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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'>Can re-invites be sent AFTER the first Asterisk server has been shut down? (If the first Asterisk server is still up then it’s a gracefull transition, but I’m assuming the first Asterisk server is simply unplugged). And can they be sent from a NEW asterisk server?</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><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"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Dovid Bender<br><b>Sent:</b> Thursday, January 12, 2017 12:06 PM<br><b>To:</b> andres@telesip.net; 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><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>As Andres mentioned you can use VMWare. Another option would be to send a re-invite to both devices and send them to another server.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Thu, Jan 12, 2017 at 12:03 PM, Andres <<a href="mailto:andres@telesip.net" target="_blank">andres@telesip.net</a>> wrote:<o:p></o:p></p><div><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'><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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></div></blockquote><p class=MsoNormal>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></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><p class=MsoNormal><span style='color:#888888'><br><br><br><span class=hoenzb><o:p></o:p></span></span></p><pre><span style='color:#888888'>-- <o:p></o:p></span></pre><pre><span style='color:#888888'>Technical Support<o:p></o:p></span></pre><pre><span style='color:#888888'><a href="http://www.telesip.net" target="_blank">http://www.telesip.net</a></span><o:p></o:p></pre></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>