Linphone is a Voice over IP ( VoIP ) application. Its purpose is to make phone calls over the data network, instead of using the telephony network.
Linphone is still not in the SHR-unstable repository but can be built following Building SHR with
cd shr-unstable/openembedded/recipes/libosip2 cp libosip2_3.1.0.bb libosip2_3.3.0.bb cd ../../.. bitbake linphone
My linphone packages are here  Because of alsa audio errors printed when running which can be seen if you run LinPhone from the commandline, I had to move away
mv /etc/asound.conf /etc/asound.conf.bak
and then put it back when done. If this is not done it get overloaded by printing the error message. I also remember there was some problem with the mic audio setting. Maybe some could add this here or maybe the koolu setting below is working. I have not tried it yet.
You may want to add gtk+ and libglade dependencies to bitbake recipe file (shr-unstable/openembedded/recipes/linphone/linphone_3.1.0.bb):
DEPENDS = "intltool libosip2 speex libogg alsa-lib readline libexosip2 gtk+ libglade"
Here is a way to get the command line linphone working on Qtopia. Ideally this could be interfaced with the dialer somehow. Download link broken.
echo "src/gz celtune http://rabenfrost.net/celtune/ipk/armv4t" >/etc/opkg/general-feed.conf opkg update opkg install libmediastreamer0 opkg -nodeps install linphone opkg -nodeps install liblinphone2 linphonec linphone-rings opkg install libexosip2 cd /etc/ wget http://www.koolu.org/asound.conf wget http://www.koolu.org/voip-handset.state alsactl -f voip-handset.state restore # This step required to set proper audio parameters linphonec soundcard use 0 proxy add
Initial testing of this had the audio routed properly through the earpiece and good audio from the microphone. There was some echo on the non-Freerunner side, and on initial connection, a bit of a beeping sound. Otherwise, it's a go. Linphone uses only about 10-12% CPU (was using a PCM codec). Sound was decent to a cell phone in Canada using a Wifi connected Freerunner based in Costa Rica.
Here is a quick and dirty how-to. Enhancements can be made to both the voip-handset.state and asound.conf files (these were my initial working scripts). Thanks to Celtune for the excellent repository that is used below.
Regards, Brian Code, Koolu
If you want to avoid installing linphone you can use the FDOM image. This includes the linphone installation. You just have to enter your sip-account details.
Configuration of Linphone
Setting of your Idenitiy, SIP-address and SIP-Server
''Enter proxy sip address:'' sip:sip.provider.com ''Your identity for this proxy:'' sip:firstname.lastname@example.org ''Do you want to register on this proxy (yes/no):'' yes ''Specify register expiration time in seconds (default is 600):'' ''Expiration:'' 600 seconds or so
If you want to an Asterisk open source PBXi, telephony engine, and telephony applications toolkit.
[jthneo] type=friend username=jthneo callerid= jth <jth> secret=thesecret qualify=no ; linphone will become unreachable if qualify=yes host=dynamic nat=yes canreinvite=yes disallow=all ; allow the sensible codecs you want allow=ulaw allow=alaw ;allow=gsm ;allow=speex