On Wednesday 29 April 2009, Nicola Mfb wrote:
> 2009/4/19 Nicola Mfb :
> > 2009/4/19 Al Johnson :
> > [...]
> > 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 st
2009/4/19 Nicola Mfb :
> 2009/4/19 Al Johnson :
> [...]
> 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 establi
2009/4/26 Rask Ingemann Lambertsen :
> On Sat, Apr 18, 2009 at 05:49:05PM +0200, Nicola Mfb wrote:
>
>> I will be happy to write an AMI gui but now I'm hold having problems
>> with the alsa channel. Using the pcm default is not compatible with
>> the default shipped /etc/asound.conf, so I just trie
On Sat, Apr 18, 2009 at 05:49:05PM +0200, Nicola Mfb wrote:
> I will be happy to write an AMI gui but now I'm hold having problems
> with the alsa channel. Using the pcm default is not compatible with
> the default shipped /etc/asound.conf, so I just tried to use
> plughw:dnsoop and plughw:dmix, t
2009/4/24 Timo Juhani Lindfors :
> Nicola Mfb writes:
>> But I'm happy, asterisk runs fine in a real case.
>
> Can you check if you get lower latency by only running linphone on fr
> and having the 3g stick connected to fr itself?
I cannot before next tuesday, but during the weekend I'll test FR
Nicola Mfb writes:
> But I'm happy, asterisk runs fine in a real case.
Can you check if you get lower latency by only running linphone on fr
and having the 3g stick connected to fr itself?
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> (I'm just thinking how many om guys got the same in the last two years! :)
>
LOL, just as many as distros and alsa states here :)
Great work
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2009/4/21 Nicola Mfb :
> 2009/4/19 Nicola Mfb :
[...]
> I'll update about my progress on AMI interface soon.
It's great night for me!
I was able to do my first VoIP->PSTN call with FR, it was to my
girlfriend of course, It may be for love or It may be to not bother
some other guy with an unpredict
thanks a lot :D
2009/4/22 Nicola Mfb
> 2009/4/21 kimaidou :
> > Hi
> > thanks for this feedback !
> > Could you please write a wiki page about this, if not already done ?
>
> I started a page at http://wiki.openmoko.org/wiki/Asterisk
> Everyone interested is invited to correct (english is not my
2009/4/21 kimaidou :
> Hi
> thanks for this feedback !
> Could you please write a wiki page about this, if not already done ?
I started a page at http://wiki.openmoko.org/wiki/Asterisk
Everyone interested is invited to correct (english is not my native
language) and collaborate, there is a lot to
Hi
thanks for this feedback !
Could you please write a wiki page about this, if not already done ?
Thanks again
Kimaidou
2009/4/21 Nicola Mfb
> 2009/4/19 Nicola Mfb :
> > Some alsa guru may take a look at the chan_alsa.c file of asterisk
> 1.4.17?
>
> Jaroslav Kysela of ALSA pointed me to the
2009/4/19 Nicola Mfb :
> Some alsa guru may take a look at the chan_alsa.c file of asterisk 1.4.17?
Jaroslav Kysela of ALSA pointed me to the problem (thanks), and
effectively asterisk code does not support dmix plugin in it's state,
I corrected it with a fast 2 line change workaround working only
2009/4/20 Esben Stien :
> Nicola Mfb writes:
>
>> But we may superseed on this actually until having a well working
>> asterisk on freerunner
>
> Rather definitely use freeswitch;).
Hi Esben,
Actually only a patch for asterisk let me use the voip line provided
by my adsl carrier (Alice/Telecom It
Nicola Mfb writes:
> But we may superseed on this actually until having a well working
> asterisk on freerunner
Rather definitely use freeswitch;).
--
Esben Stien is b...@e s a
http://www. s tn m
irc://irc. b - i . e/%23contact
2009/4/19 Nicola Mfb :
[...]
> Some alsa guru may take a look at the chan_alsa.c file of asterisk 1.4.17?
Here a little c snippet to show you easily the problem (that I have on
the desktop too). So it seems an alsa-lib bug/feature ?
#include
int main(int argc, char **argv) {
int err;
2009/4/19 Nicola Mfb :
> 2009/4/19 Al Johnson :
> [...]
> I have stuttered outgoing audio, so I think the problem is with alsa
> buffer/periods etc., the proposed asound.conf file should work as
> create longer buffer/periods both for input and output, but using it
> asterisk does not speak anymore
On Sunday 19 April 2009, Nicola Mfb wrote:
> 2009/4/19 Al Johnson :
> > For linphone I use Brian Code's asound.conf :
> >http://www.koolu.org/asound.conf
> > This uses dmix and dsnoop and gives stutter-free sound in both directions
> > with linphone. It does have echo since we can't use the
2009/4/19 Al Johnson :
[...]
>> Let's survive this interesting topic.
>> I will be happy to write an AMI gui but now I'm hold having problems
>> with the alsa channel. Using the pcm default is not compatible with
>> the default shipped /etc/asound.conf, so I just tried to use
>> plughw:dnsoop and p
On Saturday 18 April 2009, Nicola Mfb wrote:
> 2008/9/6 TL Mieszkowski :
> > I've had a lot of success running both twinkle and asterisk and I thought
> > I'd share my experiences.
> > Twinkle works well, but the gui is limiting on the touchscreen. I think
> > once configured properly
> > asterisk
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