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 _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
