[asterisk-users] Restricting transfers between SIP phones

[C. Chad Wallace]

Hello,

We are in the process of splitting our phone system into two separate
logical systems for our two departments.  One of the goals of this
switch is to restrict members of one department from transferring calls
to the other, but not restrict them from calling that department
themselves.  So what I need to know is how to detect whether a call
from a member of that department is a transfer or an original call.

I've looked at the TRANSFER_CONTEXT setting, but that's only for
transfers with # and the T and t flags to Dial().  But we use SIP
hardphones (Linksys SPA942 & Grandstream GXP2020), which have built-in
transfer functions, and we would like to continue using those for
transfers, rather than building it into features.conf or dialplan...
Because we prefer attended transfers, and the user experience seems
more modern.

So, does anyone know of a way to detect whether a call from a SIP phone
is the first step of an attended transfer or an original call?  

Thanks!

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 197 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091125/e382d0c0/attachment.pgp