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

[TSG]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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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