[Asterisk-Users] Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)

[John Klimek]

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:
Code:

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


Inside of extensions.conf, I have this:
Code:

[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?

Also, on a side note, I have a context called [home] which each SIP
Phone is associated with.  Do I need to specify each extension in
there?

For example:

exten => 50,1,Dial(SIP/50)
exten => 50,2,Hangup

exten => 21,1,Dial(SIP/21)
exten => 21,2,Hangup

Can't I just setup a default system where any two-digit number is
assumed to be an extension and it is automatically tried?

Thanks for any help!!