I need some help with a few issues… thanks all in advance for you assistance
issue number 1:
I have a jitsi server and jigasi running on prem, they are registered to work with on prem pbx. all is working fine for out going sip calls from meetings and in coming calls to meetings.
the issue i’m facing is that jigasi needs to be restarted if a certain meeting tries to invite more than 6 to 10 out going phone destinations.
if i place the calls two or there numbers at a time it works fine.
but adding a list of 10 to 15 destinations - the system calls them all, they are all connected, but there is not audio even though tcpdump shows traffic for audio.
in this situation only jigasi service restart brings back audio for new calls.
ps. the simultaneous outbound call limit for the jigasi sip extension is set to unlimited in the pbx (freepbx).
issue number 2:
when having to add the numbers for out going call, I can’t just past comma seperated list of numbers? i have to do it one by one and then click the simbol with the mouse. is there a seperator char that would allow me to paste more than one number?
issue number 3:
is there a way after issue 2 has been resolved to have an option to paste a number=name, number=name - so multiple outbound calls would have a recognizable name to them?
issue number 4:
since i’m using an anonimous jitsi environment embeded into our IIS + authentication AD login page, i find that only the first user that gets the moderator has the option to call out sip destinations. the invite with sip calls doesn’t appear to any one that hasn’t been given moderator privileges. I don’t want to integrate with ldap/AD or use jitsi auth… is there a way to let all users have this option?
issue number 5:
is there a way to leave the system open to guest unauthenticated users (since IIS front end does this) where everyone can open rooms but still give a few users the option to login so they will have moderator privileges to any room the enter even if they were not the first.