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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>FreeSwitch noob here.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I currently have a mostly working setup here but I have one rather major problem. I can send and receive calls from most numbers perfectly, the only numbers that seem to be giving me problems are AT&T mobile numbers. Inbound AT&T calls work fine, but outbound AT&T calls only have inbound audio, audio from my switch to AT&T is lost. Naturally most of the mobile numbers I need to call are AT&T.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>According to my VoIP provider the problem is I’m sending out asynchronous RTP (my send and receive ports are different) and that the media gateway on the other end only allows synchronous RTP so my send and receive ports need to be the same. Unfortunately I can’t seem to find any setting in FS to enable sync RTP. I found this under RTP issues:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'><!-- This will use the async timer (this one you can also set to none if you don't want asynchronous RTP traffic) --> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN style='font-size:10.0pt;font-family:"Courier New"'><param name="rtp-timer-name" value="soft"/><o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>But setting rtp-timer-name to none still uses different RTP ports for the outbound traffic.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>