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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Moataz,<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'>It seems you are running 12.4(x) or 12.3T, either of which
supports sip-ua, so you shouldn't have any issues. You may have to watch
out for how the SIP trunk provider is sending the REFER to you though.
3.3 may not be intelligent enough to recognize that message, our voice gateways
were not and we had to create a small TCL script to strip out the To: address
and readjust it to the dest for the leg_incoming portion of the call. You
might be better off just upgrading to 4.1 or 4.2.<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'>-ryan<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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<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"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Ryan West<br>
<b>Sent:</b> Sunday, March 23, 2008 4:15 PM<br>
<b>To:</b> Moataz Mamdouh; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] SIP trunk With CME<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif"'>Go to the feature selector and see if the
version of code supports sip-ua. I am not in front of a computer so you'll have
to verify yourself. I am pretty sure it is supported those.<o:p></o:p></span></p>
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style='font-size:10.0pt;font-family:"Arial","sans-serif"'>
<hr size=2 width="100%" align=center>
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<p class=MsoNormal style='margin-bottom:12.0pt'><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"'>Moataz Mamdouh
<moataz_mmdh@yahoo.com></span><span style='font-size:10.0pt;font-family:
"Arial","sans-serif"'><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Sent:
</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Sunday,
March 23, 2008 12:55 PM</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>To: </span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>cisco-voip@puck.nether.net
<cisco-voip@puck.nether.net></span><span style='font-size:10.0pt;
font-family:"Arial","sans-serif"'><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Subject:
</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Re:
[cisco-voip] SIP trunk With CME</span><o:p></o:p></p>
<div>
<p class=MsoNormal>but this is based on CME ver 4.1 , is not supported on ver
3.3 ?<br>
<br>
<b><i>Moataz Mamdouh <moataz_mmdh@yahoo.com></i></b> wrote: <o:p></o:p></p>
<p class=MsoNormal>thanks , it's a good guide<br>
<br>
<b><i>Michael Wilson <mikejww@googlemail.com></i></b> wrote: <o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>HI there,<br>
<br>
You might want to look at the following for setting up a sip trunck with CME.<br>
<br>
<a
href="http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml">http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml</a><br>
<br>
Hope this helps.<o:p></o:p></p>
<div>
<p class=MsoNormal><span class=gmailquote>On 23/03/2008, <b>Moataz Mamdouh</b>
<<a href="mailto:moataz_mmdh@yahoo.com">moataz_mmdh@yahoo.com</a>> wrote:</span>
<o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>dear all<br>
how to male a SIP trunk between a third party call processing system based on
SIP<br>
and cisco call manager express ver 3.3a ,<br>
any ideas?<br>
<br>
<span class=ad><o:p></o:p></span></p>
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