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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