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

[Patrick Labbett]

Keepalived + heartbeatd allows you to maintain a a floating IP between two
machines. If those two machines had configs, internal state synced, and the
IP is configured to float automatically between the two based on which is
actively up, would it be possible to not drop a call should the active host
go down.

On Thu, Jan 12, 2017 at 12:21 PM A J Stiles 
wrote:

On Thursday 12 Jan 2017, 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?
>
> 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)?

I doubt there is any chance whatsoever of that working!  For a start, you
can't have two machines on the same subnet with the same IP address.  It
just
does not work.  And there is all manner of internal state that would have to
be replicated onto the new server.

When the clicky-clicky exchange in the village where I grew up was updated
to
System Y sometime in the early 1990s, all the phones went dead for about 30
minutes and calls in progress were cut off.

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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

[A J Stiles]

On Thursday 12 Jan 2017, 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?
> 
> 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)?

I doubt there is any chance whatsoever of that working!  For a start, you 
can't have two machines on the same subnet with the same IP address.  It just 
does not work.  And there is all manner of internal state that would have to 
be replicated onto the new server.

When the clicky-clicky exchange in the village where I grew up was updated to 
System Y sometime in the early 1990s, all the phones went dead for about 30 
minutes and calls in progress were cut off.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
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Re: [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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Thursday, January 12, 2017 12:06 PM
To: and...@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 <and...@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)?

 

 

 

 






-- 
Technical Support
http://www.telesip.net


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Re: [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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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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Re: [asterisk-users] Replacing PBX during a call in progress

[Dovid Bender]

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  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)?
>
>
>
>
>
>
>
>
>
>
> --
> Technical Supporthttp://www.telesip.net
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Replacing PBX during a call in progress

[Andres]

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

[Telium Technical Support]

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?  

 

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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