[Asterisk-Users] extensions problem (SIP)

[Jon Lawrence]

On Monday 15 March 2004 16:00, Olle E. Johansson wrote:
>
> It depends on your SIP device. Asterisk places the call to your SIP device
> regardless, since by SIP protocol design the UA is not a "slave", it is
> free. So the SIP ua must answer "busy" for Asterisk to understand that
> you're busy. If not, the call is placed to you and Asterisk has no
> knowledge that you are busy. Check you SIp phone if you can limit the
> number of concurrent calls.

So does anyone know if the Grandstream handytone-286 sends this "busy" answer 
?
I'm guessing it doesn't. In that case, what other ways are there of connecting 
my dect phones to a voip * system ? - can I just connect them into the 
x100p's phone socket (how do I send calls to that port) or do I need to get a 
fxs card and run wire's everywhere  - not an option :(
How does everyone else connect up DECT phones to a * based system.

Surely * should know if a phone is in use ? After all it initiated/took part 
in the call in the first place ;)

Jon