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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Drew,<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'>Can you post a ‘deb ccsip messages’ output and also a sanitized version
of your sip-ua section?<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>
<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"'> Drew Weaver
[mailto:drew.weaver@thenap.com] <br>
<b>Sent:</b> Monday, October 13, 2008 9:04 AM<br>
<b>To:</b> 'cisco-voip@puck.nether.net'<br>
<b>Subject:</b> RE: [cisco-voip] Very new to cisco VoIP products have a few
questions/problem.<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Outbound dial-peer:<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'>dial-peer voice 1001 voip<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> description ** Outgoing call to SIP trunk (Generic SIP
Trunk Provider) **<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> translation-profile outgoing PSTN_Outgoing<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> destination-pattern 9T<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> voice-class codec 1<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> voice-class sip dtmf-relay force rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> session protocol sipv2<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> session target sip-server<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> dtmf-relay rtp-nte<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> ip qos dscp cs5 media<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> ip qos dscp cs4 signaling<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> no vad<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'>voice translation-profile PSTN_Outgoing<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> translate calling 1111<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> translate called 1112<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> translate redirect-target 410<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> translate redirect-called 410<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'>voice class codec 1<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> codec preference 1 g711ulaw<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>
<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> Monday, October 13, 2008 8:47 AM<br>
<b>To:</b> Ed Leatherman; Drew Weaver<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> RE: [cisco-voip] Very new to cisco VoIP products have a few
questions/problem.<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>To build on what Ed has already added. When you talk with
your SIP provider, ask them what type of authentication they are
expecting. Some providers will do source IP with no authentication,
others support a username and password on dial-peers, and others require the
hidden ‘credential username password realm’ under sip-ua. Try running
‘show sip-ua reg stat’ and see what shows as registered. A lot of
providers won’t like the extra registrations that CME / voice gateways send, so
if you see 1 registration with a yes and a bunch that say no, you can trim all
those down with ‘no sip-register’ under each of the offending dial-peers.<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>
<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>Ed Leatherman<br>
<b>Sent:</b> Monday, October 13, 2008 8:37 AM<br>
<b>To:</b> Drew Weaver<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Very new to cisco VoIP products have a few
questions/problem.<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>Sounds like the SIP messages
are going back and forth OK but your audio stream isn't. Perhaps ACL or firewall
causing the problem? <br>
<br>
We've got a UC520 setup at a small office across town, but no SIP trunk so i'm
not sure what exactly would be the cause.<br>
<br>
CME is running as part of the IOS on the box, if it weren't running your phones
wouldn't work at all. I would double check the dialpeers that CCA generated for
you. From my limited experience CCA was OK for a basic configuration but as
soon as we had to setup something special I got fed up with it and just started
using the CLI instead.<br>
<br>
Barring that you might want to check with your SIP provider to see if they can
help with the outbound call problems.. maybe the call is getting rejected by
them?<br>
<br>
Ed<o:p></o:p></p>
<div>
<p class=MsoNormal>On Sun, Oct 12, 2008 at 9:04 PM, Drew Weaver <<a
href="mailto:drew.weaver@thenap.com">drew.weaver@thenap.com</a>> wrote:<o:p></o:p></p>
<div>
<div>
<p>
Hi there,<o:p></o:p></p>
<p> <o:p></o:p></p>
</div>
</div>
</div>
<p class=MsoNormal><br>
-- <br>
Ed Leatherman<br>
Assistant Director, Voice Services<br>
West Virginia University<br>
Telecommunications and Network Operations<o:p></o:p></p>
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