Dan Mills <dmills@xxxxxxxxxxxxxxxxxxxxx> writes:
> A sip <-> jack "hybrid" would be way cool, but while that covers the
> audio side of the problem, it leaves the call setup and control
> side.
The way this now with freeswitch is that you send messages to the
server. No, not using OSC, though I have mentioned it to them;). They
are using a simple text protocol over Jabber. With this you can
transfer your call to a conference room, then call other participants
and raise and lower their volumes, f.ex, just by sending these
messages. (bind them to a midi controller)
> Perhaps a daemon that could connect multiple sip "lines" to jackd
> and provided a couple of fifos to communicate line status and to
> handle dialing?
This is not really necessary as you really only need one bus to send
and one to receive, using ardour;) covering the mic and different
feeds with a midi controller (like feed0, feed1) if you do all
call management on the server, by sending these messages.
--
Esben Stien is b0ef@e s a
http://www. s t n m
irc://irc. b - i . e/%23contact
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