<div>I feel embarasssed I been blaming a telco all day when it was my own ignorance that made it not work</div>
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<div>I was not matching the inbound dialpeer for my outbound calling. </div>
<div> </div>
<div>I added the following dial peer and the problem was solved</div>
<div> </div>
<div>dial-peer voice 150 voip<br> description incoming outbound<br> preference 1<br> voice-class h323 50<br> incoming called-number .T<br> dtmf-relay h245-alphanumeric<br> codec g711ulaw<br> no vad<br>!</div>
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<div>Thanks alot Nick and Ryan for all your help on this!</div>
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<div>Dane</div>
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<div class="gmail_quote">On Tue, Oct 27, 2009 at 8:49 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">So you have a H323-SIP CUBE, and your DTMF isn't working. This is<br>probably the most common problem with CUBE users.<br>
<br>For this #1 problem, the number one cause is 'incoming called-number .'<br><br>Most people don't really understand inbound dial peer matching, and<br>have used this for ages on normal TDM gateways that were<br>
single-protocol. The best way to fix this is to read the 'Understand<br>Incoming and Outgoing Dial-Peers" document on Cisco.com, and figuring<br>out the best way to match dial peers for both your incoming/outgoing<br>
SIP/H323 legs. You can prefix digits and match on incoming called<br>number, or ditch incoming called-numbers completely and use<br>answer-address.<br><br>I like using 'debug voip ccapi inout' to determine this. You can do a<br>
search for peer= after you've got the debug to find out which dial<br>peers you're hitting for each case, plus what the numbers look like<br>after translations, etc. 'debug voip dialpeer' is an alternative, but<br>
I personally find it more confusing.<br><br>For h323-SIP your dial peers should look something like this:<br><br>incoming h323 dial peer for outgoing call: dtmf-relay h245-alpha or h245-signal<br>outgoing sip dial peer for outgoing call: dtmf-relay rtp-nte<br>
digit-drop (plus any payload commands required)<br>incoming sip dial peer for incoming call: same as sip option above<br>outgoing h323 dial peer for incoming call: same as h323 option above<br><br>My best guess is that if you look at your incoming/outgoing dial peers<br>
something isn't matched correctly.<br><font color="#888888"><br>-nick<br></font>
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<div class="h5"><br>On Tue, Oct 27, 2009 at 8:17 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br>> I have a cisco 7975 phone connected to a cucm 7.x --> h323 gateway cisco<br>
> 2821 --> ITSP sip trunk<br>><br>> I am using the CUBE feature on the gateway...DTMF works calling internally<br>> to my cisco unity connection voice mail so it is able to be sent.<br>><br>> Does anyone have any ideas how I could go about troubleshooting this?<br>
><br>> Dane<br>><br>> On Tue, Oct 27, 2009 at 8:14 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> wrote:<br>>><br>>> Yes, as long as your debugs are setup correctly (they show output).<br>
>><br>>> -nick<br>>><br>>> On Tue, Oct 27, 2009 at 7:23 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>><br>>> wrote:<br>>> > Thanks for the reply Nick<br>
>> ><br>>> > I debugged voip rtp named-event and when I tried to hit 1 in the call<br>>> > for<br>>> > dtmf nothing came out of the debug. Could this possibly mean on my side<br>>> > Im<br>
>> > not sending dtmf to the service provider?<br>>> > Dane<br>>> ><br>>> > On Tue, Oct 27, 2009 at 4:30 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>><br>
>> > wrote:<br>>> >><br>>> >> That shows up in the debugs in working scenarios too. Not sure what<br>>> >> the importance of those statements are, but it's the type of thing you<br>
>> >> see when you add 'all' to a debug.<br>>> >><br>>> >> It's not the 183 you want to look at, but the 200 OK with the CSeq of<br>>> >> your INVITE. And you want a 200 OK. I've seen it where the debugs<br>
>> >> will show that we're sending DTMF but the provider won't use it, which<br>>> >> is a conversation you would need to have with the provider.<br>>> >><br>>> >> -nick<br>
>> >><br>>> >> On Tue, Oct 27, 2009 at 3:45 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>><br>>> >> wrote:<br>>> >> > Hmm that does not sound good<br>
>> >> ><br>>> >> > This is with the default settings<br>>> >> ><br>>> >> > rtp payload-type nte 101<br>>> >> > rtp payload-type nse 100<br>>> >> ><br>
>> >> > which don't show up in the config. Could there be any reason why the<br>>> >> > router<br>>> >> > is not able to use 101 below are my dial peers<br>>> >> ><br>
>> >> > dial-peer voice 100 voip<br>>> >> > description AA Publisher<br>>> >> > preference 1<br>>> >> > destination-pattern 1..<br>>> >> > voice-class h323 50<br>
>> >> > session target ipv4:10.1.80.10<br>>> >> > dtmf-relay h245-alphanumeric<br>>> >> > codec g711ulaw<br>>> >> > no vad<br>>> >> > !<br>>> >> > dial-peer voice 1000 voip<br>
>> >> > description incoming Call<br>>> >> > translation-profile incoming aa<br>>> >> > preference 1<br>>> >> ><br>>> >> > incoming called-number 6784442454<br>
>> >> ><br>>> >> > dtmf-relay rtp-nte<br>>> >> > codec g711ulaw<br>>> >> > ip qos dscp cs5 media<br>>> >> > ip qos dscp cs5 signaling<br>>> >> > no vad<br>
>> >> > !<br>>> >> > dial-peer voice 101 voip<br>>> >> > description AA Subscriber<br>>> >> > preference 2<br>>> >> > destination-pattern 1..<br>
>> >> > voice-class h323 50<br>>> >> > session target ipv4:10.1.80.11<br>>> >> > dtmf-relay h245-alphanumeric<br>>> >> > codec g711ulaw<br>>> >> > no vad<br>
>> >> > !<br>>> >> > dial-peer voice 2000 voip<br>>> >> > description outbound<br>>> >> > translation-profile outgoing addone<br>>> >> > preference 1<br>
>> >> > destination-pattern .T<br>>> >> ><br>>> >> > progress_ind setup enable 3<br>>> >> > progress_ind progress enable 8<br>>> >> > session protocol sipv2<br>
>> >> > session target dns:<a href="http://did.voip.les.net/" target="_blank">did.voip.les.net</a><br>>> >> ><br>>> >> > dtmf-relay rtp-nte<br>>> >> > codec g711ulaw<br>
>> >> ><br>>> >> > !<br>>> ><br>>> ><br>><br>><br></div></div></blockquote></div><br>