try re-installing by stopping opnfire, deleting the ofmeet folder/jar file, copy the new jar file and restart openfire
There is a change from Jitsi. This is the server-side focus user. If you are creating the chat rooms manually or re-using existing chat rooms, make sure that at least one of your users is a moderator otherwise, none of your users will be able to make telephone calls or record conferences
how do you make a moderator ??
Group Chat | Permissions on the admin web page
ok did nothing special except avoiding being on the default jitsi meet page (the blue one)
now the phone icon is back.
BUT sip audio is still crap when now with jitsi-meet + jigasi it’s fine.
Is there a chance you will switch on the SIP stack of sip communicator like the jitsi / jigais legacy product ?
cheers.
Sorry. If you want HQ audio SIP calls, then use JItsi Meet + Jigasi as external components with packet forwarding/relaying. I am not planning on using that implementation in Openfire Meetings. I need an audio mixer with G711 telephone quality audio.
Right - for sip calls, G711 is generally fairly clear and has low processor overhead and latency, unlike some HQ codecs. However, G.722 I believe is a very similar codec with the same bandwidth and low overhead, but with twice the audio frequency spectrum, and it is widely supported. I am only suggesting it if is easy to implement, otherwise its not a priority to my mind - most calls use G711.
claerly my point was not about HD audio. it was about just make the thing work properly with Asterisk.
By default Asterisk does not accept multiple SSRCs and I have to say that most of SIP system installed are on asterisk.
so if you want your wonderful app work with others it might be good to interop with Asterisk …
in what sense your sip stack is different from what Jitsi-meet + jigasi use when they use sip.communicators stuff ?
so if you want your wonderful app work with others it might be good to interop with Asterisk …
And after asterisk, I should do freeswitch, kamilio, 3cx, opensip, openser, freepbx, etc
I am only interested right now in SIP for telephony integration. I support a SIP registration service (your phone can register with openfire meetings), sip proxy and direct SIP invites. Asterisk is first and foremost a multiple protocol PBX and not a SIP server.
The underlying SIP stack for jigasi and ofmeet is NIST SIP 1.2. Jigasi is using a custom proxy that manages multiple clients which are a cut-down version of the sip client in the jitsi client that can handle multiple SSRCs. In ofmeet, I am using a SIP proxy/registrar using the kelpie implementation as my starting point.