Very basic Asterisk configuration question (re: Sipura 3000)

[EDIT] I’ve rewritten this, since I’ve figured out my first problem but now I have a new problem…

Incoming calls from my Sipura 3000 don’t seem to be correctly routing to Asterisk (or something?)

Here is my Asterisk configuration for my incoming PSTN line:

[1000]
type=friend
host=dynamic
context=incoming
secret=6769
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

Inside of extensions.conf, I have this:

[incoming]
exten => s,1,Answer( )
exten => s,2,Background(enter-ext-of-person)

When I call my PSTN line, my Sipura 3000 seems to successfully answer it because the line rings once, but then immediately switches to a second dial tone. Shouldn’t my incoming call be answered and then have “enter-ext-of-person” played to them?

What could be causing this?

you need to go into the advanced settings of the PSTN line and put in Dial Plan 2 this without the quotes: “(S0<:s@192.168.0.2:5060>)”

The IP address above should be your Asterisk server.

Then choose “2” in the input box called “PSTN Caller Default DP:”

The above instructions along with the guide found below should get things working:

voipspeak.net/index.php?option=c … &Itemid=28