[asterisk-users] Sipgate > DTMF not detected

[listuser at spamomania.co.uk]

On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote:
> On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes <steve-lists at geekinter.net> wrote:
> >
> >
> > Codec? I've had 2833 do funny things with anything other than a/ulaw
> > (might just be me though..)
> >
> > S
> >
> > --
> 
> Codecs other than G711u/a don't support inband DTMF.  Seeing as INFO
> is rarely used that pretty much leaves RFC2833.  Turn on SIP debugging
> and look in the INVITE from the provider for telephone-event.  If you
> see it, they're configured to use RFC2833.
> 
> If they are, you need to do a packet capture or other RTP debug to see
> the out of band RFC2833 events.
> 
> -- 
> Kristian Kielhofner

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?