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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It does, until you look under the covers.  It will allow you to
make outbound calls using these credentials, but inbound calls fail because UCM
never syncs and maintains presence details with the sip provider, so the
provider always sees your line as offline.  <o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Cisco is not very interested in fixing this because of the miniscule
amount of customers looking for this particular configuration.  Plus, you are
limited to one set of credentials and most small SIP providers will give you
one set of credentials per line (i.e. viatalk and the like).<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Most corporate SIP providers don&#8217;t use the auth
credentials.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</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"'> Philip Walenta
[mailto:pwalenta@wi.rr.com] <br>
<b>Sent:</b> Wednesday, August 06, 2008 11:06 AM<br>
<b>To:</b> Matt Slaga (US); 'Alessandro Bertacco'; 'Cisco Voyp List'<br>
<b>Subject:</b> RE: [cisco-voip] CUCM 6.1 SIP Trunk to Provider<o:p></o:p></span></p>

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<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>I thought the SIP in 6.1 supported digest auth on trunks?</span><o:p></o:p></p>

<p class=MsoNormal><o:p>&nbsp;</o:p></p>

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<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"'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Matt Slaga (US)<br>
<b>Sent:</b> Wednesday, August 06, 2008 9:33 AM<br>
<b>To:</b> Alessandro Bertacco; Cisco Voyp List<br>
<b>Subject:</b> Re: [cisco-voip] CUCM 6.1 SIP Trunk to Provider</span><o:p></o:p></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It depends.&nbsp; Does the provider give you user/pass
credentials?&nbsp; If so, UCM cannot do this, you will need IP2IP gateway
inbetween.&nbsp; If they are giving you a basic SIP trunk with only IP as
creds, then most likely you can terminate directly to UCM if they use a
compatible dtmf relay.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</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>Alessandro Bertacco<br>
<b>Sent:</b> Wednesday, August 06, 2008 9:44 AM<br>
<b>To:</b> 'Cisco Voyp List'<br>
<b>Subject:</b> [cisco-voip] CUCM 6.1 SIP Trunk to Provider<o:p></o:p></span></p>

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<p class=MsoNormal><o:p>&nbsp;</o:p></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi all,<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp; I&#8217;ll need to configure a SIP trunk from a CUCM6.1
to an external provider, to route incoming call to a specified Phone.<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Someone have already configured CUCM6.1 to do these? I need an
Ip to Ip Gateway or not?<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Can I use the Voice Gateway instead of IP to IP dedicated
Router?<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thank you for your response<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Best Regards<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Alessandro Bertacco<o:p></o:p></span></p>

<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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<p class=MsoNormal><b><span lang=IT style='font-size:10.0pt;font-family:"Segoe UI","sans-serif"'>Da:</span></b><span
lang=IT style='font-size:10.0pt;font-family:"Segoe UI","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>Per
conto di </b>Lelio Fulgenzi<br>
<b>Inviato:</b> giovedě 31 luglio 2008 20.37<br>
<b>A:</b> Cisco Voyp List<br>
<b>Oggetto:</b> [cisco-voip] Setting up a SIP trunk...anything special?<o:p></o:p></span></p>

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<p class=MsoNormal><span lang=IT><o:p>&nbsp;</o:p></span></p>

<div>

<p class=MsoNormal><span lang=IT style='font-size:10.0pt;font-family:"Arial","sans-serif"'>I
am setting up a SIP trunk to a Cistera appliance (I believe they are using
asterisk) in order for the box to dial out. The SIP trunk setup went fine and
without incident. Is there anything else I need to do? Someone mentioned to me
a while back that I would need something else like a transcoder.</span><span
lang=IT><o:p></o:p></span></p>

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<div>

<p class=MsoNormal><span lang=IT>&nbsp;<o:p></o:p></span></p>

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<div>

<p class=MsoNormal><span lang=IT style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Any
pointers would be great.</span><span lang=IT><o:p></o:p></span></p>

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<p class=MsoNormal><span lang=IT>&nbsp;<o:p></o:p></span></p>

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<p class=MsoNormal><span lang=IT style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,</span><span
lang=IT><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=IT>&nbsp;<o:p></o:p></span></p>

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<p class=MsoNormal><span lang=IT style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Lelio<br>
--------------------------------------------------------------------------------<br>
Lelio Fulgenzi, B.A.<br>
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ <br>
...seen on a Geek Squad patch cord: &quot;While it is the same length, this 7'
crossover cable<br>
is not regulation issue for most competitive Manhattan double dutch leagues.</span><span
lang=IT><o:p></o:p></span></p>

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