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.
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
