Asterisk 1.8.11 from tarball
Add git ignore file
Parse sip.conf to define which RTP engine to use
Add 'prepare_bridge' callback inside rtp_engine

This is necessary when activating the voice inside
the Comcerto device, because we need information
from both sides.
Still work in progress ...
Add set_chan function for RTP instance
Reposition prepare_bridge to end of ast_rtp_engine structure
Add DAHDI span and channel information inside ast_channel

When creating a DAHDI channel, copy span and channel information
present in dahdi_private to ast_channel, so that anyone can
know which analog channel is being used. This is useful to us, since
we need to bridge PSTN <-> RTP and this is done in the Comcerto
RTP Engine.
BUGFIX Dereference RTP instance after calling prepare_bridge()

Fixes memory leak on RTP instances
Comcerto: Add DTMF handling when method is SIP Info
Add support to G726 variants : 16, 24 and 40 kbps
Change CONFIG_PD3 to PLATFORM_COMCERTO on DSP dependent code
Disable software DSP functions for Comcerto platform
Link chan_dahdi.so with libAMG
PLATFORM_COMCERTO: Check DSP counters to determine if we are receiving RTP traffic
PLATFORM COMCERTO: Ugly fix when dealing with frame with wrong format

For some unknown reason, a frame with a different format was being received,
causing Asterisk to change the codec in the RTP stream. The fact is that
voice frames don't go through Asterisk here, so this must be a bogus frame.
Fix wrong codec setting on RTP engine
Remove debug for received RTP packets
Add faxpassthru callback to RTP engine
Handle Fax Passthru

Still needs some tests
Hack to configure the right format on RTP engine