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

[Dane Newman]

I feel embarasssed I been blaming a telco all day when it was my own
ignorance that made it not work

I was not matching the inbound dialpeer for my outbound calling.

I added the following dial peer and the problem was solved

dial-peer voice 150 voip
 description incoming outbound
 preference 1
 voice-class h323 50
 incoming called-number .T
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!


Thanks alot Nick and Ryan for all your help on this!

Dane



On Tue, Oct 27, 2009 at 8:49 PM, Nick Matthews <matthnick at gmail.com> wrote:

> So you have a H323-SIP CUBE, and your DTMF isn't working.  This is
> probably the most common problem with CUBE users.
>
> For this #1 problem, the number one cause is 'incoming called-number .'
>
> Most people don't really understand inbound dial peer matching, and
> have used this for ages on normal TDM gateways that were
> single-protocol.  The best way to fix this is to read the 'Understand
> Incoming and Outgoing Dial-Peers" document on Cisco.com, and figuring
> out the best way to match dial peers for both your incoming/outgoing
> SIP/H323 legs.  You can prefix digits and match on incoming called
> number, or ditch incoming called-numbers completely and use
> answer-address.
>
> I like using 'debug voip ccapi inout' to determine this.  You can do a
> search for peer= after you've got the debug to find out which dial
> peers you're hitting for each case, plus what the numbers look like
> after translations, etc.  'debug voip dialpeer' is an alternative, but
> I personally find it more confusing.
>
> For h323-SIP your dial peers should look something like this:
>
> incoming h323 dial peer for outgoing call:  dtmf-relay h245-alpha or
> h245-signal
> outgoing sip dial peer for outgoing call: dtmf-relay rtp-nte
> digit-drop (plus any payload commands required)
> incoming sip dial peer for incoming call: same as sip option above
> outgoing h323 dial peer for incoming call: same as h323 option above
>
> My best guess is that if you look at your incoming/outgoing dial peers
> something isn't matched correctly.
>
> -nick
>
> On Tue, Oct 27, 2009 at 8:17 PM, Dane Newman <dane.newman at gmail.com>
> wrote:
> > 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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