[Asterisk-Users] Limiting incoming SIP calls & Original CallerID on transfer

[Erik Barker]

I have 2 issues which I need to resolve on our production Asterisk
server:


We are currently using Polycom IP600 VOIP phones for our office which
are capable of handling 2 calls per SIP registration. What we're finding
is when staff are on the phone, Asterisk will pass them a second call
which will show up on their display, and an audible beep is heard over
the phone (regular call waiting). I would like to limit the number of
calls sent to each phone to 1 call only; otherwise respond as being
busy. I have looked at trying to accomplish this in the sip.conf by
using the 'incominglimit' and 'outgoinglimit' parameters, however, the
only one that *seems* to work is the 'incominglimit'. This prevents
further calls from reaching the phones, rings busy, but does not allow
our phones to initiate a 2nd call OR transfer their existing call. The
'outgoinglimit' parameter does not seem to have any effect on limiting
whatsoever. Is there a way to limit calls passed to the phones from
Asterisk and also allow each phone to initiate 2 calls or transfer calls
(disable call waiting)??

I have also looked at the WIKI for the parameters listed above and it
*appears* that 'outgoinglimit' should do what I want, however it also
states that this function has been disabled??

"The _outgoinglimit__ is currently disabled in the source code of the
SIP channel."
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit



My second problem is that when external calls are transferred by our
receptionist to other staff members, the CallerID of course changes to
her Name instead of the original caller. Is there a way (in the
extensions logic or other) to preserve this CallerID information so that
staff members receive calls with the proper CallerID information?


Thanks,


-- 
Erik Barker