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.
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.