[Asterisk-Users] Sip bug: Problem seem to be fixed in trunk. How do I find a 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?   Any tips to finding it in SVN?

Thanks.

 

 

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