<div dir="ltr">That depends on the IOS version...<br><br>Try it...<br><br>voice service voip<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> allow-connections h323 to h323<br>
<br><br><br><br>Jonathan<br><br><br><br><div class="gmail_quote">On Fri, Aug 8, 2008 at 2:31 PM, Alessandro Bertacco <span dir="ltr"><<a href="mailto:alessandro.bertacco@alice.it">alessandro.bertacco@alice.it</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="white" link="blue" vlink="purple" lang="IT">
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<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Hi all and thank you for your answers!</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">So I'll need an IP2IP GTW because the Sip Provider give me
username and password for the trunk authentication.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">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?</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Thank you all.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Regards</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Alessandro Bertacco</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;">Da:</span></b><span style="font-size: 10pt;"> Philip Walenta
[mailto:<a href="mailto:pwalenta@wi.rr.com" target="_blank">pwalenta@wi.rr.com</a>] <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</span></p>
</div>
</div>
<p> </p>
<p><span style="font-size: 10pt; color: blue;" lang="EN-US">I thought the SIP in 6.1 supported digest auth on trunks?</span><span lang="EN-US"></span></p>
<p><span lang="EN-US"> </span></p>
<div style="text-align: center;" align="center"><span lang="EN-US">
<hr size="2" width="100%" align="center">
</span></div>
<p style="margin-bottom: 12pt;"><b><span style="font-size: 10pt;" lang="EN-US">From:</span></b><span style="font-size: 10pt;" lang="EN-US">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <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"></span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">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.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;" lang="EN-US">From:</span></b><span style="font-size: 10pt;" lang="EN-US"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>
[mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <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</span></p>
</div>
</div>
<p><span lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Hi all,</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> I'll need to configure a SIP trunk from a CUCM6.1 to an
external provider, to route incoming call to a specified Phone.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Someone have already configured CUCM6.1 to do these? I need an
Ip to Ip Gateway or not?</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Can I use the Voice Gateway instead of IP to IP dedicated
Router?</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Thank you for your response</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Best Regards</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US">Alessandro Bertacco</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);" lang="EN-US"> </span></p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;">Da:</span></b><span style="font-size: 10pt;">
<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <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?</span></p>
</div>
</div>
<p> </p>
<div>
<p><span style="font-size: 10pt;">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></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Any
pointers would be great.</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Thanks,</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">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></p>
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