Asterisk on Freerunner
nicola.mfb at gmail.com
Fri May 1 20:30:23 CEST 2009
2009/4/29 Al Johnson <openmoko at 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
> 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 (220.127.116.11) 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:
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!
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