I realise that Jigasi is not meant for video-calls, but I still think this category fits the most for this question…
I’m interested in setting up a system such that a SIP endpoint can dial a sip server, such as Asterisk, be converted to WebRTC and connect to a Jitsi Conference. I’ve looked into multiple solutions such as Asterisk, Freeswitch and Kamailio, but so far I’m finding the conversion a tad difficult. Does anybody have any experience or success with anything like this? Maybe any recommendations?
I know that Jibri has some video-enabled SIP functionality, but this seems undocumented atm and I have not had much luck with this functionality.
If we ever get it working I’ll be sure to try and help with getting some info in the docs.
Working on getting more of Jitsi translated to Danish as well.
First off, thanks for the script, very neat! It’s helping me to understand the setup process a bit further.
I had to comment out a few steps as they weren’t compatible with my test server setup, but the script did run all the way through and all components - I believe - are installed. I’ve configured pjsua.config to connect to my Asterisk server, but now I’m a bit lost.
To my understanding you have to use a Jibri API to call on the sipgw somehow, but I can’t find any documentation of this being executed anywhere? Also I’m quite confused as to what goes into the dial-plan-owner/member JSON files…
Any help would be greatly appreciated! My hope is that once I get it working, I can hopefully translate the process into an Ansible playbook and document the process.