[Asterisk-Users] DTMF not working

[Innocent Evil]

Everywhere it is RFC2833 including in SIP phone, Asterisk's sip.conf.
DTMF work only from the phone that is hooked with asterisk box.

Thanks,


> -----Original Message-----
> From: jnovack at stromberg-carlson.org
> Sent: Wed, 24 Aug 2005 12:04:04 -0400
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] DTMF not working
>
>
>
> Innocent Evil wrote:
>
> >I am having same problem .. DTMF is not working from a SIP phone while
> >sending to Asterisk cmd VoiceMailMain.
> >
> >
> >
> Have you set DTMF to out of band RFC2833?
>
> In band won't work. At least in my version of HEAD
>
> John Novack
>
> >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
> >>>_______________________________________________
> >>>Asterisk-Users mailing list
> >>>Asterisk-Users at lists.digium.com
> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
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> >>>
> >>>
> >>>
> >>>
> >>--
> >>Robert Tarte
> >>Pacific CodeWorks
> >>P.O. Box 29050
> >>San Francisco, CA 94129
> >>
> >>(p) 831-426-7582
> >>(f) 831-426-7584
> >>
> >>_______________________________________________
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> >>
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> >>
> >>
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> >
> >
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