitsi running on Debian 4.19.37-5+deb10u1
Asterisk 13 running on CentOS7
Zoiper 5 Pro as softphone
MSF Edge Chromium as browser