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<DIV dir=ltr><FONT size=2>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. </DIV></DIV>
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<P>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. </P>
<P>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.</P>
<P>Anyone know what fix solved this problem? Any tips to finding it
in SVN?</P>
<P>Thanks.</P>
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