[Asterisk-Users] Can we replace existing SIP call with new one?

[JS]

Is there a way to configure Asterisk to do the following:

1) If calls comes in for Alice, call Alice on SIP channel
2) If Alice is busy, terminate existing call (assuming Alice was in call
with somebody via Asterisk)
Note: We can use n+101 priority here.
3) Try again to call Alice on SIP channel

I know how to do step1 and step3 :) I am concerned about step2.

I see two possible ways:

step2 can be done by sending BYE for existing call. Then trying
Dial(SIP/Alice) again would work.
I don't know if there is a command to "terminate existing call" OR "bring a
specific channel down" in Asterisk.
Note: I have access to call-id, channel-id, username, peername for existing
call that Alice is in, if they help.

steps 2 and 3 can be done together if I can reINVITE Alice to join new call.
I don't know how to
send reINVITEs manually though (canreinvite=yes won't help).

-Jim
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