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

Modify Fedora Core spec to support fedora core 6 (libsamplerate, libeXosip2)

Adding magnetism to dtmf window
Adding sample rate combobox

Samplerate is now a dependancy.

(adding codec map to new incoming call)

Adding debugging information

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

Makefile.am and m4 updates for warnings and paths, README updates.

Add hangup/hold/unhold to sflphone-cli

Fix onhold/getcallstatus issues

Documentation generation (remove copyright as doxygen brief summary)
doygen Doxyfile

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 old class/struct