That fixed it!<div><br></div><div>Yay.... </div><div><br></div><div>Now, will faxing still work...</div><div><br><div class="gmail_quote">On Thu, May 17, 2012 at 2:15 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">They're actually right.<br><br>Remove this:<br><font face="arial, helvetica, sans-serif"><div class="im">modem passthrough nse payload-type 101 codec g711ulaw<br>
<br></div>Or change it to something similar like:<br></font><font face="arial, helvetica, sans-serif">modem passthrough nse payload-type 100 codec g711ulaw</font><span class="HOEnZb"><font color="#888888"><br>
<br>-nick</font></span><div class="HOEnZb"><div class="h5"><br><br><div class="gmail_quote">On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF setting in CUCM on the trunk? And lastly, your MTP Required check box setting on the trunk?<span><font color="#888888"><div>
<br></div></font></span><div><span><font color="#888888">-Anthony</font></span><div><div><br><br><div class="gmail_quote">
On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Even that example shows:<div><br></div><div>a=rtpmap:101 telephone-event/8000</div><div><br></div><div>Whereas I am seeing</div><div><br></div><div><span>a=rtpmap:101 X-NSE/8000 </span> </div><div><br></div><div><br>
</div><div>A: Why?</div><div>B: How do I change it?</div><div><div><div><br><div class="gmail_quote">On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Here is the VoE I was talking about:<div><br></div><div><a href="https://communities.cisco.com/docs/DOC-7823" target="_blank">https://communities.cisco.com/docs/DOC-7823</a> </div>
<div><br></div><div>Look towards the top for "VoE - CUBE SIP Trunking"</div>
<div><br></div><div>Then download the PDF, and goto page 90. The page is also discuss in the Webex recording @ 1h 48m 55s. For those who cannot see this, it says:</div><div><br></div><div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
“c” parameter identifies the IP<br>address (20.1.1.1) that the peer<br>device should send the media to<br></blockquote><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
“m” parameter identifies:<br>the type of call (audio)<br>port number for media (16950)<br>payload type for the 1st<br>preferred codec (18 for G729)<br>dtmf (101 for RFC2833)<br></blockquote><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
“a’” parameter identifies all the<br>codecs and other descriptors for this<br>call leg</blockquote><div><br></div><div>This VoE event is very informative. Hope that helps.</div><span><font color="#888888"><div>
<br></div><div>-Anthony</div></font></span><div><div><br><div class="gmail_quote">
On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
It is not.<div><br></div><div>Per Verizon tech:</div><div><br></div><div><p class="MsoNormal"><span style="font-size:10.0pt"> Octet1058 SIP Message Body: SDP <br>
--------------------------------------------------------------------------------<br>
........ Header
Field v=0<br>
........
o=CiscoSystemsSIP-GW-UserAgent 794 632 IN IP4 1,1,1,1<br>
........
s=SIP Call<br>
........
c=IN IP4 1.1.1.1<br>
........
t=0 0<br>
........
m=audio 17176 RTP/AVP 0 101<br>
........
c=IN IP4 1.1.1.1<br>
........
a=rtpmap:0 PCMU/8000<br>
........
a=rtpmap:101 X-NSE/8000 <-- should be telephone-event/8000<br>
........
a=fmtp:101 192-194<br>
........
a=ptime:20</span></p></div><div><br></div><div>They say the problem is on our end, and since we are sending the wrong DTMF, they are closing their ticket.</div><div><div><div><br></div><div><br>
</div><div><br><br><div class="gmail_quote">
On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I'm glad you posted that.<div><br></div><div>The m= is the actual setting for that call. The a= are the available settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101 (telephony).</div><div><br>
</div>
<div>This looks correct.</div><span><font color="#888888"><div><br></div></font></span><div><span><font color="#888888">-Anthony</font></span><div><div><br><br><div class="gmail_quote">
On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Added it, no change.<div><br></div><div><div><br></div><div>v=0</div><div>o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 <a href="tel:157.130.97.178" value="+15713097178" target="_blank">157.130.97.178</a></div>
<div>s=SIP Call</div><div>c=IN IP4 1.1.1.1</div><div>t=0 0</div><div>m=audio 18130 RTP/AVP 0 101</div>
<div>c=IN IP4 <a href="tel:157.130.97.178" value="+15713097178" target="_blank">157.130.97.178</a></div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 X-NSE/8000 <------------- this needs to be <span>telephone-event/8000</span></div>
<div>a=fmtp:101 192-194</div><div>
a=ptime:20</div></div><div><div><div><br></div><div><br></div><div><br><br><div class="gmail_quote">On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>I see you are setting EO = Forced on the CUBE, which the telco requires, but are you using EO on the SIP trunk form CUCM to the CUBE? What is your DTMF Signaling Method set to on that Trunk?</div>
<div><br></div><div>
The only command I run which I can see is missing from your config is:</div><div><br></div><div>voice service voip</div><div> dtmf-interworking rtp-nte</div><div><br></div><div>But I'm not positive that's your problem.</div>
<div><br></div><div>-Anthony</div><div><br><div class="gmail_quote"><div><div>On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><font face="arial, helvetica, sans-serif">We have a SIP trunk to Verizon, Long Distance, Local and international work fine, however, for toll free calls, DTMF does not function.</font><div>
<font face="arial, helvetica, sans-serif"><br>
</font></div><div><font face="arial, helvetica, sans-serif">We are set to send RTP-NTE, but Verizon is saying that we are sending this:</font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">a=rtpmap:101 X-NSE/8000 </font></div>
<div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">And it should be:</font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">telephone-event/8000</font></div>
<div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">And that is why it is failing.</font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div>
<font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">What configuration change can we do to force it to send the right DTMF method?</font></div>
<div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request), there is a software MTP and Transcoder on the router (both in use)... Verizon says it is not their problem and closed their ticket.</font></div>
<div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif">Relevant SIP Config:</font></div><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif"><div>
<br></div><div>!</div><div>voice call send-alert</div><div>voice rtp send-recv</div><div>!</div><div>voice service voip </div><div> allow-connections h323 to h323</div><div> allow-connections h323 to sip</div><div> allow-connections sip to h323</div>
<div> allow-connections sip to sip</div><div> no supplementary-service sip refer</div><div> redirect ip2ip</div><div> h323</div><div> h225 display-ie ccm-compatible</div><div> modem passthrough nse payload-type 101 codec g711ulaw</div>
<div> sip</div><div> bind media source-interface MFR1</div><div> early-offer forced</div><div> midcall-signaling passthru</div><div>!</div><div>!</div></font></div><div><font face="arial, helvetica, sans-serif"><div><br>
</div><div>dial-peer voice 800 voip</div><div> description OUTBOUND Voice SIP calls to VzB</div><div> destination-pattern 1800[2-9]......</div><div> voice-class sip dtmf-relay force rtp-nte</div><div> session protocol sipv2</div>
<div> session target sip-server</div><div> incoming called-number .</div><div> dtmf-relay rtp-nte</div><div> codec g711ulaw</div><div> no vad</div></font></div><div><font face="arial, helvetica, sans-serif"><br></font></div>
<div><font face="arial, helvetica, sans-serif"><div><br></div><div>!</div><div>sip-ua </div><div> retry invite 2</div><div> retry bye 2</div><div> retry cancel 2</div><div> registrar dns:verizonsipgateway expires 3600</div>
<div> sip-server
dns:verizonsipgateway:5071</div><div> g729-annexb override</div><div>!</div></font></div><span><font color="#888888"><div><font face="arial, helvetica, sans-serif"><br></font></div><div><font face="arial, helvetica, sans-serif"><br>
</font></div><div>
<font face="arial, helvetica, sans-serif">Jonathan</font></div>
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