On Tue, 2010-01-12 at 10:37 -0500, Kristian Kielhofner wrote: > On Mon, Jan 11, 2010 at 12:19 PM, Steve Howes <steve-li...@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? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users