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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='color:#1F497D'>We have attempted to upgrade to the latest 1.8 release as well as the latest asterisk 10 release and are experiencing the same issue with session-timers and TCP where the session-timers seem to trigger and go out according to the CLI, but we are not seeing that traffic via tcpdump on the asterisk server and the call stays up when it should be disconnected. <o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Anyone have any ideas? If more information is needed I would be happy to provide, just let me know what may help.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Thanks<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><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"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Aaron Hamstra<br><b>Sent:</b> Friday, June 01, 2012 1:28 PM<br><b>To:</b> asterisk-users@lists.digium.com<br><b>Subject:</b> [asterisk-users] Session-timers and TCP<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><a name="_top"></a><o:p> </o:p></p><p class=MsoNormal>All,<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>We are having issues with one of our customers. They typically are using remote sip clients on smart phones. For the purpose of allowing the apps to work properly in the background we have to utilize TCP which works fine.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>The problem comes up when the softphone loses connectivity for some reason. The session timers are not ending the call as they do on a UDP session. Basically from the sip debug it sends the re-invite for the session timer according to the sip debug and it appears all is fine instead of not getting a response back from the client and disconnecting the call as it does with udp. There is no way it is getting a response back from the client however as the client has no network connectivity.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I have run some tcpdump’s on the server and when tracing the call I actually never see those re-invites going out at all from the server.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>We are running asterisk 1.8.7.0 currently.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I can reproduce the issue at will by establishing a call from a softphone and then putting it into airplane mode to simulate the connectivity loss. <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Are session-timers expected to work with tcp? If so can anyone tell me where to look to see what might be going on?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thanks in Advance.<o:p></o:p></p></div></body></html>