On Tue, Jan 12, 2010 at 12:09 PM, listu...@spamomania.co.uk
<listu...@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). 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. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com -- _____________________________________________________________________ -- 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
