[Asterisk-Users] Settings for SIP to dial PSTN with TDM400P w/FXO module

steven c baloveaa at yahoo.com
Tue Feb 22 17:06:37 MST 2005


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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