[moving to asterisk-users by request] On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jt...@digium.com> wrote: > > On Jan 26, 2009, at 7:38 PM, James Lamanna wrote: > >>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote: >>> >>>> Hi, >>>> Is it just me, or does DTMF queuing not work properly? >>>> I'm consistently faced with the issue that users (and myself) will >>>> dial digits quickly and all I get in the logs are: >>>> >>>> end 'X' put into dtmf queue on SIP/xxxxxxxxxx >>>> etc... >>> >>> >>> What version are you talking about? If it's not 1.4.23, please try >>> that, as there are some related fixes in that version. >> >> Sorry, I neglected to mention this is on 1.4.18.1. >> I will try and test 1.4.23 and see if things are better. >> In the meantime, I'll report my findings to see if you guys can better >> explain >> to me what is going on. >> >> The best DTMF combination (between phone and asterisk) I have found is: >> >> sip.conf - dtmfmode=info >> Phone (SPA962) - DTMF Mode = Auto >> >> This works very well for outbound SIP and Zap trunks and on both ulaw >> and g726 codecs. >> However, this does NOT work for any prompt that is internal to asterisk >> that >> needs to detect DTMF (Voicemail, Authenticate, etc..). >> The only way for these prompts to work is to explicitly put >> SIPDTMFMode(inband) >> in the dial plan. Of course, this breaks when the codec is g726. Why >> do these prompts not work with this setup? >> I've also noticed that when in this mode, nothing is put into the dtmf >> log. >> Does that mean that the phone and asterisk have negotiated inband >> (though if this was the case why would it work with g726..)? >> >> Thanks. >> (please CC me directly since I'm on digest mode at the moment). > > > Actually, I'm a little surprised you get DTMF working at all in this > combination. > > Setting dtmfmode=info means that Asterisk will be looking for SIP INFO > messages that contain DTMF events. Have you watched the SIP channel debug > during DTMF events, or set up a tshark or other interceptor to watch port > 5060 as you send DTMF? Perhaps you've got a few things mucking up the > works there. What does RFC2833 get you if you set all the gear to that? > > Try setting everything to RFC2833 and try again. I'd also suggest follow-up > messages go to asterisk-users and not to this list, as this is not sounding > particularly like a question for the -dev list where core Asterisk code is > involved, and you'll probably get more answers over on -users.
Still haven't had a chance to upgrade to 1.4.23, however I did try setting everything to rfc2833. Works ok @1.4.22.1, but not as good as the info/Auto combination. I looked at the SIP signaling trace, and I see no sign of INFO headers in the signaling packets. What's going on here then? I see no payload 101 packets in the RTP stream either. So I guess it negotiated inband? (then why the hell would it work while in g726 - am I just getting lucky??) I have verified that the RTP payload packets have type 2 which is g726aal2. Note, It is getting transmitted inband, if I switch back to ulaw and look at the packets, I can see no audio until I press a digit. And why doesn't this get detected by Asterisk for any of its internal applications? -- James > > JT > > --- > John Todd email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.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