Asterisk AMD to detect free tone

Hi all,

i have a strange problem so i need a workaround…i place outbound calls through a Panasonic (via call files) pbx to playback message and received dtmf input from people…but Asterisk don’t recognized dialout (and so human answer) and playback message as soon as system dial number…so message heard by people is cut by several seconds…I can’t modify sip.conf too much due to several services running after days and days of work…
In italy we have 4 second silence and one long beep to show free line tone…
I have set a wait for silence for 4,3 sec and it works…but…is not so practical…
Is there a way to detect free line tones so to start playback after free tones stop?