[Asterisk-Users] A WORKING EXAMPLE

[Rich Adamson]

> I've looked at the reference examples they are all for SIP.  I have two 
> X100p and a TDM400P.  Can someone send me a working example so I can 
> receive calls and make them.  I'm stuck at first base. [I'm using standard 
> phones - not SIP] Help please!

I just implemented two X100p cards, but not the TDM card. Here's some steps
that I used in a lab environment, but keep in mind I'm also new to this so
I might have missed a couple of steps.

1. Use the two page document that came with the X100p cards and do everything
mentioned in that document.
2. Execute a "/sbin/modprobe wcfxo" as root (assuming you're RedHat)
3. Modify or create a /etc/zaptel.conf file and put only:
 fxsks=1-2                                          
 loadzone=us 
in it. The 1-2 indicates a range of x100p cards (eg, #1 and #2), and it is
configuring the cards as FXS (attaching to an incoming pots line).
4. Execute "/usr/src/zaptel/ztcfg -vv" as root.

Note: the above steps are installing the linux drivers, etc, getting ready
for asterisk to use them.

5. In the /etc/asterisk/extensions.conf file, put:
 [from-sip]
 ignorepat => 9
 exten => _9X.,1,Dial,Zap/1/${EXTEN:1}
The above says... when an * extension dials "9", drop that digit and send
all remaining digits out the Zap/1 (first x100p) interface.

6. In the same /etc/asterisk/extensions.conf file, towards the top put:
[globals]                                                             
PHONE1=SIP/3000                                                  
PHONE2=SIP/3001  

towards the bottom of the file, put:
[inbound-bus]                                                                 
exten => s,1,Dial(${PHONE1}&${PHONE2},15)                        
; exten => s,2,Wait,2                                           
; exten => s,2,Voicemail,u3001                                  
; exten => s,102,Voicemail,b3001 

The above says... were defining two global variables (PHONE1 and PHONE2)
and setting their values to extension 3000 and 3001 (these are assuming
sip extensions, regardless of whether they are sip phones or ata186).
Then, when a call comes in to the [inbound-bus] context, it will ring
both extensions at the same time for up to 15 seconds. If you uncomment
the three lines shown, the call will roll over to Voicemail box 3001 if
the call is "u"nanswered, or, to Voicemail box 3001 if it is "b"usy.

7. In the /etc/asterisk/zapata.conf file, towards the bottom put:
context=inbound-bus    ; this is the context that appears in extensions.conf
switchtype=national                                                            
signalling=fxs_ks                                                              
usecallerid=yes                                            
hidecallerid=no                                              
callwaiting=yes                                             
callwaitingcallerid=yes                                  
threewaycalling=yes                                      
echocancel=yes                                           
echocancelwhenbridged=yes                                       
rxgain=0.0                                                       
txgain=0.0                                                     
pickupgroup=1                                            
immediate=no                                                
callprogress=no                                                 
musiconhold=default                    
channel => 1	; this is the x100p #1 or #2 card (#1 is specified here).

In effect, step 7 configures and receives the incoming pots calls (for one 
line, duplicate it for the second x100p card), and hands the incoming call
to the "inbound-bus" context in extensions.conf, which then rings whatever
extension you've configured in step 6.

For everyone else reading this on the list that are more experienced then
I, feel free to add/change/delete steps as technically necessary. No
pride of authorship here. ;)

Rich