[asterisk-users] Random DTMF Tones Only on heard on ATA

[Travis Langhals]

Thanks Sherwood for all the info.

The devices are using ulaw and rfc2833.  There is no transcoding on my
server, but not sure what my trunk providers are doing.

I was thinking about the frequency detection issue as it seems to be
primarily involving women so I'll try adjusting the input/output gain to see
if that helps.  I've tried all other combinations of DTMF settings on the
ATAs so that's my last hope on the device end.

I'm hoping to avoid the packet capture as that is never a fun road to go
down but that is probably the next step.

Any idea why Asterisk still ques/replays the tone considering it see's it's
shorter than the 80s minimum?

Travis

Hoping to avoid
On Wed, Jul 28, 2010 at 7:16 PM, Sherwood McGowan <
sherwood.mcgowan at gmail.com> wrote:

> 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 <travis at 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+usenet at amorsen.dk<benny%2Busenet at amorsen.dk>
> >
> > wrote:
> >>
> >> Travis Langhals <travis at 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
> >>
> >
> >
> > --
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