[Asterisk-Users] DTMF not working

[Innocent Evil]

Hi Rob,

I am using RFC2833 everywhere including SIP phone, asterisk's sip.conf
Do you think, to raise the value from 100 to 400, would solve my issue?

Thanks,



> -----Original Message-----
> From: rtarte at pacificcodeworks.com
> Sent: Wed, 24 Aug 2005 08:46:43 -0700
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] DTMF not working
>
> Hi Mr. Evil,
>
> I'm not sure if the problem that I am describing relates to the problem
> that you are having.  It seems that when you press a key on a SIP phone
> that is set for inband DTMF, asterisk absorbs the tones until you
> release the key.  This way if you are using DTMF to do things like
> transfer calls, the user won't get tone blasts in their ear until
> asterisk has had a chance to interpret the tones.   After asterisk has
> figured out what to do with the tone, it generates and transmits it's
> own tones in the routine do_senddigit() (assuming that the DTMF tone
> should be passed on).  The duration of the DTMF tones that asterisk
> generates is fixed and independent of how long you pressed the key on
> your phone.
>
> In the line "!941+1336/100,!0/100", the 941 is one tone of the DTMF
> (dual tone multi-frequency), and 1336 is the other tone.  The 100 is the
> duration of those tones.   The tones are in Hz.  I'm not sure what units
> the duration is in, but I bumped mine from 100 to 400 and that seems to
> do the trick.  The part of the string that reads "!0/100" just shuts the
> tone generator off.
>
> Rob
>
> Innocent Evil wrote:
>
> >I am having same problem .. DTMF is not working from a SIP phone while
> >sending to Asterisk cmd VoiceMailMain.
> >
> >Would you please explain this line
> >"!941+1336/100,!0/100", /* 0 */
> >
> >what  value is what and how it affect on DTMF tone generation.
> >
> >Thanks,
> >
> >
> >
> >>I had a similar problem that seems to be caused by the DTMF tone
> lengths
> >>being to short.  Try this:
> >>
> >>Asterisk generates DTMF tones in  do_senddigit() in the file channel.c.
> >>The tones are defined in a const char array called dtmf_tones[].  Each
> >>DTMF tone is a string that looks something like:
> >>
> >>"!941+1336/100,!0/100", /* 0 */
> >>
> >>The part that reads !941+1336/100 is the part that you want.  Change
> the
> >>"100" to something bigger and recompile.  You will have to do that for
> >>every tone.   I'm using 400 right now, and it seems to be working.
> >>
> >>I hope that helps.
> >>
> >>Rob
> >>
> >>Peter Osborne wrote:
> >>
> >>
> >>
> >>>Hi all,
> >>>
> >>>I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no
> >>>
> >>>
> >>longer
> >>
> >>
> >>>works with external phone systems. I have a Wildcard TDM400P with 4
> >>>
> >>>
> >>FXO's?
> >>
> >>
> >>>(it connects to analog lines). No changes were made to the config
> files.
> >>>
> >>>Here's my config:
> >>>
> >>>/etc/zaptel.conf
> >>>fxsks=1-4
> >>>loadzone = us
> >>>defaultzone=us
> >>>
> >>>/etc/asterisk/zapata.conf
> >>>[channels]
> >>>usecallerid=yes
> >>>hidecallerid=no
> >>>callwaiting=yes
> >>>usecallingpres=yes
> >>>threewaycalling=yes
> >>>transfer=yes
> >>>cancallforward=yes
> >>>callreturn=yes
> >>>echocancel=yes
> >>>echotraining=yes
> >>>rxgain=2.0
> >>>txgain=2.0
> >>>callgroup=1
> >>>pickupgroup=1
> >>>musiconhold=default
> >>>context=incoming
> >>>group=1
> >>>signalling=fxs_ks
> >>>echocancel=64
> >>>echocancelwhenbridged=yes
> >>>relaxdtmf=yes
> >>>channel => 1-3
> >>>
> >>>[pete_desk]
> >>>;Pete's Desk phone (Polycom IP 300)
> >>>type=friend
> >>>username=pete_desk
> >>>secret=pass
> >>>context=longdistance
> >>>callerid=Pete <601>
> >>>host=dynamic
> >>>mailbox=601
> >>>dtmfmode=inband
> >>>disallow=all
> >>>allow=ulaw
> >>>allow=alaw
> >>>
> >>>Thanks,
> >>>Pete
> >>>
> >>>
>
> --
> Robert Tarte
> Pacific CodeWorks
> P.O. Box 29050
> San Francisco, CA 94129
>
> (p) 831-426-7582
> (f) 831-426-7584
>
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