[asterisk-users] Sipgate > DTMF not detected by Asterisk 1.6 [dtmfmode=rfc2833]

[listuser at spamomania.co.uk]

On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote:
> On Tue, Jan 12, 2010 at 12:09 PM, listuser at spamomania.co.uk
> <listuser at spamomania.co.uk> wrote:
> >
> > Assuming that I enable debugging using:
> > asterisk -rvvvvvvvvvv
> > CLI> sip set debug on
> >
> > Then with this:
> > dtmfmode=rfc2833
> > disallow=all
> > allow=ulaw
> > allow=alaw
> >
> > I see nothing nothing showing keypresses scroll past me. Even a SIP TCP
> > dump shows nothing. SIPGATE have said;
> >
> > "you should be able to set the dtmfmode to rfc2833 in your default
> > sip.conf.
> >
> > Best regards,
> >
> > Frederik"
> >
> > I've tried other combinations such as info, inband et al. I'm guessing
> > {that's all it is} that rfc2833 will signal the dtfm over sip as opposed
> > to in the audio stream?
> >
> 
> RFC2833 is carried in RTP like the audio stream.  However, it uses a
> different payload type from the RTP packets used to transport the
> audio.  If you did an RTP capture you would be able to see the RFC2833
> events (which correspond to DTMF presses).
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:

Real-Time Transport Protocol
10.. .... = Version: RFC 1889 Version (2)
Payload type: ITU-T G.711 PCMU (0)

And as you say, the DTMF events are clear to see:
RFC 2833 RTP Event
Event ID: DTMF One 1 (1)
..00 1010 = Volume: 10

So, as these can be seen in the stream, do I need to tell Asterisk to
detect these? Does it not do that when I set: dtmfmode=rfc2833
???

> 
> The SIP debug, however, will tell you if the remote end is configured
> to use RFC2833 or not.  That's why I was telling you to look for
> telephone-event in the INVITE from your provider.  Keep in mind SIP
> (most likely) runs over UDP between you and your provider, not TCP.
> 
I see a 'telephone-event' :

a=rtpmap:101 telephone-event/8000

buried in the chunk below. but I have to be honest, SIP is new to me so
I'm not sure of myself with this:

v=0
o=root 27089 27089 IN IP4 217.10.69.13
s=session
c=IN IP4 217.10.69.13
t=0 0
m=audio 19990 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv