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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hello,<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways. I have almost everything working with a few exceptions. One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can’t match up to a call because it tore the call down already.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The sequence:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>-<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC. <o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>-<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways).<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>-<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>My cell phone rings, I pick it up, then hang up the call.<o:p></o:p></p><p class=MsoListParagraph style='text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><span style='mso-list:Ignore'>-<span style='font:7.0pt "Times New Roman"'> </span></span><![endif]>The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>If I originate a call from my machine with a soft phone, it works fine. The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I have pasted logs for the session at <a href="http://pastebin.freeswitch.org/16037">http://pastebin.freeswitch.org/16037</a>.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thanks!<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>-----------------------<o:p></o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;color:#1F497D'>Eric Z. Beard, CTO</span></b><span style='font-size:10.0pt;color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>Loop LLC<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>w (877) 850-2010 x9249<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'>m (727) 776-2768<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;color:#1F497D'><a href="mailto:eric@loopfx.com"><span style='color:blue'>eric@loopfx.com</span></a><o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>