HI, I have used the steps specified in the installation guide for installing Jitsi Meet & Jigasi. I would like to use Asterisk server as VoIP Server.
Set up is like this (Jitsi Meet + Jigasi) vc. myjitsitest. com <-> Asterisk Server sip. myjitsitest. com <-> VoIP Client (1001).
during installation of jigasi the sip account used was 1004@sip. myjitsitest. com
can some one provide the right sip. conf and extensions. conf to be used for this setup for the following scenarios
- Conference Room with Password
- Conference Room without password
Also associated right sip communicator conf
Thanks & Regards
Because of problems to getting dial-in work, I have been reading most of Jigasi stopped working after upgrading before, and now this thread. It is also adviced to start without advanced IVR stuff. For this, I’m testing with the siptest conference name, because it should be the default name if no SIPAddHeader is used.
Asterisk 13 runs on the same Debian 9 server as jitsi meet & jigasi.
dialplan specifies the extension for the jigasi client (in my case 555), and when jigasi is started, asterisk log shows that 555 is registered. Sip.conf excerpt:
when I dial in to jitsi, the conference does not answer. The reverse way, inviting from the conference, after hitting the “invite” button, an error appears: error while adding participants (translated from German text).
2020-04-01 17:52:27.982 SEVERE:  org.jitsi.meet.ComponentMain.log() host-unknown, host:localhost, port:5347
This latter log was just a spontanous idea for checking. How to localize the problem?
You are missing the component in the prosody config. But you better not use component, try with mucs. GitHub - jitsi/jigasi: Jigasi: a server-side application acting as a gateway to Jitsi Meet conferences. Currently allows regular SIP clients to join meetings and provides transcription capabilities.
Once you configure and you got jigasi running without errors, you will see a log in jicofo that a new jigasi instance had been added.
Thanks for your reply. I followed this install guide till (and not including) the section " Configuring SIP and Transcription". The stuff regarding MUC I considered relevant only for sophisticated use cases. I’ll take care of it tomorrow, and return here with the results.
BTW: If jigasi service is stopped, it does not unregister its sip client to asterisk. A certain time I was looking whether the same sip user was registered by another client as well or not…
I did manage to have the component active (a while ago - yet don’t remember how exaclty). At first, an authentication problem occured, which was cause by (once) reinstalling a module, and SIPPWD in jigasi/config did not match the entry in another config. I did continue this component route, because it’s the default debian way. The github manual you refer to is based on a self-built environment.
When I use the + icon (not telephone icon as described here), the invite is no longer rejected. But the 555 asterisk number as assigned to jigasi does not result in an effective call from the siptest room (no event can be seen in asterisk). Same behaviour the reverse way (calling from anywhere to 555 - does not answer). As already noted: jigasi registers to asterisk.
Some SEVERE errors from jigasi.log:
2020-04-07 20:44:21.913 SEVERE:  impl.configuration.ConfigurationActivator.log() Error creating c lib instance for fixing file permissions
java.nio.file.FileSystemException: /etc/jitsi/jigasi: Operation not permitted
2020-04-07 20:45:04.218 WARNING:  org.jitsi.xmpp.component.ComponentBase.log() PROCESSING TIME LIMIT EXCEEDED - it took 167ms to process:
2020-04-07 20:45:04.306 SEVERE:  impl.certificate.CertificateServiceImpl.verify().1064 Missing CertificateDialogService by default will not trust!
2020-04-07 20:45:04.307 INFO:  impl.certificate.CertificateServiceImpl.checkCertTrusted().832 Untrusted certificate