On Wednesday 29 April 2009, Nicola Mfb wrote:
2009/4/19 Nicola Mfb nicola@gmail.com:
2009/4/19 Al Johnson openm...@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
2009/4/19 Nicola Mfb nicola@gmail.com:
2009/4/19 Al Johnson openm...@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
2009/4/26 Rask Ingemann Lambertsen r...@sygehus.dk:
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
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, the
2009/4/24 Timo Juhani Lindfors timo.lindf...@iki.fi:
Nicola Mfb nicola@gmail.com 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,
Nicola Mfb nicola@gmail.com 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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2009/4/21 Nicola Mfb nicola@gmail.com:
2009/4/19 Nicola Mfb nicola@gmail.com:
[...]
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
snip
(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
snip
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thanks a lot :D
2009/4/22 Nicola Mfb nicola@gmail.com
2009/4/21 kimaidou kimai...@gmail.com:
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
2009/4/19 Nicola Mfb nicola@gmail.com:
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
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 nicola@gmail.com
2009/4/19 Nicola Mfb nicola@gmail.com:
Some alsa guru may take a look at the chan_alsa.c file of asterisk
1.4.17?
2009/4/21 kimaidou kimai...@gmail.com:
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,
Nicola Mfb nicola@gmail.com 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 .
2009/4/20 Esben Stien b...@esben-stien.name:
Nicola Mfb nicola@gmail.com 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
On Saturday 18 April 2009, Nicola Mfb wrote:
2008/9/6 TL Mieszkowski mieszkow...@gmail.com:
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
2009/4/19 Al Johnson openm...@mazikeen.demon.co.uk:
[...]
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
On Sunday 19 April 2009, Nicola Mfb wrote:
2009/4/19 Al Johnson openm...@mazikeen.demon.co.uk:
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
2009/4/19 Nicola Mfb nicola@gmail.com:
2009/4/19 Al Johnson openm...@mazikeen.demon.co.uk:
[...]
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,
2009/4/19 Nicola Mfb nicola@gmail.com:
[...]
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 alsa/asoundlib.h
int main(int argc,
2008/9/6 TL Mieszkowski mieszkow...@gmail.com:
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 will make an excellent voip backend
TL Mieszkowski wrote:
On Thu, Oct 2, 2008 at 8:44 AM, Davide Scaini
[EMAIL PROTECTED] wrote:
Ekiga? did you tryed that on [EMAIL PROTECTED]
so curious!
I haven't, but I see no reason why it wouldn't work. It doesn't do IAX
though, only SIP. And sip is problematic behind a NAT firewall.
TL Mieszkowski wrote:
1.) You need the alsa state for voip handset. Can be got here:
http://svn.openmoko.org/trunk//src/target/audio/om-gta02/
This goes in /usr/share/openmoko/scenarios/
load it with the command :
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
1.) You need the alsa state for voip handset. Can be got here:
http://svn.openmoko.org/trunk//src/target/audio/om-gta02/
This goes in /usr/share/openmoko/scenarios/
load it with the command :
alsactl -f
Alastair Johnson wrote:
Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
About this, using this alsa control file, can you get the caller voice
only in the earpiece?
If I use it in a Om2008 I get the voice both in the
Marco Trevisan (Treviño) wrote:
Alastair Johnson wrote:
Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
About this, using this alsa control file, can you get the caller voice
only in the earpiece?
If I use it in a
Alastair Johnson ha scritto:
Marco Trevisan (Treviño) wrote:
Alastair Johnson wrote:
Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
About this, using this alsa control file, can you get the caller voice
only in the
Marco Trevisan (Treviño) wrote:
Alastair Johnson ha scritto:
Marco Trevisan (Treviño) wrote:
Alastair Johnson wrote:
Marco Trevisan (Treviño) wrote:
TL Mieszkowski wrote:
alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore
About this, using this alsa control file, can you
Alastair Johnson ha scritto:
Marco Trevisan (Treviño) wrote:
Was it you who mentioned having patched linphone to switch alsa states ,
and to tweak the GUI to fit the screen better?
Yes I was, but my work isn't complete yet :P
Unfortunately I've to write my bits in too many places... :|
--
TL Mieszkowski [EMAIL PROTECTED] writes:
asterisk will make an excellent voip backend for the neo
Asterisk is dead. Long live freeswitch.
Didn't you get the memo?;)
--
Esben Stien is [EMAIL PROTECTED] s a
http://www. s tn m
irc://irc. b -
Ekiga? did you tryed that on [EMAIL PROTECTED]
so curious!
d
On Thu, Oct 2, 2008 at 2:56 PM, Esben Stien [EMAIL PROTECTED] wrote:
TL Mieszkowski [EMAIL PROTECTED] writes:
asterisk will make an excellent voip backend for the neo
Asterisk is dead. Long live freeswitch.
Didn't you get the
PM, Esben Stien [EMAIL PROTECTED] wrote:
Asterisk is dead. Long live freeswitch.
Didn't you get the memo?;)
I was under the impression that freeswitch was more geared to low
level operations, carrier level stuff.(?)
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http://n2.nabble.com/voip-on-Debian
.nabble.com/voip-on-Debian-tp842903p1132559.html
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Interesting. Good find and setup.
I am currently working on some iax2 related code for gta02 and future
phones as far as the voip stack.
I was running into some audio problems for awhile, but am starting to
get all those little bugs worked out.
stuff with asterisk it has
quite a bit of functionality.
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On Saturday 06 September 2008, TL Mieszkowski wrote:
There is the potential to do some really cool stuff with asterisk it
has quite a bit of functionality.
We should really write a channel driver for the Neo (wolfson codec GSM
modem daemon). We could then use asterisk for custom voicemail
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