[SR-Users] Balancing Asterisk

[Daniel-Constantin Mierla]

Hello,

On 3/7/11 12:51 AM, Andy Lippitt wrote:
>
> Hello all,
>
> I've read as many of the asterisk balancing threads as I can find.  
> Either my situation is unusual or I simply haven't understood anything 
> I've read.
>
> In short, I'm building an web/phone mashup which uses Asterisk's AGI 
> to get its work done.  My only users are on the PSTN connected to 
> Asterisk through a SIP trunk provider.  So presently, in and out 
> through the same trunk, apps live on the single Asterisk box.
>
> My goal is scaling and failover.  I don't have any need for cross talk 
> or transfers between the asterisk instances, and the algo's in 
> dispatcher seem fine.  It seems to me that I should be setting the 
> sip-router up a replacement for the existing peer in Asterisk.  What 
> leaves me scratching my head is how I then register the sip-router 
> with the upstream provider.  Alternatively, if I use the sip-router as 
> an outboundproxy from asterisk (which seems like it's going to take 
> some hacking to make this work in 1.4), doesn't this now mean I have 
> multiple UAC's trying to register for the same name?
>
> Can someone set me on the right track?
>
the recommended way is to get IP-based authentication and peering with 
your provider, in this way you don't need to authenticate calls out 
neither send registrations - kamailio/ser is a proxy at its core.

The alternative is to use uac module, beware of its limitations 
regarding authentication:
http://kamailio.org/docs/modules/stable/modules_k/uac.html

In case you still need a b2bua-like interaction with the provider, see 
our related project - sip express media server (sems): 
http://iptel.org/sems - the sources are in the same git repository 
hosted at sip-router.org

Cheers,
Daniel

> Thanks,
>
> Andy Lippitt
>
>
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-- 
Daniel-Constantin Mierla
http://www.asipto.com

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