Asterisk on Freerunner was: voip on Debian

Nicola Mfb nicola.mfb at
Wed Apr 29 01:38:40 CEST 2009

2009/4/19 Nicola Mfb <nicola.mfb at>:
> 2009/4/19 Al Johnson <openmoko at>:
> [...]
> As AMI emits all needed events I'll add fso support for the GUI to
> handle the switching automatically, while for a true voip fso


I added fso support to switch between stereoout when ringing and
voip-handset when the call is established but asterisk does not reacts
well on this and stop to capture audio.
It works well if I set the voip scenario before launching it and never
switches to stereoout.
Before digging again in the asterisk alsa code I'd like to know if the
scenario switching is transparent to alsa applications, or may brings
underrun/overrun or other problems that needs to be managed in a
stronger way.


More information about the community mailing list