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update to new sofia snapshot.  Fixes a memory leak on invite, a timer bug, and several other issues as follows:

Thu Nov 16 07:23:30 Eastern Standard Time 2006  Pekka.Pessi@nokia.com
  * nta.c: setting the local sequence number of nta_leg_t only when first reques
t is sent.

  Application can now set the initial value of CSeq either in nta_leg_create()
  or in nta_outgoing_*create() (or nta_msg_request_complete()).

  * nua_session.c: fixed session timer negotiation when UAS does refreshing with
 INVITEs

  The session-expires header had "uac" even when uac did not support timer.
  The UAS failed to send re-INVITEs.

  Thanks for Chung Pak Lai for reporting this problem.

  * bnf: added host_cmp().

  * outbound.c: using host_cmp() to check if Via host and received parameter dif
fer

  Bug reported by Marc Blanchet.

  * nua_session.c: fixed leak in incomin INVITE processing.

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fix memory leak, don't need su_home_init after su_home new, and use su_home_unref instead of su_home_deinit

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disable registration probe and options keep-alive for now as there is a bug in the implementation of this on the sofia side, and it causes the registration to never happen, and for it to spin re-registering out of control forever in some situations with UA's that send 404's to OPTIONS packets, or when behind nat on either side.  This will get fixed in sofia, then become a configurable option on our side.

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ampersand was being used to indicate a function and a double call change it to carrot ^ for functions

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This addition lets you set artifical ringback on a channel
that is waiting for an originated call to be answered.

the syntax is 

<action application="set" data="ringback=[data]"/>

where data is either the full path to an audio file
or a teletone generation script..


syntax of teletone scripts

LEGEND:

0-9,a-d,*,# (standard dtmf tones)

variables: c,r,d,v,>,<,+,w,l,L,%

c (channels)        - Sets the number of channels.
r (rate)            - Sets the sample rate.
d (duration)        - Sets the default tone duration.
v (volume)          - Sets the default volume.
> (decrease vol)    - factor to decrease volume by per frame (0 for even decrease across duration).
< (increase vol)    - factor to increase volume by per frame (0 for even increase across duration).
+ (step)            - factor to step by used by < and >.
w (wait)            - default silence after each tone.
l (loops)           - number of times to repeat each tone in the script.
L (LOOPS)           - number of times to repeat the the whole script.
% (manual tone)     - a generic tone specified by a duration, a wait and a list of frequencies.

standard tones can have custom duration per use with the () modifier
7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds

EXAMPLES

UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms]
%(400,200,400,450);%(400,2200,400,450)

US Ring Tone [440+480 hz on for 2000ms off for 4000ms]
%(2000,4000,440,480)

ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms
v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)

SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait
%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)

ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait
%(100,100,1400,2060,2450,2600)



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logic fix on rtp_common_read on a not ready session, init bytes to 0 so the checks outside of the while loop work.

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