<div>I have a cisco 7975 phone connected to a cucm 7.x --> h323 gateway cisco 2821 --> ITSP sip trunk</div>
<div> </div>
<div>I am using the CUBE feature on the gateway...DTMF works calling internally to my cisco unity connection voice mail so it is able to be sent. </div>
<div> </div>
<div>Does anyone have any ideas how I could go about troubleshooting this?</div>
<div> </div>
<div>Dane<br><br></div>
<div class="gmail_quote">On Tue, Oct 27, 2009 at 8:14 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">Yes, as long as your debugs are setup correctly (they show output).<br><font color="#888888"><br>-nick<br>
</font>
<div>
<div></div>
<div class="h5"><br>On Tue, Oct 27, 2009 at 7:23 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> 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 for<br>
> dtmf nothing came out of the debug. Could this possibly mean on my side 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>> 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></div></div></blockquote></div><br>