[asterisk-ha-clustering] New setup

[David Ruggles]

Ok, I understand what you're saying about losing the calls, I don't really
see any way around losing calls now that I think about, because there's also
transcoding and bridging going on in the pbx, it's not pure sip. So, throw
that desire out the window and where do I start with the failover config?

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer	Safe Data, Inc.
(910) 285-7200	david at safedatausa.com



-----Original Message-----
From: asterisk-ha-clustering-bounces at lists.digium.com
[mailto:asterisk-ha-clustering-bounces at lists.digium.com] On Behalf Of Olle
E. Johansson
Sent: Wednesday, April 15, 2009 5:09 PM
To: Asterisk High Availability and Clustering List - Non-Commercial
Discussion
Subject: Re: [asterisk-ha-clustering] New setup



15 apr 2009 kl. 22.57 skrev Leif Madsen:

> On Wed, Apr 15, 2009 at 4:41 PM, David Ruggles  
> <david at safedatausa.com> wrote:
>> I'm getting ready to install an Asterisk PBX in our office. I have  
>> two
>> identical boxes and I would like to setup a configuration that  
>> provides for
>> failover, preferably without the loss of any active calls.
>>
>> Based on messages in this list, I believe this should be possible,  
>> but I'm
>> not sure where or how to start.
>
> This is possible as long as you reinvite the media between the devices
> directly; if Asterisk is handling the media and the box goes down,
> then the media will be taken with it. If you are using Asterisk 1.6.x,
> then you can also use SIP session timers to make sure the devices
> don't go away without telling you they are going away (for example a
> power outage at a the remote location, or someone rebooting their
> phone).

Using session timers actually mean that the call will die if the server
dies...

In most setups, unless you do what Leif started with telling you,
it's gonna be very hard keeping active calls if a PBX starts. There  
are many
ways to handle the next call seconds after a server fails, but in  
order to
keep the call you might need something else than Asterisk in reality.
Requirements for DTMF transfers, media recording or something else
in many cases means that media goes through your asterisk server.

A few years ago IBM demoed a solution that kept the calls on the SIP
side of Asterisk. It was just a demo. I vaguely remember a Digium
OEM that had a solution that kept running calls, but fail to remember
the name of the company. They had the booth on the left side of IBM...

Oh, the joy of getting old and forget stuff...

/O

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