Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers

[Doug G]

I think I have 4 options.  

1, Modify chan_sip.c to update a new field in sipusers realtime table
with the status of the sip peer/user.  Then use agi to dial sip calls.
Check the status field if OK then dial the fullcontact from the sip
table.  If not goto voicemail or where ever else I want the call to go..
The UA would only register to one server, so only one server *should* be
writing to the database. (If not the code will be modified to do so.)
Other servers would be read only from my AGI.  

2, Use Asterisk management interface to find the status of the sip peer.
Then dial fullcontact if peer is active. Should be easy to implement.
Problem is I would have to actively poll each server in the farm.  

3, Use SER as the sip router and asterisk as an application/media
server. Then all sip UA would register to the SER. Should scale higher,
but does add a level of complexity.    

4, Continue to use IAX trunk to dial the other switch. Then hope that
realtime has been improved by the time I need the 3rd server.  It is a
failover but not a load balancer. 


Any thoughts?   Am I completely off here?  


Doug



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter
Bowyer
Sent: Wednesday, January 04, 2006 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]
RealtimeMultipleAsterisk boxes, iaxusers

On 04/01/06, Alistair Cunningham <acunningham at integrics.com> wrote:
> Tijmen,
>
> We use SER for this to load balance across multiple Asterisks. We then
> use a custom program to monitor the health of the Asterisks and update
> SER's configuration should one go down. 2 SERs share a single IP
address
> for users to contact using heartbeat.

I was thinking along the same lines, but for a dynamic setup it should
be possible to have SER/OpenSER load balance REGISTER requests
according to some strategy/metrics, and then forward INVITEs and other
call-related traffic to the 'right' back-end server.

Probably lots of reasons why this is too complicated, though....

Peter

--
Peter Bowyer
Email: peter at bowyer.org
Tel: +44 1296 768003
VoIP: sip:peter at bowyer.org
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