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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Darren,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I will also want use this setup where the remote PBX is Cisco,
Avaya or Nortel. I need this setup in my lab purely to make some of my testing
and monitoring easier (long story, not really important here).<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Simon<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Darren Sessions
[mailto:dmsessions@gmail.com] <br>
<b>Sent:</b> Thursday, September 04, 2008 6:28 PM<br>
<b>To:</b> Simon Shaw; users@lists.opensips.org<br>
<b>Subject:</b> Re: [OpenSIPS-Users] Forwarding REGISTER messages to Asterisk<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>I would highly highly highly recommend you *DO NOT* use Asterisk
if you'll have any sort of volume of registrations (for the record, I am an
Asterisk advocate).<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>I speak from experience and can say first hand that this is a
**bad idea**.<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>Asterisk's sip stack and configuration setup really just don't
allow for a high concentration of SIP registrations without a
very significant penalty on the Asterisk host. I had a couple hundred
registrations on a single Asterisk server that started to visibly effect
the performance of the host server. When I hit 500 registrations, the Asterisk
server was unusable. I have consistently been able to reproduce this
effect to varying degrees on several Linux distros and on four different hardware
vendor platforms.<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>Offloading the registrations to OpenSIPS is the smart way to do
it. If you're wanting to send the call simultaneously to Asterisk and the
end-point you can make a creative OpenSIPS config file using the LCR module or
dbalias or something.<o:p></o:p></span></p>
</div>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<div>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>If you're wanting to do processing on the Asterisk server before
the end-point is called, you could use the opensips config to route all the
calls to the Asterisk server(s) and then have the Asterisk servers do their
processing bit and send the call back to OpenSIPS which could detect the call
as coming from your pool of Asterisk server(s) and do a location lookup at that
point.<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>In any case, be creative, it's quite possible to do almost
anything with the call flows you'd like having OpenSIPS at the front of your network. <o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>Good Luck!<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>_____________________________<o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'>Darren Sessions<o:p></o:p></span></p>
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<div>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><a href="mailto:dmsessions@gmail.com">dmsessions@gmail.com</a><o:p></o:p></span></p>
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<div>
<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><a href="http://www.darrensessions.com">http://www.darrensessions.com</a><o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal><span class=apple-style-span><span style='font-size:9.0pt;
font-family:"Arial","sans-serif";color:black'>_____________________________</span></span><span
style='font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'><o:p></o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";
color:black'><o:p> </o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>On Sep 4, 2008, at 8:58 AM, Simon Shaw wrote:<o:p></o:p></p>
</div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal>I was hoping to solve this issue in a more generic manner.<br>
Any idea how I change the IP address in the contact header?<br>
<br>
-----Original Message-----<br>
From: users-bounces@lists.opensips.org [<a
href="mailto:users-bounces@lists.opensips.org">mailto:users-bounces@lists.opensips.org</a>]
On Behalf Of I?aki Baz Castillo<br>
Sent: Thursday, September 04, 2008 11:33 AM<br>
To: <a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a><br>
Subject: Re: [OpenSIPS-Users] Forwarding REGISTER messages to Asterisk<br>
<br>
El Wednesday 03 September 2008 19:44:42 Simon Shaw escribió:<br>
<br>
<o:p></o:p></p>
<p class=MsoNormal>I would like to use OpenSIPS as a pure router between some
of my SIP UAs<o:p></o:p></p>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>and Asterisk such that when one of these UAs registers with
OpenSIPS the<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>REGISTER message will be redirected to Asterisk where it
will be<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>handled. When a UA that is connected directly to
the Asterisk box<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>dials one of the OpenSIPS UAs I would like the SIP INVITE to
be sent<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>from the Asterisk box via OpenSIPS.<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal><o:p> </o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>I have managed to forward the REGISTER messages to Asterisk
using<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>rewritehost(<IPofAsterisk>), however when a UA that is
registered<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>directly with the Asterisk dials a UA that registered via
OpenSIPS the<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>INVITE message is sent directly from Asterisk to the UA.<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal><o:p> </o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>I guess this is happening because Contact header in the
REGISTER message<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>has the IP of the UA and not that of OpenSIPS.<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal><o:p> </o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>If this assumption is correct what is the correct method to
change the<o:p></o:p></p>
</blockquote>
<blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal>IP in the contact header? Could the NATHelper be
useful here?<o:p></o:p></p>
</blockquote>
<p class=MsoNormal><br>
It's easy: if you forward the REGISTER from OpenSIPS to Asterisk then when <br>
Asterisk calls that user it will do it directly, since of course
"Contact" in <br>
REGISTER has the UA IP as it's logical.<br>
<br>
This would be possible with "Path" feature (see "path" module
and related <br>
RFC), but of course, be sure that Asterisk doesn't implement it.<br>
<br>
<br>
<br>
<br>
<br>
-- <br>
Iñaki Baz Castillo<br>
<a href="mailto:ibc@in.ilimit.es">ibc@in.ilimit.es</a><br>
<br>
_______________________________________________<br>
Users mailing list<br>
Users@lists.opensips.org<br>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br>
_______________________________________________<br>
Users mailing list<br>
Users@lists.opensips.org<br>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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