I'm pretty sure he's wrong.<div><br></div><div>From the RFC:</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">
<font face="arial, helvetica, sans-serif">The connection (`c=') and attribute (`a=') information in the<br></font><font face="arial, helvetica, sans-serif"> session-level section applies to all the media of that session unless<br>
</font><font face="arial, helvetica, sans-serif"> overridden by connection information or an attribute of the same name<br></font><font face="arial, helvetica, sans-serif"> in the media description.</font><font face="arial, helvetica, sans-serif"><br>
</font><font face="arial, helvetica, sans-serif">
</font><pre style="word-wrap:break-word;white-space:pre-wrap"><font face="arial, helvetica, sans-serif">The media
description starts with an `m=' line and continues to the next media
description or end of the whole session description. In general,
session-level values are the default for all media unless overridden
by an equivalent media-level value.</font></pre></blockquote><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><a href="http://www.ietf.org/rfc/rfc2327.txt">http://www.ietf.org/rfc/rfc2327.txt</a> </div>
<div><br></div><div>And here's a link to the Media Description definition:</div><div><br></div><div><a href="http://tools.ietf.org/html/rfc4566#page-22">http://tools.ietf.org/html/rfc4566#page-22</a>
</div><div><br></div><div>Here's a Stackoverflow discussion on the m= field, and it confirms that the trailing 101 is the telehony event for DTMF:</div><div><br></div><div><a href="http://stackoverflow.com/questions/2930288/sdp-media-field-format">http://stackoverflow.com/questions/2930288/sdp-media-field-format</a>
</div><div><br></div><div>Since you are a partner, I will look for a recent VoE session which also talk about the same thing...looking now.</div><div><br></div><div>-Anthony</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 class="HOEnZb"><div class="h5"><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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