Re: Qtopia and VOIP
Al Johnson wrote: On Tuesday 16 September 2008, Nicola Mfb wrote: [EMAIL PROTECTED] wrote: What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) I've used the CLI version of linphone, but the GUI should be small enough to fit in 480x640 too. Well, the 2.1.1 version of linphone works quite well (after editing a little the code) [1], however the problem is another: we miss the alsa states needed to use the phone speaker as default output device and the microphone as a capture device. This night I've played a lot with this software but I wasn't able to use it as a standard phone... :| [1] http://3v1n0.tuxfamily.org/openmoko/linphone-VoIP-SIP-call.png -- Treviño's World - Life and Linux http://www.3v1n0.net/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
On Wednesday 17 September 2008, Marco Trevisan (Treviño) wrote: Al Johnson wrote: On Tuesday 16 September 2008, Nicola Mfb wrote: [EMAIL PROTECTED] wrote: What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) I've used the CLI version of linphone, but the GUI should be small enough to fit in 480x640 too. Well, the 2.1.1 version of linphone works quite well (after editing a little the code) [1], That's great, partly for reasons I'll get to below. I'll scratch this from my todo list then :-) What changes did you need to make? And do youhave a bitbake recipe in OpenEmbedded yet? however the problem is another: we miss the alsa states needed to use the phone speaker as default output device and the microphone as a capture device. This night I've played a lot with this software but I wasn't able to use it as a standard phone... :| The alsa state was relatively simple to set up - so much so that I don't think I saved it. There has been at least one state file for voip posted to the list though, and I think there is one in FSO milestone 3. The bit that caused problems was the audio interface. I was using 2007.2 so I killed pulseaudio to start with. The default alsa interface uses dmix, and linphone complained that this didn't allow a duplex connection. I could hear things on the Neo, but the other end couldn't hear me. I changed the .linphonerc to use OSS for the mic instead of alsa: [sound] playback_dev_id=ALSA: default device ringer_dev_id=ALSA: default device capture_dev_id=OSS: /dev/dsp This gave me a fully functional CLI linphone, except for needing to switch state files to get the ring on the speaker and the call in the earpiece. Echo was present as expected, and I didn't try enabling linphone's echo cancellation. If anyone knows how to get alsa to work full-duplex I would like to know! The reason I'm glad you've got =1.7 running is that hooks for external control of linphone were included in that version. I've seen this working in yeaphone [2] and the code seemed fairly simple. If this is available in the GUI version it may give us a way to quickly add alsa state changing. It also gives us a relatively easy way to use linphone as a SIP backend for the FSO telephony interface. [1] http://3v1n0.tuxfamily.org/openmoko/linphone-VoIP-SIP-call.png [2] http://www.devbase.at/voip/yeaphone.php ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
Al Johnson wrote: On Wednesday 17 September 2008, Marco Trevisan (Treviño) wrote: Al Johnson wrote: On Tuesday 16 September 2008, Nicola Mfb wrote: [EMAIL PROTECTED] wrote: What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) I've used the CLI version of linphone, but the GUI should be small enough to fit in 480x640 too. Well, the 2.1.1 version of linphone works quite well (after editing a little the code) [1], That's great, partly for reasons I'll get to below. I'll scratch this from my todo list then :-) What changes did you need to make? And do youhave a bitbake recipe in OpenEmbedded yet? Well no... I'd have to say that I was never able to use the mokomakefile to get a working OE environment, that's way I've always used the Toolchain to compile. And also this time. So I could post just bad ipkgs here :P However I've not made many changes, just fixed some issues (like crash if there's no sip: text and automatic transformation from number to sip url [0123456789 = sip:[EMAIL PROTECTED]) and added a brute alsa state changing. Then I'd like to change the interface to make it more usable in the moko (mostly the preferences should be fixed). however the problem is another: we miss the alsa states needed to use the phone speaker as default output device and the microphone as a capture device. This night I've played a lot with this software but I wasn't able to use it as a standard phone... :| The alsa state was relatively simple to set up - so much so that I don't think I saved it. There has been at least one state file for voip posted to the list though, and I think there is one in FSO milestone 3. Well, yesterday was too late, but I didn't test the file (coming from om packages) voip-handset.state. I've tested it, but it simply set the volume of the main speaker to a lower value; it doesn't route the audio output to the phone headset speaker (the one we generally use to hear a call!). Was you able to do so? If you did it, how? I've not found any other working state file. The bit that caused problems was the audio interface. I was using 2007.2 so I killed pulseaudio to start with. The default alsa interface uses dmix, and linphone complained that this didn't allow a duplex connection. I could hear things on the Neo, but the other end couldn't hear me. I changed the .linphonerc to use OSS for the mic instead of alsa: [sound] playback_dev_id=ALSA: default device ringer_dev_id=ALSA: default device capture_dev_id=OSS: /dev/dsp My default linphone configuration was that of using only OSS. I've not tested if a called person was hearing me, however. This gave me a fully functional CLI linphone, except for needing to switch state files to get the ring on the speaker and the call in the earpiece. Echo was present as expected, and I didn't try enabling linphone's echo cancellation. That is inusable... It uses too much CPU I guess, since enabing it I can't hear the called people as expected. The reason I'm glad you've got =1.7 running is that hooks for external control of linphone were included in that version. I've seen this working in yeaphone [2] and the code seemed fairly simple. If this is available in the GUI version it may give us a way to quickly add alsa state changing. It also gives us a relatively easy way to use linphone as a SIP backend for the FSO telephony interface. Yes I guess it could be but it should be include dbus support before :P, (it has not support for it, if I'm not wrong). I've also to say that the first linphone version I got running in my freerunner was the unstable 2.9.9. It was a little more hard to compile it, but the new glade interface is too sophisticated for a so small device. Bye -- Treviño's World - Life and Linux http://www.3v1n0.net/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
On Wednesday 17 September 2008, Marco Trevisan (Treviño) wrote: Al Johnson wrote: On Wednesday 17 September 2008, Marco Trevisan (Treviño) wrote: Al Johnson wrote: On Tuesday 16 September 2008, Nicola Mfb wrote: [EMAIL PROTECTED] wrote: What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) I've used the CLI version of linphone, but the GUI should be small enough to fit in 480x640 too. Well, the 2.1.1 version of linphone works quite well (after editing a little the code) [1], That's great, partly for reasons I'll get to below. I'll scratch this from my todo list then :-) What changes did you need to make? And do youhave a bitbake recipe in OpenEmbedded yet? Well no... I'd have to say that I was never able to use the mokomakefile to get a working OE environment, that's way I've always used the Toolchain to compile. And also this time. So I could post just bad ipkgs here :P Worked for me with 2007.2 but I've not tried with anything else yet. Probably time to give the FSO equivalent a try, and this should be as good a way as any to learn about bitbake. However I've not made many changes, just fixed some issues (like crash if there's no sip: text and automatic transformation from number to sip url [0123456789 = sip:[EMAIL PROTECTED]) and added a brute alsa state changing. Then I'd like to change the interface to make it more usable in the moko (mostly the preferences should be fixed). Would be good to see. however the problem is another: we miss the alsa states needed to use the phone speaker as default output device and the microphone as a capture device. This night I've played a lot with this software but I wasn't able to use it as a standard phone... :| The alsa state was relatively simple to set up - so much so that I don't think I saved it. There has been at least one state file for voip posted to the list though, and I think there is one in FSO milestone 3. Well, yesterday was too late, but I didn't test the file (coming from om packages) voip-handset.state. I've tested it, but it simply set the volume of the main speaker to a lower value; it doesn't route the audio output to the phone headset speaker (the one we generally use to hear a call!). Was you able to do so? If you did it, how? If that's all that's wrong with it then turn down the speaker (Control 3: Headphone Playback Volume) and turn up the earpiece (Control 4: Speaker Playback Volume) I've not found any other working state file. Doesn't look like I've got it saved anywhere. I'll try to make another one but it might take a few days to get everything in place. The bit that caused problems was the audio interface. I was using 2007.2 so I killed pulseaudio to start with. The default alsa interface uses dmix, and linphone complained that this didn't allow a duplex connection. I could hear things on the Neo, but the other end couldn't hear me. I changed the .linphonerc to use OSS for the mic instead of alsa: [sound] playback_dev_id=ALSA: default device ringer_dev_id=ALSA: default device capture_dev_id=OSS: /dev/dsp My default linphone configuration was that of using only OSS. I've not tested if a called person was hearing me, however. This gave me a fully functional CLI linphone, except for needing to switch state files to get the ring on the speaker and the call in the earpiece. Echo was present as expected, and I didn't try enabling linphone's echo cancellation. That is inusable... It uses too much CPU I guess, since enabing it I can't hear the called people as expected. :-( Looks like I'll have to have another play with the Wolfson noise gate then. The reason I'm glad you've got =1.7 running is that hooks for external control of linphone were included in that version. I've seen this working in yeaphone [2] and the code seemed fairly simple. If this is available in the GUI version it may give us a way to quickly add alsa state changing. It also gives us a relatively easy way to use linphone as a SIP backend for the FSO telephony interface. Yes I guess it could be but it should be include dbus support before :P, (it has not support for it, if I'm not wrong). It wasn't using dbus in 1.7 and I don't think it's been added since, but I may have missed something. I'll have a look at the current state of linphone and the FSO telephony interface. I've also to say that the first linphone version I got running in my freerunner was the unstable 2.9.9. It was a little more hard to compile it, but the new glade interface is too sophisticated for a so small device. Bye ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
Al Johnson wrote: however the problem is another: we miss the alsa states needed to use the phone speaker as default output device and the microphone as a capture device. This night I've played a lot with this software but I wasn't able to use it as a standard phone... :| The alsa state was relatively simple to set up - so much so that I don't think I saved it. There has been at least one state file for voip posted to the list though, and I think there is one in FSO milestone 3. Well, yesterday was too late, but I didn't test the file (coming from om packages) voip-handset.state. I've tested it, but it simply set the volume of the main speaker to a lower value; it doesn't route the audio output to the phone headset speaker (the one we generally use to hear a call!). Was you able to do so? If you did it, how? If that's all that's wrong with it then turn down the speaker (Control 3: Headphone Playback Volume) and turn up the earpiece (Control 4: Speaker Playback Volume) Unfortunately it doesn't do the work. Also changing these values I keep hearing the called voice from the main speaker and not from the earpiece. This is so bad! :( I've not found any other working state file. Doesn't look like I've got it saved anywhere. I'll try to make another one but it might take a few days to get everything in place. Ok, cool... If you need my packages/sources/diffs I could upload them somewhere. Just ask! -- Treviño's World - Life and Linux http://www.3v1n0.net/ ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Qtopia and VOIP
I rebuilded Qtopia with -voip for the freerunner, but i cannot see settings-voip. Any hint? Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
Am Tuesday 16 September 2008 16:49:38 schrieb Nicola Mfb: I rebuilded Qtopia with -voip for the freerunner, but i cannot see settings-voip. Any hint? Nicola The voip option was taken out after 4.3.1 so it's not more available in 4.3.2 or 4.3.3. On the ML i've read the voip was obsolent Alex ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
On Tue, Sep 16, 2008 at 6:41 PM, Alexander Syring [EMAIL PROTECTED] wrote: Am Tuesday 16 September 2008 16:49:38 schrieb Nicola Mfb: I rebuilded Qtopia with -voip for the freerunner, but i cannot see settings-voip. Any hint? Nicola The voip option was taken out after 4.3.1 so it's not more available in 4.3.2 or 4.3.3. On the ML i've read the voip was obsolent Alex What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) regards Nicola ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
Nicola Mfb wrote: I rebuilded Qtopia with -voip for the freerunner, but i cannot see settings-voip. Any hint? Sorry, voip had to be removed from Qtopia. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
2008/9/16 Lorn Potter [EMAIL PROTECTED]: Sorry, voip had to be removed from Qtopia. Something to do with Nokia's wider commercial interests/partnerships? Neil ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
On Tuesday 16 September 2008, Nicola Mfb wrote: On Tue, Sep 16, 2008 at 6:41 PM, Alexander Syring [EMAIL PROTECTED] wrote: Am Tuesday 16 September 2008 16:49:38 schrieb Nicola Mfb: I rebuilded Qtopia with -voip for the freerunner, but i cannot see settings-voip. Any hint? Nicola The voip option was taken out after 4.3.1 so it's not more available in 4.3.2 or 4.3.3. On the ML i've read the voip was obsolent Alex What a pity!, it would be nice to have gsm/voip dialer integrated in the same application. Thanks Alex for the information, src/html tree should be cleaned :) Are there other voip clients suitable for the freerunner? (for x11 too?) I've used the CLI version of linphone, but the GUI should be small enough to fit in 480x640 too. ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
2008/9/16 Lorn Potter [EMAIL PROTECTED]: Sorry, voip had to be removed from Qtopia. Something to do with Nokia's wider commercial interests/partnerships? No, something to do with Nokia's more vigilant attorneys. Hi lorn, what other features have been removed or are under threat since Nokia joined the table?[question suitable for a software engineer...] why do Nokia view VOIP as legally dodgy? [question not suitable for a software engineer but still relevant to Qtopia's status within the FOSS community so perhaps one of your lawyers could answer?] JW ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: Qtopia and VOIP
JW wrote: 2008/9/16 Lorn Potter [EMAIL PROTECTED]: Sorry, voip had to be removed from Qtopia. Something to do with Nokia's wider commercial interests/partnerships? No, something to do with Nokia's more vigilant attorneys. Hi lorn, what other features have been removed or are under threat since Nokia joined the table?[question suitable for a software engineer...] why do Nokia view VOIP as legally dodgy? [question not suitable for a software engineer but still relevant to Qtopia's status within the FOSS community so perhaps one of your lawyers could answer?] It's not voip itself they think is dodgy. http://www.qtopia.net/modules/news/article.php?storyid=51 We are looking at other voip/sip stacks for future integration, to replace the missing one. Like they say, 'back to the drawing board'. -- Lorn 'ljp' Potter Software Engineer, Systems Group, Trolltech, a Nokia company ___ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
