[Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

[Walter Doerr]

On Thu, Jan 29, 2004 at 05:04:22PM +0100, Cristian Manoni wrote:
> Hi All
> i have continuos error:
> Unable to handle DTMF tone 'f' for 'SIP
> on the asterisk console.
> after this the call hang up.

Look at softdtmf in capi.conf.
Setting the parameter to 0 solved the problem for me.

-Walter



-- 
  Walter Doerr   =*=   wd at infodn.rmi.de   =*=   FAX: +49 2421 962001
              "The poor folks who only have 100MBytes of RAM five years
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