[linux-audio-user] Re: GNU Audio Community Conference Room

[Dan Mills]

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.