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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'>That’s the same VM guest moved to a different VM host (not really what I was looking forward). In this case it’s an entirely new host with Asterisk having no state/session information, but my app would repopulate the session info and try to re-establish the call.<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'>Given SIP over TCP I suspect the answer is still now (since opening the connection on a new host would result in a new syn handshake, different source port used by Asterisk etc…)<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";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Andres<br><b>Sent:</b> Thursday, January 12, 2017 12:04 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Replacing PBX during a call in progress<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On 1/12/17 11:09 AM, Telium Technical Support wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>This was asked many years ago but I thought I would check to see if things have changed. Is it possible to take over a call in progress – using a replacement Asterisk server? <o:p></o:p></p></blockquote><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'>One plausible scenario I can think of is if you are running VMware VMs. Using the vMotion feature would accomplish subsecond VM live moves.<br><br><o:p></o:p></span></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>In other words, if 2 user agents are connected through an Asterisk PBX, and I tracked the call ID, IP of each UA (and anything else needed), could I remove the PBX and put a new one in its place (at the same IP address) and resume the call? Somehow keeping the call up on the UA’s and telling Asterisk to just resume a call given specified parameters (so the UA’s wouldn’t notice the change)?<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br><br><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br><br><br><o:p></o:p></span></p><pre>-- <o:p></o:p></pre><pre>Technical Support<o:p></o:p></pre><pre><a href="http://www.telesip.net">http://www.telesip.net</a><o:p></o:p></pre></div></body></html>