We have an Opensips 1.4 installation that routes calls to multiple Asterisk servers. We have a perl module that Opensips runs that does an SQL query to find the Asterisk server that the call should be sent to. All works great and Opensips handles only the SIP traffic - all the SDP/RTP traffic is between the UAs and the Asterisk servers.
Getting a new Opensips server ready to go online. Using the same config (with minor changes such as the addition of loading signal.so, removing xlog.so, etc) and Opensips 1.6.3. In testing, I was finding there was no audio (either direction) for calls. Did a packet capture on the Asterisk server and Opensips server and found that the outgoing SDP/RTP packets were also being routed by Asterisk back to the Opensips server and the incoming packets were also going to Opensips. This is not what I want - would like the same behavior as we have with 1.4 where only the SIP traffic goes through the Opensips server. Have done a good amount of research to resolve this and I am not finding anything helpful..... Can anyone tell me why I am seeing this change in 1.6 v.s. 1.4 and how I can get 1.6 to behave the same as with 1.4 with regards to the audio traffic? Chris _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
