[asterisk-users] duration of rfc2833 generated dtmf

[Kristian Kielhofner]

On Mon, Apr 13, 2009 at 5:32 PM, John covici <covici at ccs.covici.com> 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.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com

John,

  Assuming this is Asterisk 1.4 or later...  The duration used by
Asterisk is the same duration sent from the phone.  The duration of
those DTMF key presses should match the time the user is holding down
the key.  What type(s) of phones are these?  You should also look into
using Asterisk 1.4.24.1 or later (if you aren't already).  There have
been many improvements to the RTP code to better handle quirks with
the equipment (especially Sonus) used by various providers.

  Assuming your provider is to spec (and so is your phone) your
provider should not be complaining that the duration of your DTMF key
presses are too short...

  With that being said AFAIK there is no way to specify a minimum
duration for an RFC 2833 DTMF in Asterisk on a bridged channel.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com