Basic SIP usage of SPA3102

Hi All,
Whilst I'm debugging, I just wanted to ask a quick few basic questions on using Asterisk and the 3102 together.
The only port I was able to connect into the 3102 via IP was the Internet port, so my LAN is connected into that. The 3102 doesnt connect directly to the Internet in any way, so my LAN should be "the internet" from its perspective.
All I want to make use of the 3102 for is directing some specific outbound calls from Asterisk to the PSTN via the 3102.  Maybe at some point have any inbound call direct into Asterisk, but thats for a later stage.
I've defined a SIP account and the 3102 registers into A*k ok.
I dont understand the dialplan syntax on the 3102 (any docs?? the user manual had nothing), so thats my starting point for investigation.  At the moment its set to the default of:
 (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) 
When I place a call via Asterisk I get :
-- Executing [2090@from-sip:1] Set("SIP/427-08ef25d0", "CALLERID(num)=02071833427") in new stack
    -- Executing [2090@from-sip:2] Dial("SIP/427-08ef25d0", "SIP/[email protected]") in new stack
    -- Called [email protected]
    -- SIP/192.168.3.54-08ef0758 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [2090@from-sip:3] Hangup("SIP/427-08ef25d0", "") in new stack
I'm assuming this is the correct way to direct the call to the 3102.
I'm thinking its the dialplan thats the problem, but how do I make all calls go via the PSTN ??
Thanks,

try to set the dial plan to
xx. |  <9, xx. < : @GW0>
the xx. will let you dial anything
dialing 9 then the pstn number will go out of the ports line
if you have the PSTN line registered as well to the asterisk, just dial that extension number and you should get a second dial tone for dialing to the pots line.

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