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.
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.) From: [email protected] [mailto:[email protected]] On Behalf Of Andres Sent: Thursday, January 12, 2017 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replacing PBX during a call in progress On 1/12/17 11:09 AM, Telium Technical Support wrote: 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? One plausible scenario I can think of is if you are running VMware VMs. Using the vMotion feature would accomplish subsecond VM live moves. 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)? -- Technical Support http://www.telesip.net
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