[Freeswitch-users] SIP Re-invite

[DJB]

We
have a customer that we are sending calls to off the FS and here is the issue: 
 
Call
is initially setup fine and they send a first re-invite with media 0.0.0.0 to
place the caller on hold. FS sends a 200 ok to this first re-invite fine 
 
They
then send a second re-invite with their media IP to cut through media and the
FS sends a 200 OK to this fine. At this point the call is fine 
 
30
minutes later they send a third re-invite because according to them it is
strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has
the exact same media IP and UDP pot information as the second re-invite does.
The problem is FS does not respond to this third re-invite AT ALL. It doesn’t
send a 100 trying a 200 OK nothing so this causes the call to be dropped as the
other end does not recieve a response from FS.  

One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds.

We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode.

Thank you very much.


      
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/e9398029/attachment-0002.html