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@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?