[asterisk-users] duration of rfc2833 generated dtmf

[Brent Davidson]

To the best of my knowledge, the only way for you to control the 
duration sent to the PSTN lines is for you to be directly connected to 
the lines so you can set the tone duration in zapata.conf / dahdi.conf 
or to use inband signalling.

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.

-Brent

John covici wrote:
> OK, thanks.  If I could convince them to use info, would that be
> better as far as the duration is concerned?
>
>
> on Monday 04/13/2009 Brent Davidson(brent at texascountrytitle.com) wrote
>  > John covici wrote:
>  > > Hi.  I have a SIP provider which wants RFC2833 for the dtmfmode,
>  > > however I would like to increase the duration of the tone, its pretty
>  > > short and some IVR's are unhappy or don't detect it.  I did poke
>  > > around, but it looks like when RFC2833 is used, it actually generates
>  > > rtp packets of some sort, so I have no idea how to increase that
>  > > duration.
>  > >
>  > > Any assistance would be appreciated.
>  > >
>  > >   
>  > 
>  > If your provider insists on rfc2833, then their servers will be 
>  > responsible for setting the tone duration sent to PSTN lines.
>
>