[asterisk-users] Replacing PBX during a call in progress

[TSG]

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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Technical Support
http://www.telesip.net


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