[Asterisk-Users] fine-tuning asterisk questions

[M.Hockings]

I have asterisk running more or less ok but I would like to turn off 
call waiting and be selective about the incoming sip connections.  This 
is running asterisk 1.2.8 with a fxs and fxo card and a configured voip 
(sip) line.  Currently I'm using freePBX 2.1.1 to configure asterisk.

Problem 1) if someone is on the phone already and another call comes in 
for an already engaged extension I want it to go to voicemail directly 
rather than have that distracting call-waiting beep going on.
As far as I can tell I have turned off call waiting in the zaptel config 
files.  What else should be set to avoid call-waiting ?

Problem 2) Incoming sip calls from my voip provider get rejected unless 
I allow anyone to connect with sip. I have an incoming route set up with 
the right DID that matches the DID that asterisk picks out but it still 
rejects the call.  Any suggestions about how to get this to work without 
allowing any sip connection?


Mike