asterisk users - May 2013 - chan_alsa and confbridge

OK, somebody may have a much better way of doing what I'm attempting.  If
so, I'm open to suggestions.

I am trying to configure confbridge to create a "conference" room with
an
audio stream coming from my sound card.  The idea is for a group of people
to be able to call in and listen to someone giving a speech but not
necessarily interact.  I've got confbridge configured and it seems to work
when I connect via other SIP phones.

To get the alsa input into the conference I configured the alsa module and
did this at the console:

console dial 100 at conferences

This seems to work, once I got my alsamixer stuff set right.  However,
within about 10 seconds the audio goes bad.  Lots of distortion, echo,
etc.  So I recorded a snippet right out of the sound card and loaded it
into audacity.  The snippet was fine.  No distortion at all.  So the
problem seems to be something in asterisk.

Any ideas what I'm missing here?  Is there a better way to do this?
-- 
Chris
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Answering my own question.  Setting the following in alsa.conf fixed my
problem:

input_device=plughw:0,0
output_device=null

Changing the input device to plughw helped some but didn't completely clear
the audio up.  Setting the output device to null did the trick.  I'm
wondering if there was some kind of interrupt hammering going on here with
my particular hardware.  Even before the audio completely fell apart I
could hear some little "pops" that sounded like the interrupts were
not
being serviced fast enough.



On Mon, May 6, 2013 at 8:31 PM, Chris Gentle <gentlec at gmail.com> wrote:
> OK, somebody may have a much better way of doing what I'm attempting. 
If
> so, I'm open to suggestions.
>
> I am trying to configure confbridge to create a "conference" room
with an
> audio stream coming from my sound card.  The idea is for a group of people
> to be able to call in and listen to someone giving a speech but not
> necessarily interact.  I've got confbridge configured and it seems to
work
> when I connect via other SIP phones.
>
> To get the alsa input into the conference I configured the alsa module and
> did this at the console:
>
> console dial 100 at conferences
>
> This seems to work, once I got my alsamixer stuff set right.  However,
> within about 10 seconds the audio goes bad.  Lots of distortion, echo,
> etc.  So I recorded a snippet right out of the sound card and loaded it
> into audacity.  The snippet was fine.  No distortion at all.  So the
> problem seems to be something in asterisk.
>
> Any ideas what I'm missing here?  Is there a better way to do this?
> --
> Chris
>


-- 
Chris
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