Here is an example, very basic.

sip.conf
=========
[1000]            ;assuming that your first SIP extension is 1000
...               ;all the other paramters
context=internal  ;default context is [internal]

extensions.conf
================
[globals]
OUTBOUNDCHANNEL=Zap/2

[internal]
include => outbound
... ;rest of the [internal] context

[outbound]
exten => 911,1,Dial(${OUTBOUNDCHANNEL}/911}

exten => _9N.,1,Dial(${OUTBOUNDCHANNEL}/${EXTEN:1})
exten => _9N.,2,Congestion()
exten => _9N.,102,Congestion()

...     ;the rest of the dialplans

This will allow anyone who dials a 9+number to call that number out Zap/2
(your FXO port). If it cannot dial out that port then the dialer heres a
busy signal. There is a specific line for 911 since it is so important.
Again, this is a very simple example, there have been more in depth ones
posted on the mailing list that when 911 was dialed checked if the FXO Zap
channel was in use and would hangup the channel then dial 911. not that
hard, but I don't remember how to do it off the top of my head (I'm still
learning :) ).

Hope this helps.

Ryan


> I have one SIP phone (and soon a 2nd phone) and a Digium TDM11B (1 FXO +
> 1 FXS) card.  I would like to be able to dial out the analog line via
> Asterisk.  How do I configure that?
>
> i.e  I'd like any extension to be able to dial 411, 911, 0,  (617)
> 555-1212, 16175551212, etc... and have these routed out the POTS line.
> Just like a regular telephone.
>
> % cat /proc/zaptel/1
> Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
>
>            1 WCTDM/0/0 FXOKS (In use)
>            2 WCTDM/0/1 FXSKS (In use)
>            3 WCTDM/0/2
>            4 WCTDM/0/3
>
> % cat /etc/zaptel.conf
> fxoks=1
> fxsks=2
>
>
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