Re: [ANN] Tweakable finger-friendly Xmonad + xvkbd + dzen2 ready-to-use config
Hey all, Please find the README at https://gitorious.org/xmonad-smartphone-config/xmonad-smartphone-config/blobs/raw/master/README.org Definitely going to try this out as well. Was working my way to something similar actually. Great job! ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: tizen released
it seems that the first samsung device with such so will be similar to galaxy III under the point of view of the hardware... On Mon, 18 Feb 2013 18:51:34 +0100, Denis 'GNUtoo' Carikli gnu...@no-log.org wrote: On Mon, 18 Feb 2013 23:44:58 +0800 Paul Wise pa...@bonedaddy.net wrote: git repos here: https://review.tizen.org/git/ was it ported on some real devices beside the emulator? Denis. -- 不要催我!你曾經問過梵谷畫很快嗎? ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Debian on Freerunner+SIP
On 02/18/2013 07:47 AM, David Matthews wrote: I tried using SIP (Linphone) on Debian without any luck. After removing echo cancellation in Linphone settings I can hear other side but the I'd be interested in any success reports with linphone or any other sip client on a current freerunner distro Anyone? I'm using Linphone on QtMoko with QX. I used this [1] as a guide to get things working. The author has some broken links to his configuration files, though. Here are the corrected links: http://pub.acaia.ca/profiles.conf http://pub.acaia.ca/favourites.conf http://pub.acaia.ca/.linphonerc I did find that it is necessary to restore the gsmhandset state scenario after exiting Linphone. Otherwise the phone hangs when suspending. alsactl -f /opt/qtmoko/etc/alsa-scenarios/gsmhandset.state restore I've been making calls over an OpenVPN tunnel on WiFi back to my asterisk server. The call quality is very nice with only an occasional audio stutter. It is definitely usable. Regards, Aaron [1] http://acaia.ca/~tiago/posts/VoIP_in_Neo_Freerunner_with_Qtmoko_and_Linphone/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Debian on Freerunner+SIP
Hi Aaron I'm using Linphone on QtMoko with QX. I used this [1] as a guide to get things working. The author has some broken links to his configuration files, though. Here are the corrected links: http://pub.acaia.ca/profiles.conf http://pub.acaia.ca/favourites.conf http://pub.acaia.ca/.linphonerc Thanks for those links - that is very encouraging - I'll be giving this a try Regards -- David Matthews m...@dmatthews.org ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Debian on Freerunner+SIP
Hello Aaron, Can you please provide more information, e.g. 1. Which kernel are you using? 2. Can you use arecord to record the microphone, i.e. can you hear a recording of yourself when running the following commands (replace path-to-voip-handset.state by the correct path)? alsactl -f path-to-voip-handset.state restore arecord -D hw:0,0 -r 8000 -f S16_LE -c 2 record.wav aplay record.wav Thanks for your help, Alon. On Tue, 2013-02-19 at 13:35 -0500, Aaron Sells wrote: I'm using Linphone on QtMoko with QX. I used this [1] as a guide to get things working. The author has some broken links to his configuration files, though. Here are the corrected links: http://pub.acaia.ca/profiles.conf http://pub.acaia.ca/favourites.conf http://pub.acaia.ca/.linphonerc I did find that it is necessary to restore the gsmhandset state scenario after exiting Linphone. Otherwise the phone hangs when suspending. alsactl -f /opt/qtmoko/etc/alsa-scenarios/gsmhandset.state restore I've been making calls over an OpenVPN tunnel on WiFi back to my asterisk server. The call quality is very nice with only an occasional audio stutter. It is definitely usable. Regards, Aaron [1] http://acaia.ca/~tiago/posts/VoIP_in_Neo_Freerunner_with_Qtmoko_and_Linphone/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Debian on Freerunner+SIP
On 02/19/2013 04:20 PM, alonivtsan wrote: Hello Aaron, Can you please provide more information, e.g. 1. Which kernel are you using? root@neo:~# uname -a Linux neo 2.6.34-qtmoko-v48 #1 Wed Sep 12 11:31:51 UTC 2012 armv4tl GNU/Linux 2. Can you use arecord to record the microphone, i.e. can you hear a recording of yourself when running the following commands (replace path-to-voip-handset.state by the correct path)? alsactl -f path-to-voip-handset.state restore arecord -D hw:0,0 -r 8000 -f S16_LE -c 2 record.wav aplay record.wav No, that doesn't seem to work. This works though: alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore arecord -r 8000 -f S16_LE -c 2 record.wav aplay record.wav Regards, Aaron ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
QtMoko audio quality
Hi all,
just would like to share my 'patching' of gsmhandset.state file. The
sound from speaker was too weak for making calls on the street or
shopping center. Therefore I had to change the values of control.4 to 127.
After this it much better to hear the remote side.
BR,
--
Peter Viskup
root@neo:~# diff -u /opt/qtmoko/etc/alsa-scenarios/gsmhandset.state.orig
/opt/qtmoko/etc/alsa-scenarios/gsmhandset.state
--- /opt/qtmoko/etc/alsa-scenarios/gsmhandset.state.orig
2013-02-19 22:46:46.0 +0100
+++ /opt/qtmoko/etc/alsa-scenarios/gsmhandset.state 2013-02-19
23:09:27.0 +0100
@@ -36,8 +36,8 @@
comment.range '0 - 127'
iface MIXER
name 'Speaker Playback Volume'
- value.0 115
- value.1 115
+ value.0 127
+ value.1 127
}
control.5 {
comment.access 'read write'
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Re: qtmoko calendar
Hi Robin,
Le lundi 11 février 2013 à 11:44 +, robin a écrit :
Hi Adrien,
thanks for this very handy tool. It works almost perfectly with my ics
calender
I export from orgmode. there is just one minor thing: it fails if I have a
date
on -07-31 which as far as I checked my calendar does actually exist:
for the error I found the following bug [1] on the internet with the solution
to
set the database datestyle like:
ALTER DATABASE bugzilla SET datestyle = 'iso, dmy';
is this also possible with the qtopia database?
I don't know exactly what does alter database sql request and I'd
prefer to not touch this database because it contains other informations
(like call history).
this is the output I get
Prepare the appointment recid=248
description=Bruno Brezinas %d. Geburtstag
startDate=1945-07-31T00:00:00
startDateTimeZone=
The 'day' parameter (0) to DateTime::set did not pass the 'an integer which
is a possible valid day of month' callback
at /usr/lib/perl5/DateTime.pm line 1875
DateTime::set(undef, 'day', 0) called at /usr/lib/perl5/DateTime.pm
line 1886
DateTime::set_day('DateTime=HASH(0x8f2618)', 0) called at
./ics2qtcal.pl line 107
main::extractDateFromIcalLine('ARRAY(0xa00790)', 1) called at
./ics2qtcal.pl line 260
Ok, I've tried to work on extractDateFromIcalLine function to remove
some assumptions made by the original author.
I'm not a perl expert, as you can see in my commits (I've made a lot of
commits because I use git to transfer my modifications on my phone), but
it worked better with my calendars.
Can you download latest version and try again with this calendar ?
https://github.com/Trim/qtmoko-ics2qtcal/raw/master/downloads/ics2qtcal-0.7.2.zip
Thanks,
Adrien
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Re: Debian on Freerunner+SIP
Thank you Aaron for your help. I simply copied the voip-handset.state file from qtmoko v48 and now everything works. To fix the quality of my output getting worse over time I enabled only GSM codec and disabled Enable adaptive rate control. On Tue, 2013-02-19 at 16:38 -0500, Aaron Sells wrote: On 02/19/2013 04:20 PM, alonivtsan wrote: Hello Aaron, Can you please provide more information, e.g. 1. Which kernel are you using? root@neo:~# uname -a Linux neo 2.6.34-qtmoko-v48 #1 Wed Sep 12 11:31:51 UTC 2012 armv4tl GNU/Linux 2. Can you use arecord to record the microphone, i.e. can you hear a recording of yourself when running the following commands (replace path-to-voip-handset.state by the correct path)? alsactl -f path-to-voip-handset.state restore arecord -D hw:0,0 -r 8000 -f S16_LE -c 2 record.wav aplay record.wav No, that doesn't seem to work. This works though: alsactl -f /usr/share/openmoko/scenarios/voip-handset.state restore arecord -r 8000 -f S16_LE -c 2 record.wav aplay record.wav Regards, Aaron ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
