[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Nick Matthews
matthnick at gmail.com
Tue Oct 27 20:14:58 EDT 2009
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
>> >
>> > !
>
>
More information about the cisco-voip
mailing list