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Hello,<br>
<br>
On 3/7/11 12:51 AM, Andy Lippitt wrote:
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<p class="MsoNormal">Hello all,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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?<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Can someone set me on the right track?</p>
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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.<br>
<br>
The alternative is to use uac module, beware of its limitations
regarding authentication:<br>
<a class="moz-txt-link-freetext" href="http://kamailio.org/docs/modules/stable/modules_k/uac.html">http://kamailio.org/docs/modules/stable/modules_k/uac.html</a><br>
<br>
In case you still need a b2bua-like interaction with the provider,
see our related project - sip express media server (sems):
<a class="moz-txt-link-freetext" href="http://iptel.org/sems">http://iptel.org/sems</a> - the sources are in the same git repository
hosted at sip-router.org<br>
<br>
Cheers,<br>
Daniel<br>
<br>
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<p class="MsoNormal"><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks,<o:p></o:p></p>
<p class="MsoNormal">Andy Lippitt<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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