On Tue, 2010-01-12 at 16:52 -0500, Kristian Kielhofner wrote: > 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). 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 -- _____________________________________________________________________ -- 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