You are re-posting the same question again without providing any additional info: http://lists.opensips.org/pipermail/users/2011-February/016626.html
Regards, Ovidiu Sas On Mon, Feb 7, 2011 at 12:20 PM, Chris Stone <axi...@gmail.com> wrote: > 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 > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users