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

2017-01-12 Thread 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

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

2017-01-12 Thread 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

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

2017-01-12 Thread 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:

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

2017-01-12 Thread 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

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

2017-01-12 Thread 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

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

2017-01-12 Thread 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

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

2017-01-12 Thread 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

Re: [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

2017-01-12 Thread Olivier
Hello, I filed issue https://issues.asterisk.org/jira/browse/ASTERISK-26717 for this. Thanks for helping 2017-01-03 16:17 GMT+01:00 Joshua Colp : > On Tue, Jan 3, 2017, at 11:04 AM, Olivier wrote: > > Hello, > > > > On a newly built Asterisk 13.13.1 system, I can't make HEP

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

2017-01-12 Thread Olivier
I removed my /usr/src/asterisk-13.13.1 directory and rebuilt from scratch: everything worked OK. There must be a reason why previous attempts failed with libsrtp but my knowledge of make and Asterisk building is too limited to allow me to constructive findings (which is very frustrating and