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<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi all and thank you for your answers!<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>So I’ll need an IP2IP GTW because the Sip Provider give me
username and password for the trunk authentication.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Now, the question is, can I add to the existing voice
Gateway/SRST router the rule of an Ip2IP gateway, or I’ll need to purchase
another router?<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thank you all.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Alessandro Bertacco<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<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:"Segoe UI","sans-serif"'>Da:</span></b><span
style='font-size:10.0pt;font-family:"Segoe UI","sans-serif"'> Philip Walenta
[mailto:pwalenta@wi.rr.com] <br>
<b>Inviato:</b> mercoledì 6 agosto 2008 17.06<br>
<b>A:</b> 'Matt Slaga (US)'; 'Alessandro Bertacco'; 'Cisco Voyp List'<br>
<b>Oggetto:</b> RE: [cisco-voip] CUCM 6.1 SIP Trunk to Provider<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span lang=EN-US 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><span
lang=EN-US><o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<div class=MsoNormal align=center style='text-align:center'><span lang=EN-US>
<hr size=2 width="100%" align=center>
</span></div>
<p class=MsoNormal style='margin-bottom:12.0pt'><b><span lang=EN-US
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
lang=EN-US 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><span
lang=EN-US><o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It depends. Does the provider give you user/pass
credentials? If so, UCM cannot do this, you will need IP2IP gateway
inbetween. 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 lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US 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>
</div>
</div>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi all,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> I’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 lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US 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 lang=EN-US 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 lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US 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 lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Best Regards<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Alessandro Bertacco<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<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:"Segoe UI","sans-serif"'>Da:</span></b><span
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> giovedì 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>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal><span 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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Any
pointers would be great.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks,</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span 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: "While it is the same length, this 7'
crossover cable<br>
is not regulation issue for most competitive Manhattan double dutch leagues.</span><o:p></o:p></p>
</div>
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<hr size=1 width="100%" align=center>
</span></div>
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