Telium Technical Support
2017-Jan-12 16:09 UTC
[asterisk-users] Replacing PBX during a call in progress
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)? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170112/54871fc1/attachment.html>
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170112/b0e39046/attachment.html>
Dovid Bender
2017-Jan-12 17:05 UTC
[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)? > > > > > > > > > > > -- > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170112/15f80a09/attachment.html>
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170112/a3a31a28/attachment.html>
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 .
Patrick Labbett
2017-Jan-12 17:39 UTC
[asterisk-users] Replacing PBX during a call in progress
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 <asterisk_list at earthshod.co.uk> 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 . -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170112/7de24bbb/attachment.html>