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

From: Mike Taht <mike.taht@email-addr-hidden>
Date: Wed Mar 08 2006 - 03:08:14 EET

On 3/7/06, Lee A. Azzarello <lee@email-addr-hidden> wrote:
> Even though I have done it before, I still have the belief that installing
> Asterisk locally is overkill for a single person to make calls.

In wanting to bridge the world of Linux audio and asterisk telephony I
had desires far beyond
making a single phone call. I would argue, first, that a local setup
of "single user" asterisk could be made a lot easier (freeswitch is
veering in that direction) and the flexibility of having a pbx on your
laptop or wherever (local voicemail. call presence information.
Text-to-speech support. Etc)

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...

surround sound conferencing becomes feasible.

stuff like that. It doesn't make sense to me that these two worlds -
telephony and professional audio - should be seperated.

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...

> 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...

Mike Taht
PostCards From the Bleeding Edge
Received on Wed Mar 8 04:15:07 2006

This archive was generated by hypermail 2.1.8 : Wed Mar 08 2006 - 04:15:07 EET