Small trick I learned a few years back: remove the "partner" from the URL so non-partners can view it:<div><br></div><div><a href="http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html">http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</a></div>
<div><br></div><div>-Anthony<br><br><div class="gmail_quote">On Thu, May 17, 2012 at 1:07 PM, miken miken <span dir="ltr"><<a href="mailto:miken@sisna.com" target="_blank">miken@sisna.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>Configuration examples and explanations for all of the primary North America SIP providers can be found on this link. You need partner access to view it.</div>
<div> </div>
<div><a href="http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html" target="_blank">http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html</a></div>
<div> </div>
<div>Thank you</div><span class="HOEnZb"><font color="#888888">
<div>MikeN<br><br></div>
</font></span><div class="gmail_quote"><div><div class="h5">On Thu, May 17, 2012 at 11:01 AM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br>
</div></div><blockquote style="BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PADDING-LEFT:1ex" class="gmail_quote"><div><div class="h5"><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></font></span><br></div></div><div class="im">_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
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