[cisco-voip] dtmf from cucm to 2821 cube to sip trunk

[Dane Newman]

I have a cisco 7975 phone connected to a cucm 7.x --> h323 gateway cisco
2821 --> ITSP sip trunk

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.

Does anyone have any ideas how I could go about troubleshooting this?

Dane

On Tue, Oct 27, 2009 at 8:14 PM, Nick Matthews <matthnick at gmail.com> wrote:

> Yes, as long as your debugs are setup correctly (they show output).
>
> -nick
>
> On Tue, Oct 27, 2009 at 7:23 PM, Dane Newman <dane.newman at gmail.com>
> wrote:
> > Thanks for the reply Nick
> >
> > I debugged voip rtp named-event and when I tried to hit 1 in the call for
> > dtmf nothing came out of the debug.  Could this possibly mean on my side
> Im
> > not sending dtmf to the service provider?
> > Dane
> >
> > On Tue, Oct 27, 2009 at 4:30 PM, Nick Matthews <matthnick at gmail.com>
> wrote:
> >>
> >> That shows up in the debugs in working scenarios too.  Not sure what
> >> the importance of those statements are, but it's the type of thing you
> >> see when you add 'all' to a debug.
> >>
> >> It's not the 183 you want to look at, but the 200 OK with the CSeq of
> >> your INVITE.  And you want a 200 OK.  I've seen it where the debugs
> >> will show that we're sending DTMF but the provider won't use it, which
> >> is a conversation you would need to have with the provider.
> >>
> >> -nick
> >>
> >> On Tue, Oct 27, 2009 at 3:45 PM, Dane Newman <dane.newman at gmail.com>
> >> wrote:
> >> > Hmm that does not sound good
> >> >
> >> > This is with the default settings
> >> >
> >> > rtp payload-type nte 101
> >> > rtp payload-type nse 100
> >> >
> >> > which don't show up in the config.  Could there be any reason why the
> >> > router
> >> > is not able to use 101 below are my dial peers
> >> >
> >> > dial-peer voice 100 voip
> >> >  description AA Publisher
> >> >  preference 1
> >> >  destination-pattern 1..
> >> >  voice-class h323 50
> >> >  session target ipv4:10.1.80.10
> >> >  dtmf-relay h245-alphanumeric
> >> >  codec g711ulaw
> >> >  no vad
> >> > !
> >> > dial-peer voice 1000 voip
> >> >  description incoming Call
> >> >  translation-profile incoming aa
> >> >  preference 1
> >> >
> >> >  incoming called-number 6784442454
> >> >
> >> >  dtmf-relay rtp-nte
> >> >  codec g711ulaw
> >> >  ip qos dscp cs5 media
> >> >  ip qos dscp cs5 signaling
> >> >  no vad
> >> > !
> >> > dial-peer voice 101 voip
> >> >  description AA Subscriber
> >> >  preference 2
> >> >  destination-pattern 1..
> >> >  voice-class h323 50
> >> >  session target ipv4:10.1.80.11
> >> >  dtmf-relay h245-alphanumeric
> >> >  codec g711ulaw
> >> >  no vad
> >> > !
> >> > dial-peer voice 2000 voip
> >> >  description outbound
> >> >  translation-profile outgoing addone
> >> >  preference 1
> >> >  destination-pattern .T
> >> >
> >> >  progress_ind setup enable 3
> >> >  progress_ind progress enable 8
> >> >  session protocol sipv2
> >> >  session target dns:did.voip.les.net
> >> >
> >> >  dtmf-relay rtp-nte
> >> >  codec g711ulaw
> >> >
> >> > !
> >
> >
>
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