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<p class=MsoNormal>Ryan,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>There’s no required authentication for the provider
(username/password). The show command just shows empty.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Thanks everyone for the input so far in case I forgot to say
that!<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Mike Wilusz, CCNA<o:p></o:p></p>
<p class=MsoNormal>Networking Systems Programmer<o:p></o:p></p>
<p class=MsoNormal>Price Chopper Supermarkets / The Golub Corporation<o:p></o:p></p>
</div>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Ryan West
[mailto:rwest@zyedge.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 4:17 PM<br>
<b>To:</b> Wilusz, Mike; Matt Slaga (US); cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Are you required to register
your sip-ua? If so, please also post ‘show sip-ua reg stat’<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>-ryan<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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>Wilusz, Mike<br>
<b>Sent:</b> Thursday, October 02, 2008 4:11 PM<br>
<b>To:</b> Matt Slaga (US); cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Matt,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Tried that and I still get the same result. The voice
config from the CUBE is this:<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>voice service voip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> address-hiding<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> allow-connections
h323 to h323<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> allow-connections
h323 to sip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> allow-connections
sip to h323<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> allow-connections
sip to sip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>dial-peer voice 2
voip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> destination-pattern
1..........<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session
protocol sipv2<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session target
ipv4:64.206.114.6 (Carrier Gateway)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> dtmf-relay
rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> codec g711ulaw<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>dial-peer voice 3
voip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> destination-pattern
3474179896<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session
protocol sipv2<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session target
ipv4:10.50.120.33 (Our CUCM)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> dtmf-relay
rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> codec g711ulaw<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>!<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>sip-ua<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> registrar
ipv4:64.206.114.6 expires 3600<o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Mike Wilusz, CCNA<o:p></o:p></p>
<p class=MsoNormal>Networking Systems Programmer<o:p></o:p></p>
<p class=MsoNormal>Price Chopper Supermarkets / The Golub Corporation<o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Matt Slaga (US)
[mailto:Matt.Slaga@us.didata.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 4:06 PM<br>
<b>To:</b> Wilusz, Mike; cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Try adding this to your config:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>voice class codec 1<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> codec preference 1
g711ulaw<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> codec preference 2 g729br8<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> codec preference 3 g729r8<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>And to your voip dial peers:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>voice-class codec 1<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Wilusz, Mike
[mailto:mikewilusz@pricechopper.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 3:42 PM<br>
<b>To:</b> Matt Slaga (US); cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Yup… set here in this dial-peer. Which is matched
properly according to the debug.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>dial-peer voice 2
voip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> destination-pattern
1..........<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session
protocol sipv2<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> session target
ipv4:64.206.114.6<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> dtmf-relay
rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'> codec g711ulaw<o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Mike Wilusz, CCNA<o:p></o:p></p>
<p class=MsoNormal>Networking Systems Programmer<o:p></o:p></p>
<p class=MsoNormal>Price Chopper Supermarkets / The Golub Corporation<o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Matt Slaga (US)
[mailto:Matt.Slaga@us.didata.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 3:40 PM<br>
<b>To:</b> Wilusz, Mike; cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Where are you setting the
codec? The line below says you are not using VCC, are your dial peers
specifying a codec?<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>Oct 2
10:21:18: //10/36ECCAF0801A/SIP/Info/sipSPIGetCallConfig: Not using Voice Class
Codec<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Wilusz, Mike
[mailto:mikewilusz@pricechopper.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 3:35 PM<br>
<b>To:</b> Matt Slaga (US); cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Matt,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Not sure what means. I see g726r32 is selected as a
codec…but why? I don’t have the set as my codec on the dial-peer and the
carrier is set (or supposed to be) to the same as me. I guess I’m not
really sure what that line is telling me.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Mike Wilusz, CCNA<o:p></o:p></p>
<p class=MsoNormal>Networking Systems Programmer<o:p></o:p></p>
<p class=MsoNormal>Price Chopper Supermarkets / The Golub Corporation<o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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"'> Matt Slaga (US)
[mailto:Matt.Slaga@us.didata.com] <br>
<b>Sent:</b> Thursday, October 02, 2008 3:14 PM<br>
<b>To:</b> Wilusz, Mike; cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Probably this:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-family:"Courier New"'>Oct 2
10:21:18: //9/36ECCAF0801A/SIP/Info/sipSPISelectDynamicPayload: Selecting
Dynamic Payload : 96 for Codec: g726r32<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<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>Wilusz, Mike<br>
<b>Sent:</b> Thursday, October 02, 2008 10:31 AM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] Outbound Dial with Carrier SIP Trunk<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hey All,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>We’re in the process of trialing a SIP trunk solution from
our provider, PAETEC. I followed the Cisco docs for configuring a CUBE,
but we can only receive calls and not place them. The setup is CUCM
6.1(2) -> SIP Trunk -> CUBE (2811) -> SIP Trunk -> PAETEC SIP
Gateway. I’m very new to SIP and am trying to analyze the output from <span
style='font-family:"Courier New"'>debug ccsip all</span>. Nothing is
jumping out at me as the exact reason why the calls outbound from our CUCM
aren’t working. Anybody have any ideas?<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p><strong> </strong><o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'>
<hr size=1 width="100%" align=center>
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