[Asterisk-Users] Newbie - Looking for pointers

[Adams, Gavin]

Hi All,

I've been lurking on the list hoping to absorb all the knowledge
flitting past while the bits and pieces of my new * server arrive. Well,
most of the bits and pieces are here, and I've got the Digium hardware
installed and (I think) loaded properly (RedHat 9, Compaq 1850R, T100P,
TDM400P 2 port FXS). I'm still waiting on my Cisco 7940/7960 handsets
(ordered with SIP).

CVS'ed down all the components, sample configs, and "asterisk -vvvgc"
starts clean. The only change made so far is to noload chan_oss.

Anyway, I'd like to get comfortable with * prior to replacing our
Siemens Hicom system. The PSTN is PRI for BellSouth, and we're going to
use SIP phones internally.

Now for the stupid questions....

1) How do I figure out what the Zap channels are for the FXS ports on
the TDM400P? I have green lights on the 2 active ports, but no dialtone
when * is running. I assume I have to bind the port somehow to an
extension and/or dial plan?

2) Architecturally, I'd like to be able to support NATted phones both
locally, via the Internet, and from our remote offices. Currently the *
server is on a screened subnet off our firewall with registered IP
address space. All local phones are on a 172.16.x.x private address
space network, and NAT hidden to a single public address viewable by the
server. Can multiple SIP phones be hidden behind a single IP address, or
do they each need their own static address?

If this isn't possible, the plan would be to place the server on the
same subnet as the phones, and deal with the SIP Phone
->NAT_>Internet->NAT->* in the future.

Okay, back to the handbook to learn how contexts all play nice together!
:)

Regards,

--- Gavin