Sampling rate values are no more hardcoded. The sampling rate of the audio layer
 and the frame size can be set in the user config file. The clock rate of the codec we use
in the rtp session is set with his actual value, but can be changed by modifying
the available codec in the user config file (only G711 for now)

Tiny doc fixes and typos.

This was due to libsamplerate's call to src_simple, which created glitches.
Now we use src_process, the 'Full API' as written on the SRC (libsamplerate)
website: http://www.mega-nerd.com/SRC/api_full.html

+ Add .gitignore

Samplerate is now a dependancy.

(adding codec map to new incoming call)

Adding debugging information

Fix onhold/getcallstatus issues

New account implementation.
Need to test more and add dynamic handling in Qt Interface.
SipCall, Call, VoIPLink were rewritten completly
Configuration now in [SIP0] section of .sflphonedrc
The debug of sflphoned is very verbose.
sflphone-cli may not work right now (need to send SIP0 as accountId).

VoIPLink integration inside account
getAccount() patch for managerimpl
Remove old voiplink vector in managerimpl

New STUN logic to discover true firewall port (NAT)
Send true IP/port in SDP message
Remove big ###### debugging message
New logic to listening on SIP port: 2 tries

Remove debug message
Add changes to project files
Fix a nonat-nonat bug (reinvite)

Fixe issue for current call id switching and for incoming call signal

Only switch call when no one are selected

ReInvite implementation for two no-nat telephone

Handling onhold answering call

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

Added debugging messages to detect the stop bugs when you play a tone

Try to remove unlock from deamon
But it is unstable

Debugging incoming call

Remove some comment
Correcting some sound issue
Using fix FRAME_PER_BUFFER is better than 0

Forget to commit

Fixes for wrong audio device index when starting

Corrected the no-mic-sound send bug
But there is again some noise on the playtone, I
really don't know why...

Now, only the audiolayer control the volume

Call "establish" message
Add url to tonezone web site

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