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
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
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:
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
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
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
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
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
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