Hello - We have an Opensips 1.4 server that routes incoming calls to a couple of different Asterisk servers and to upstream providers. All working great and with the current config, the Opensips server only handles the SIP traffic - all of the audio is between the UAs and Asterisk servers.
Am building another Opensips server and decided to do it with the 1.6.3 release. With virtually the same config (really only had to change a couple of things at the top like loading signal.so, dropping the loading of xlog.so, etc), now Opensips is in the picture for all of the audio as well as SIP traffic. Was troubleshooting an issue with no audio in either direction when calling in - so was capturing traffic on the Opensips server, saw the SIP traffic, call relayed correctly to the Asterisk server where and IVR was played and all the audio was going back to the Opensips server. Did the same test on the Opensips 1.4 server and no audio packets - which is what I want. Done a days searching and am not finding the fix. Someone know what changed and how I can get the same behavior I have with 1.4 with the 1.6 release? Thanks! Chris _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users