[asterisk-users] duration of rfc2833 generated dtmf

[D Tucny]

2009/4/15 John covici <covici at ccs.covici.com>

> Its not there and the link you gave me says its for sip originating
> rather than calls to a sip channel.
>
> on Tuesday 04/14/2009 Brent Davidson(brent at texascountrytitle.com) wrote
>  > It's been around awhile.  I've used it in 1.4  Check out this link for
>  > basic info:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode
>  >
>  > John covici wrote:
>  > > Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
>  > > Is this new in 1.6?
>  > >
>  > > on Tuesday 04/14/2009 Brent Davidson(brent at texascountrytitle.com)
> wrote
>   > >  > One thing you might try is researching the "SipDtmfMode" command.
>  It
>  > >  > allows you to change the DTMF mode on an active channel.  A
> suggestion
>  > >  > might be to set up the dial command with the M() option that point
> to a
>  > >  > Macro that changes the DTMF to INBAND once you are connected to the
>  > >  > problem number.  At least in theory, if your provider is expecting
>  > >  > RFC2833 and they get inband, they should just ignore the inband
>  > >  > signaling and pass it on as part of the audio stream.  The only
> problem
>  > >  > is that this may only work if you use uLaw or aLaw for your codec
> and I
>  > >  > don't know exactly how to set the tone duration without having a
>  > >  > zapata.conf or dahdi.conf entry.  Even with one of those files, I
> don't
>  > >  > know how Asterisk chooses to do the rfc2833 to inband translation
> or
>  > >  > where it pulls the toneduration setting from if no PSTN interface
> is
>  > >  > involved in the call.
>
>
It's been there from at least 1.0...

But, you are correct, it's for use on incoming SIP calls rather than
outgoing SIP calls...

d
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