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 =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001
"The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck." (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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