Qtopia and VOIP
"Marco Trevisan (Treviño)"
mail at 3v1n0.net
Thu Sep 18 00:41:06 CEST 2008
Al Johnson wrote:
> On Wednesday 17 September 2008, Marco Trevisan (Treviño) wrote:
>> Al Johnson wrote:
>>> On Tuesday 16 September 2008, Nicola Mfb wrote:
>>>> openmoko at asyring.homeip.net> 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) ,
> 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:0123456789 at voip.provider.net]) 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
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:
> 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
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  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.
Treviño's World - Life and Linux
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