ReInvite implementation for two no-nat telephone

Don't close sflphone when portaudio failed to load
Remove enable_audio variable (non-protected) in SipCall

Try to remove unlock from deamon
But it is unstable

Debugging incoming message
It's seems like we don't listen to the good port

Now, only the audiolayer control the volume

Namespace cleaning
Remove exception bug on 'stop'

Enlever des debugs
Changer les short CallId en CALLID (unsigned int)
Completer mutex pour la gestion des call

Cleaning the call classes, remove the setStatus()
Hangup call when we leave the phone

Add my copyright to the file
Add a CodecMap (the map) and CodecDescriptorMap (the one that initialize it)
for the request: list seq1 codecdescriptor

Remove the old config
Add mutex to audiolayer (its share between 3 or 4 threads)
Add mutex to the tone (its share by two threads)
Simplify the tone interface...
Remove the qtGUImainwindow from the compilation with the ENABLE_MAINTENER

It's working
Lot of debugging code right now
You need STUN, because it doesn't work without it