<div dir="ltr">Answering my own question. Setting the following in alsa.conf fixed my problem:<br><br>input_device=plughw:0,0<br>output_device=null<br><br>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.<br>
<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, May 6, 2013 at 8:31 PM, Chris Gentle <span dir="ltr"><<a href="mailto:gentlec@gmail.com" target="_blank">gentlec@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions.<br>
<br>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.<br>
<br>To get the alsa input into the conference I configured the alsa module and did this at the console:<br><br>console dial 100@conferences<br><br>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.<br>
<br>Any ideas what I'm missing here? Is there a better way to do this?<span class="HOEnZb"><font color="#888888"><br>-- <br>Chris
</font></span></div>
</blockquote></div><br><br clear="all"><br>-- <br>Chris
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