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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>All,<o:p></o:p></p><p class=MsoNormal> Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (<a href="http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS">http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS</a>) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2<o:p></o:p></p><p class=MsoNormal>Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>-Peter<o:p></o:p></p></div></body></html>