asterisk users - Feb 2005 - Settings for SIP to dial PSTN with TDM400P w/FXO module

I've setup * with TDM400P w/1 FXS, 3 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of
the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP).
 
The calls between SIPs and zap phone (fxs) are OK.  But 2 issues cannot be
solved:
 
1. To dial to PSTN via zap phone, the setup in extensions.conf with the
following
     exten => _9Nxxxxxx, 1, zap/2
   doesn't work well.  The symptom is that you have to dial several times to
get success dial out.
   if I used 
     exten => _9., 1, zap/2
   then I have to pick th phone, dial 9 first, wait for dial tone again, and
then dial local call to PSTN.  I works every time I dial out.
 
   Does anyone can give me suggestion that what did I do wrong to make the
setting:
     exten => _9Nxxxxxx, 1, zap/2
   unstable?
 
2. When trying using SIP phone to dial PSTN, I got no luck.
   The call either got busy tone or some prefix missing warning from CO.
   Please advice if any solution.
 
Thank you so much.
 
Chichi
 


		
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Please help me, i can only able to register 1 port of my 6 port fxo (sip) with
asterisk, it alway last one register. not all port. how to fix this proble.
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