[asterisk-users] One server, multiple companies

Sun Dec 9 19:37:06 CST 2007

Eric C. wrote:
> Hello all, 
>
> Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies.
> So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using  
>
> exten => _X.,1,NoOp(FROM NUMBER:  ${SIP_HEADER(TO):5:10})
>
> to determine which number is being dialed by the caller and then using a gotoif to get to correct greeting (correct company).
>
> My question is... lets assume all three companies have extension numbers being 2000, 2001 & 2002, how does one separate them?
> Or, lets say the extensions are:
>
> company A --> 2000, 2001,2002
> company B --> 3000, 3001, 3002
> company C --> 4000, 4001, 4002
>
> Since they're on one server with one asterisk process, how can I use context correctly so that the user at 4002 cannot get through to the user at company A whose extension is 2000 as currently, I can dial 2000 from phone 4002.
>
> That's my current problem, how should this be setup?  Is my architecture correct? Should I be running different processes for each company? Can context resolve what I need?
>
> Please advise.
>
> thanks,
> Otto
>   

First off, *nuke* the default context in sip.conf, extensions.conf, and 
voicemail.conf ... it will just get you into trouble!

I do something like in my extensions.conf file:

[incoming]
exten => 2082292222,1,Goto(s,1,incoming-acme)
exten => 2082293333,1,Goto(s,1,incoming-fido)
exten => 2082294444,1,Goto(s,1,incoming-big-jims)
...

[incoming-acme]
exten => s,1,Answer()                                                          
exten => s,n,Wait(0.75)
exten => s,n(greeting),Playback(brief-directory-acme)               
exten => s,n(exten),Background(vm-enter-num-to-call)                           
exten => s,n,WaitExten(5)                                                      
exten => s,n(goodbye),Playback(vm-goodbye)                                     
exten => s,n(end),Hangup()                                                     

; these are the extensions that are exposed both to internal callers as
; well as to incoming calls... be careful what you put here.
include => extens-acme     

exten => i,1,Playback(pbx-invalid)                                             
exten => i,n,Goto(s,exten)                                                     

exten => t,1,Goto(s,goodbye)                  

[internal-acme]                                                                     
exten => s,1,Answer()             
exten => s,n(exten),Background(vm-enter-num-to-call)                           
exten => s,n,WaitExten(5)                                                      
exten => s,n(goodbye),Playback(vm-goodbye)                                     
exten => s,n(end),Hangup()                                                     

include => outbound-toll                                                       
include => outbound-local                                                      
include => extens-acme

; for our SIP phones, we can program a non-numeric extension                    
exten => voicemail,1,VoicemailMain(${CALLERIDNUM}@acme)
exten => voicemail,n,Hangup()

; and for DTMF coming through an ATA...
exten => 777,1,Goto(voicemail)

[extens-acme]
exten => 111,1,Macro(stdexten,111,${PHILIP})                                   
exten => 111,n,Goto(s,exten)
...

[outbound-local]                                                               
exten => _NXXXXXX,1,Dial(${TRUNK}/${AREA}${EXTEN},,r)                          
exten => _NXXXXXX,n,Congestion()                                               
exten => _NXXXXXX,n,Hangup()                                                   

[outbound-toll]
exten => _NXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN},,r)                              
exten => _NXXXXXXXXX,n,Congestion()                                            
exten => _NXXXXXXXXX,n,Hangup()                                                

exten => _011.,1,Dial(${TRUNK}/${EXTEN:3},,r)                                  
exten => _011.,n,Congestion()                                                  
exten => _011.,n,Hangup()                                                      

Note: we had to modify the stdexten macro to be:

[macro-stdexten];                                                              
;                                                                              
; Standard extension macro:                                                    
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well)      
;   ${ARG2} - Device(s) to ring                                                
;                                                                              
exten => s,1,Dial(${ARG2},20)                                   ; Ring the inter
exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump based on

exten => s-NOANSWER,1,Voicemail(u${ARG1})               ; If unavailable, send t
exten => s-NOANSWER,2,MacroExit()                       ; If they press #, retur

exten => s-BUSY,1,Voicemail(b${ARG1})                   ; If busy, send to voice
exten => s-BUSY,2,MacroExit()           ; If they press #, return to start     

exten => _s-.,1,Goto(s-NOANSWER,1)                              ; Treat anything

exten => a,1,VoicemailMain(${ARG1})

that is, we got rid of all instances of "Goto(s,1,default)".  The result 
is that if the caller bails out of voicemail or completes his message, 
the macro returns to the next priority.  Hence having:

exten => 111,1,Macro(stdexten,111,${PHILIP})                                   
exten => 111,n,Goto(s,exten)

you could also "Hangup()" as the 2nd priority...  You would need to have 
"exten" defined as a label for both the "incoming-acme" as well as 
"internal-acme" contexts (and they will probably do different things).

There are undoubtedly better ways to do this, but we find it's simple 
and maintainable and works well enough for us.

-Philip