Multi line SIP phone and Asterisk

I’m using a fleet of 40 Polycom 501 phones, each have 2 lines that needed for call forwarding, conferancing etc. (without asterisk help…)
When a user is using the phone and another call is entering the calling party get a noraml dialing tone while the called party get from the 501 a call waiting signal. If the called party ignore the new call then the other side will not get any answer without knowing that the called party is actually there and talking. Is there a way to signal the calling party that the called party is talking on the other line? some tone or signal?