On 8 Mar 2006, at 01:08, Mike Taht wrote:
In particular I wanted to make it easier to do a call in radio station
(see rivendell) and integrating the voice to mp3 function and music to
mp3 function struck me as asterisk with jack as a natural bridge...
If ladspa plugins could be run through asterisk or a jack compliant
sip phone you could give your outgoing voice calls a little bass boost
for that "voice of god" effect...
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.
<Snip>
I haven't had much spare time recently to work on these ideas, but
freeswitch seems to be a bit more hackable than asterisk has become,
so I've been looking at that...
So many potential programs, so little time....
I know that one.
I'm not even
sure how an Asterisk jack channel would function for RTP input to
Asterisk.
What would do the signalling?
Mentally to me, a jack port is a inband telephone connection, no real
signalling save perhaps silence suppression need be used... DTMF, etc,
generated in band...
That would be a pair of jack ports, and to be at all usable in a
radio context it needs
to support at least "ring indicator" and ideally call termination
detection.
Being able to set up (and terminate) calls would also be kind of nice.
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 something that has also been on my todo list for a while
(with exactly the same intended use).....
Regards, Dan.
