[Asterisk-Users] Sip bug...problem seem to be fixed in trunk. How do I find the patch for 1.2

[Doug G]

I am having a problem with sip in asterisk 1.2.1 & 1.2.8 .   I have an
account setup with a sip provider.  The inbound call is coming from a
SIP proxy, the call is setup (I have audio) and then drops down after
15sec.    

What I see in sip traces is that the sip proxy is sending "200 ok"
asterisk is responding with a "ACK" however the ACK is send to the to a
different host then the one that sent the "200 ok".  After the remote
proxy retransmits a few times and receives no ACK it sends a BYE.  

I tried loading trunk, (from a few days ago) and this problem appears to
be fixed as it works just fine.    The only problem is this is a
production system and I do not feel ok running trunk.  What I would like
to do is just load the patch for this problem then wait for 1.4 Release.
I have been reading the SVN log for chan_sip, however I am unable to
identify the problem.

Anyone know what fix solved this problem?   

Doug