2009/4/29 Al Johnson <openm...@mazikeen.demon.co.uk>: > On Wednesday 29 April 2009, Nicola Mfb wrote: >> [...] > 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.
I restored the "take and hold voip state" behaviour in my dialer, and all worked perfectly but I got a weird issue, while a call is up launching alsamixer, playing with the "Speaker" control and quitting, stops audio capturing after few seconds. I tryied the new version of asterisk (1.4.24.1) too hoping some alsa code was fixed, but I got stuttered audio again and when the call is answered asterisk get a "resource temporary unavailable" error on the alsa channel and continues to ring, so I cannot hear the other peer, I need more time to investigate and go deeper in asterisk to understand channels setup, switch and so on, the next step will be to backport alsa code in 1.4.21 to 1.4.17 in little steps to know where it brokes. > 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. I'll investigate on this asap. I tested all that with WiFi and it works nice, but cannot go far from my AP for more than 7/10 meters, the delay over the voip/dsl router is very acceptable, and playing with voice/speaker capture volume reduces the echo in a manner that conversation is quite comfortable. A dirty coded dialer prototype is quite ready, some screenshots at: https://apps.sourceforge.net/mediawiki/noko/index.php?title=Image:Nokoami1.png https://apps.sourceforge.net/mediawiki/noko/index.php?title=Image:Nokoami2.png https://apps.sourceforge.net/mediawiki/noko/index.php?title=Image:Nokoami3.png I added request WiFi resource, occupy cpu resource, swtich to voip scenario, and asterisk daemon starting directly in the dialer, so actually I have an one-click ready voip phone, but there is a *lot* of works to do and few time, so help is appreciated! Regards Nicola _______________________________________________ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community