[asterisk-users] Replacing PBX during a call in progress
support at telium.ca
Thu Jan 12 11:11:28 CST 2017
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?
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dovid Bender
Sent: Thursday, January 12, 2017 12:06 PM
To: andres at telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Replacing PBX during a call in progress
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.
On Thu, Jan 12, 2017 at 12:03 PM, Andres <andres at telesip.net> wrote:
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)?
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