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