are you running that on the FS box ? run the inline sip console trace to confirm it's not sending to somewhere else based on a changed contact addr. On Wed, Apr 21, 2010 at 2:05 PM, Emil Berg <emilbergg@gmail.com> wrote:
Hello,
I'm not sure that this is the right mailing list for this question, but I'll ask anyway :) Are there any known issues with sip over tcp?
In some scenarios, freeswitch doesn't behave in a standard way. Attached a picture with flow graph (from wireshark), where you can see such a problematic behavior. When user1 invites user2 (through the server), the server routes the invite message to user2, but when user2 responds, the server doesn't send an ack to this user!
Looking forward to your reply, Thanks,
Emil.
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