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)

Use iax/iax-client.h instead of iax-client.h (from libs/libiax2)

Must be fixed, with bug #10

Loading dynamic number of account, depending on user_cfg.h constant

Allow different stun server foreach account

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).

First Account-based save/load implementation

Two variables in sflphonedrc configuration file to disable tone or dtmf sound

Add driver in/out capacity (experimental)
Add switch <CSeq> audiodriver command (experimental)

Add debugging information for initialization, the audiolayer initialization is too long

Fixes for handling line: don't stop the tone, if it's not for the actual line
Fixes for the TCP IO: accept every incoming line, before sending one

Fixes for check inside global ringtones directory if the preference don't have a path
Remove warning inside request.h
Add two constants for directory seperator inside user_cfg

Cleaning compilation and debug message

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

Add a Local_IP variable inside the configuration file,
but later it will use the IP inside the request/response.

I finally call the asterisk phone with my computer.

Méga ménage pour enlever le module de configuration
et le remplacer par Conf::Config qui n'est pas singleton.

Correct some typo
Try to correct two servers failure, but didn't work
Correct getcallstatus 205/206

configgetall works
configget works
configset not tested yet
getzeroconf not tested yet

Initialize the sound volume inside init() -> initVolume()
Add the argtokenizer.cpp file
I didn't remove the sound configuration for qt (x/y)

Patch to integrate a text server
--enable-maintener to play with it

Zeroconf Integration
and
managerimpl.cpp bug fix for gui setup (when sflphonerc file doesn't exist)