asterisk users - Jan 2004 - Cisco FXO as PSTN gateway

I have been compiling information on this configuration onto the Wiki:
http://voip-info.org/wiki-Asterisk+cisco+FXO

I can call out to the PSTN just fine, but inbound calls all appear in my 
[bogon-calls] context.

Can anyone help me locate why?
(Config files are on the Wiki)

I have done a packet sniff & decoded using Ethereal-0.10.0, but this 
doesn't tell me a great deal - I just see the rejection message:
y.y.y.y	x.x.x.x	INVITE sip:1234@x.x.x.x:5060
x.x.x.x	y.y.y.y	Status: 100 Trying
x.x.x.x	y.y.y.y	Status: 503 Service Unavailable
y.y.y.y	x.x.x.x	Request: ACK sip:1234@x.x.x.x:5060


Thanks a lot,
Fran.

Hi!
> I have been compiling information on this configuration onto the Wiki:
> http://voip-info.org/wiki-Asterisk+cisco+FXO
Try this in your [pstn] section:
autocreatepeer=yes

Add this to each (!) of your SIP users:

disallow=all 
allow=ulaw 

Also: The syntax is "username=" and not "user=".
Same for "dtmfmode" instead of "dtmf".

Cheers, Philipp