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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Make sure you have the RTP ports forwarded in on your router.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> app_rpt-users-bounces@ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] <b>On Behalf Of </b>Scott<br><b>Sent:</b> Saturday, November 10, 2012 5:45 PM<br><b>To:</b> App_rpt-users@ohnosec.org<br><b>Subject:</b> [App_rpt-users] sip.allstarlink.org sip session no-audio<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>When trying to connect from a sip-phone to the portal, <a href="http://sip.allstarlink.org">sip.allstarlink.org</a> using, <br><br>sip://<a href="http://1:allstar@sip.allstarlink.org:5060/0">1:allstar@sip.allstarlink.org:5060/0</a><br>-or-<br>sip://<a href="http://1:allstar@sip.allstarlink.org:5060/">1:allstar@sip.allstarlink.org:5060/</a><node_number><br>-or-<br>sip://<a href="mailto:0@sip.allstarlink.org">0@sip.allstarlink.org</a><br><br>It connects, but there's no audio. No errors. PCMU@8000 correctly negotiated. I've tried two SIP-phone clients and a FreeSWITCH user-agent with the same negative outcomes. I've tried variations on the connect stream with same negative results. The BRIA soft-phone even handshakes GSM but still no audio.<br><br>Is there something I'm missing?<br><br>Thanks,<br><br><br><o:p></o:p></p></div></body></html>