Hello,

We have a scenario like this:

SUA -> Kamailio with registrar module -> Asterisk

A call from the SUA is set up with SIP timers, and after 15 minutes Asterisk sends a re-INVITE to Kamailio to forward on to the SUA. That re-INVITE has a RURI with the address and port of the SUA at the time the call started.

Now if the SUA re-registers after the call starts and before the re-INVITE, and is on a new address or port number, then the re-INVITE never gets to the phone.

Obviously Kamailio should send the re-INVITE to the new address/port, but is not. The re-INVITE is routed using the lookup() function.

Can anyone suggest why this is happening?

Thanks in advance,

--
David Cunningham, Voisonics
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