Asterisk on Freerunner was: voip on Debian

[Al Johnson]

On Wednesday 29 April 2009, Nicola Mfb wrote:
> 2009/4/19 Nicola Mfb <nicola.mfb at gmail.com>:
> > 2009/4/19 Al Johnson <openmoko at mazikeen.demon.co.uk>:
> > [...]
> > 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.

Scenario switching ought to be transparent to apps, but that might not be true 
if there's a change in the 'DAI mode' setting. There's more on this in the 
wiki:
http://wiki.openmoko.org/wiki/Neo_1973_audio_subsystem
I don't have the state files too hand to see if this is being changed, but 
it's the only setting I can think of that might upset an app.

Can you reload chan_alsa after the state change? I don't remember how granular 
the asterisk reload options are, but it might be a quick'n'dirty workaround.