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:
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 . e
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
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
a voip session.
But we may superseed on this actually until having a well working
asterisk on freerunner, and I'll complete the AMI gui.
However if I use plughw:0,0 in asterisk alsa.conf I may hear the ring
in the earpiece (the only problem was that I had to change the speaker
alsa control from 0
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
On 17 May 2008, at 18:16, Doug Hawkins wrote:
...
There are a few hassles with the Nokia software that I'm looking
forward to making sure are clean with the OpenMoko system when I
get to start playing on one. One is that on some free (airport
community) WiFi systems, you have to open a
On Saturday 17 May 2008 23:42, Brandon Kruse wrote:
The freerunner images would be great,
Ok, I'll do that soon.
and its great that you can get
it to build with the latest toolchain stuff, etc. I might build a
Ahh well, no. Not quite. I can get the supporting libs to build just not the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Stroller wrote:
|
| On 17 May 2008, at 18:16, Doug Hawkins wrote:
| ...
| There are a few hassles with the Nokia software that I'm looking
forward to making sure are clean with the OpenMoko system when I get
to start playing on one. One is that on
I never said it was perfect, just another option. I have no affiliation to
either digium or trixbox, but I did try both ISOs when i was starting out
with asterisk. I don't use either any more - CentOS, asterisk, vi.
I didn't give AsteriskNOW much time because it wouldn't play easily with the
On May 18, 2008, at 9:09 AM, Al Johnson
[EMAIL PROTECTED] wrote:
I never said it was perfect, just another option. I have no
affiliation to
either digium or trixbox, but I did try both ISOs when i was
starting out
with asterisk. I don't use either any more - CentOS, asterisk, vi.
I
Powell [EMAIL PROTECTED] wrote:
On Saturday 17 May 2008 23:42, Brandon Kruse wrote:
The freerunner images would be great,
Ok, I'll do that soon.
and its great that you can get
it to build with the latest toolchain stuff, etc. I might build a
Ahh well, no. Not quite. I can get the
On May 18, 2008, at 3:45 AM, Stroller [EMAIL PROTECTED] wrote:
On 17 May 2008, at 18:16, Doug Hawkins wrote:
...
There are a few hassles with the Nokia software that I'm looking
forward to making sure are clean with the OpenMoko system when I
get to start playing on one. One is that on
On May 18, 2008, at 9:09 AM, Al Johnson
[EMAIL PROTECTED] wrote:
I never said it was perfect, just another option. I have no
affiliation to
either digium or trixbox, but I did try both ISOs when i was
starting out
with asterisk. I don't use either any more - CentOS, asterisk, vi.
I
On Sunday 18 May 2008 16:15, Brandon Kruse wrote:
I have to make a clean install tonight, so I will work on it ;) like I
said, my build environment was not standard.
:D good stuff
I had portaudio in its own ipkg, I hope someone can fix that :)
essentially thats what I ended up doing
On May 18, 2008, at 11:40 AM, Andy Powell [EMAIL PROTECTED] wrote:
On Sunday 18 May 2008 16:15, Brandon Kruse wrote:
I have to make a clean install tonight, so I will work on it ;)
like I
said, my build environment was not standard.
:D good stuff
Totally. Hopefully the end result will
On Sunday 18 May 2008, Brandon Kruse wrote:
On May 18, 2008, at 9:09 AM, Al Johnson
[EMAIL PROTECTED] wrote:
I never said it was perfect, just another option. I have no
affiliation to
either digium or trixbox, but I did try both ISOs when i was
starting out
with asterisk. I don't use
On Saturday 17 May 2008 01:06, Brandon Kruse wrote:
One more thing,
The Digium Asterisk-GUI was designed ALL clientside (It is ALL javascript).
That's not always a good thing.
Trixbox uses PHP/mysql/apache2, whereas the AsteriskGUI uses the builtin
Asterisk HTTP Server, and javascript files
On May 17, 2008, at 10:31 AM, Andy Powell [EMAIL PROTECTED] wrote:
On Saturday 17 May 2008 01:06, Brandon Kruse wrote:
One more thing,
The Digium Asterisk-GUI was designed ALL clientside (It is ALL
javascript).
That's not always a good thing.
I agree.
Compliance has been extremely
Hi Brandon,
This sounds great! I currently run two Nokia E-series phones (mine my
wife's) that are connected to both WiFi GSM doing exactly this
(incoming calls on both networks, outgoing preferred over one of several
VoIP connections). Also, I switch WiFi networks regularly throughout
On Sat, May 17, 2008 at 12:23 PM, Andy Powell [EMAIL PROTECTED] wrote:
On Saturday 17 May 2008 18:07, Brandon Kruse wrote:
That's not always a good thing.
I agree.
Compliance has been extremely difficult.
Amen!
Another ugh for compliance across browsers :(
Trixbox uses
On Saturday 17 May 2008 20:55, Brandon Kruse wrote:
snip
I think that it's a dialer function - however it would be nice if other
applications could tell the dialer how to dial. Since dbus seems to be
the interface that's going to be used it might be nice to have the option
there too.
On Saturday 17 May 2008 22:19, Andy Powell wrote:
libogg, portaudio and libspeex compiled ok, although I have to change
libtool to say where arm-angstrom-linux-gnueabi-ranlib was located.
iaxclient_moko however refuses to find the installed portaudio
Managed to sort that out, then needed to
On Saturday 17 May 2008 23:09, Andy Powell wrote:
On Saturday 17 May 2008 22:19, Andy Powell wrote:
libogg, portaudio and libspeex compiled ok, although I have to change
libtool to say where arm-angstrom-linux-gnueabi-ranlib was located.
iaxclient_moko however refuses to find the
On May 17, 2008, at 5:26 PM, Andy Powell [EMAIL PROTECTED] wrote:
On Saturday 17 May 2008 23:09, Andy Powell wrote:
On Saturday 17 May 2008 22:19, Andy Powell wrote:
libogg, portaudio and libspeex compiled ok, although I have to
change
libtool to say where arm-angstrom-linux-gnueabi-ranlib
On May 17, 2008, at 4:19 PM, Andy Powell [EMAIL PROTECTED] wrote:
On Saturday 17 May 2008 20:55, Brandon Kruse wrote:
snip
I think that it's a dialer function - however it would be nice if
other
applications could tell the dialer how to dial. Since dbus seems
to be
the interface that's
Another bootable ISO to look at its trixbox
http://www.trixbox.com/products/trixbox-ce/features
Both make setting up an asterisk server very easy. They also run reasonably in
a virtual machine. trixbox has a few more bells and whistles; whether this is
good or bad is a matter of opinion, as is
Heh,
Try to actually edit the config files and then use it :P
From experience, asteriskNOW is my favorite, and the first platform
I am going to get the client to work with automatically.
(I am going to add an 'openmoko' option in the AsteriskGUI)
AsteriskNOW Also has Digital / Analog card
One more thing,
The Digium Asterisk-GUI was designed ALL clientside (It is ALL javascript).
Trixbox uses PHP/mysql/apache2, whereas the AsteriskGUI uses the builtin
Asterisk HTTP Server, and javascript files (because we believe that there
should never be unneeded load on the box that your phone
Hey Guys,
Just want to keep the community updated.
I was one of the few developers who have received a freerunner (gta02) in
the mail a couple days back.
Since then I have been updating all my latest packages (http://bkruse.com,
and the mokoiax project page), will check in my code tonight.
The
As a future user, I'm glad to hear about progress in this area. It might get
me to actually set up an Asterisk server. :) Can we really get the
datastream small enough for GPRS?
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On Thu, May 15, 2008 at 1:59 PM, Brandon Kruse [EMAIL PROTECTED] wrote:
The goal of this project is to seamlessly tie into the openmoko dialer
application as a 'gateway', so that you could chose to dial out over GSM or
dial out over IAX2 (wifi, possibly GPRS).
If you would like to help in the
Thanks Brandon, we appreciate your contributions.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse
Sent: Thursday, May 15, 2008 2:00 PM
To: List for Openmoko community discussion
Subject: IAX2/Asterisk + Openmoko FreeRunner
Hey Guys,
Just want
Hi Brandon,
I'm an end-user for the most part.
Do you think you can get the freerunner to emulate a common ip phone?
I'm imagining using it with TalkSwitch or any general ip based system.
Thanks
On Thu, May 15, 2008 at 2:59 PM, Brandon Kruse [EMAIL PROTECTED] wrote:
Hey Guys,
Just want to keep
Hi Brandon,
Thanks for the update. This might be the excuse I'm looking for to set
up Asterisk at home.
Although I tested your phone before I sent it to you. Unless I made a
mistake, it worked then. What concerns me is that I do have one other
phone that doesn't seem to take sound in via
Yes,
The iaxclient library I am implementing it in supports very very low
bandwidth protocols.
I have made a call of GPRS before, the only thing is the latency, but it's
somewhat useable still.
I have worked on the GUI for Digium, so go here and install asterisk + the
asteriskGUI (AsteriskNOW
Great!
Basically, when the source can be built into an ipkg pretty easily.
I want someone to test the testcall application I ported to see about
audio quality, controls, etc.
-bk
On Thu, May 15, 2008 at 4:11 PM, Steven Kurylo [EMAIL PROTECTED]
wrote:
On Thu, May 15, 2008 at 1:59 PM, Brandon
, May 15, 2008 2:00 PM
*To:* List for Openmoko community discussion
*Subject:* IAX2/Asterisk + Openmoko FreeRunner
Hey Guys,
Just want to keep the community updated.
I was one of the few developers who have received a freerunner (gta02) in
the mail a couple days back.
Since then I have been
On Thu, May 15, 2008 at 5:06 PM, Vinc Duran [EMAIL PROTECTED] wrote:
Hi Brandon,
I'm an end-user for the most part.
Do you think you can get the freerunner to emulate a common ip phone?
I'm imagining using it with TalkSwitch or any general ip based system.
Thanks
On Thu, May 15, 2008 at
I will do Michael.
That could have been the problem with my 1973 prototype, so thank you for
keeping me updated.
I am hoping to have the packages rebuilt by Friday and start the overall
integration, at least giving
the end user a simple console application to start testing :)
I passed a call
Thanks Brian, That sounds very cool. I was wondering how that would
work. I'm very excited to see how it all works out.
V
On Thu, May 15, 2008 at 6:39 PM, Brandon Kruse [EMAIL PROTECTED] wrote:
On Thu, May 15, 2008 at 5:06 PM, Vinc Duran [EMAIL PROTECTED] wrote:
Hi Brandon,
I'm an end-user
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