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) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users