[Asterisk-Users] probing a SIP device for redirection information?

[Louis-David Mitterrand]

Hello,

I'd like to find a way to probe a SIP phone for forwarding information
before I actually Dial() it. For instance, if an absent user entered a
forwarding number in his (Cisco or Polycom) phone, it will anwser a
Dial() with a REDIRECT and asterisk will comply if the context allows.

However I'd like to intercept that REDIRECT, get the number and use the
telco's call deflection service to deflect the call without answering
it. That way, once the deflection is done, no telco lines are used.
Whereas with asterisk's default behavior, two lines would be used: one
for the incoming call and one for the forwarded call.

I've looked in the docs without finding, for a way to get a SIP device's
status: will it forward the call? if so to what number? 

Thanks for your insights,

-- 
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