[OpenSIPS-Users] Asterisk and ACK

[Douglas Lane]

Hi All,

I've picked up an issue with Asterisk not adding a Via header when calls 
are passed to it from OpenSIPS. Now this doesn't seem to affect the 
following call flow:

UA ------> OpenSIPS ------> Asterisk --------> Callee

However, when the below call flow happens, the callee side answers the 
call, but the ACK never reaches the asterisk server, and we only get 1 
way audio:

UA ------> OpenSIPS 1 -----> OpenSIPS 2 -------> Asterisk ------> Callee

The 200 OK is passed back from the Callee all the way to the UA using 
stateful transactions, however, because Asterisk never adds itself to 
the Via header, OpenSIPS 2 hits the last Via message which is its own IP 
address and loops the ACK around forever till the transaction eventually 
dies.

Any suggestions if there is a fix for the above. I've observed this in 
the 1.4.29 version of asterisk, as well as version 1.6.2.9

Look forward to your feedback.

Many thanks for all the assistance thus far.

Thanks
Doug