Sorry, I came into this late...what codec is the device using, and is the audio being trascoded?
Back at Voxitas, we had a couple of customers complain about random DTMF tones coming across their line, and Asterisk WAS actually "hearing" DTMF tones...want to know what it was?..... In that particular case (just a place to start looking) it was G729 on customer ATAs (don't remember the models)....Here's the freaky thing....It only happened with CERTAIN people talking on the phone...IIRC, we determined that the ATA's G729 processor was mistaking certain audio frequencies in the speaker's voice and believing it was a DTMF tone from the analog device and sending the appropriate DTMF signal to our servers... I'm sorry, I don't remember how we fixed it...I think we did some audio tweaking (advanced ATA config, input level, out level, etc..), be we may have just ended up having to tell that client to not use G729 on those ATAs.... This _MAY_ happen with other codecs, but I think it's mainly either G729..maybe primarily transcoding? NERDY FUn Crap below: capture SIP and RTP between your Asterisk and an offending device (writing to a file)....then start doing everything you can to cause the DTMF issue to occur. NOW, open your capture in wireshark...dump the RTP payload to a file and open that file in an audio editor.... Now, go through the wireshark capture...see if you see any DTMF events (if rfc2833 it'll be an RTP EVENT, if SIP INFO, it'll be a sip info, and if you're using inband **SHUDDER** you can just listen to the audio).....note the time in seconds from the beginning of the audio stream whenever a DTMF event occurs, and then go to that spot in the audio file....If you're feeling REALLY frisky, do a frequency analysis...I'll bet you'll see that the voice that is speaking at the time of the DTMF event on your various captures will have a frequency range in common...a very close range...maybe look up DTMF tone definition and get the freqs....(did it....more detail than even I feel like doing right now :D) Cheers, Sherwood McGowan On Wed, Jul 28, 2010 at 6:43 PM, Travis Langhals <tra...@netitek.com> wrote: > SIP/5211 is a Grandstream device. > Did not add relaxdtmf=no, but sip show settings verifies it's already set to > no. > Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 > Travis > > On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen <benny+use...@amorsen.dk> > wrote: >> >> Travis Langhals <tra...@netitek.com> writes: >> >> > [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on >> > SIP/5211-00000078 >> >> Is SIP/5211 a Linksys or a Grandstream or something else? >> >> Do you have relaxdtmf=no? >> >> Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? >> >> >> /Benny >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users