Hi All i have continuos error: Unable to handle DTMF tone 'f' for 'SIP on the asterisk console. after this the call hang up.
I have a BGT 101 that make and receive call from the capi channel Thanks _______________________________________________ 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
