[OpenSIPS-Users] Asterisk and ACK

Bogdan-Andrei Iancu bogdan at voice-system.ro
Thu Jun 10 22:51:18 CEST 2010

Hi Doug,

so probably the 200 OK gets back to UA by skipping Asterisk (if no VIA 
is added). But ACK has nothing to do with the VIA in INVITE request....

maybe you can post a trace of the looping ACK.


Douglas Lane wrote:
> 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
> Look forward to your feedback.
> Many thanks for all the assistance thus far.
> Thanks
> Doug
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Bogdan-Andrei Iancu

More information about the Users mailing list