[Asterisk-Users] Processing incoming calls with multiple contextst over PRI

[Jason Brown]

So I have a problem. A customer of mine wants a PBX, owns an office
building. I want to sell him on asterisk.  He has 4 tenants. I am using
my asterisk box to simulate it. My asterisk box has a TDM400P card, not
a PRI card. Don't know if it makes any difference.

 

Anyway, I want to route incoming phone calls to different contexts based
on the phone number being called.

 

Here is my extensions.conf

 

[incoming-calls]

exten => _4078698350,1,Goto,bpns-external|${EXTEN}|1

exten => _4078698353,1,Goto,demo1-external|${EXTEN}|1

exten => _4078698359,1,Goto,demo2-external|${EXTEN}|1

exten => _4078698360,1,Goto,demo3-external|${EXTEN}|1

 

[outgoing-calls]

exten => _407NXXXXXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _321NXXXXXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1800NXXXXXX,1,DIal(ZAP/g1/${EXTEN},60)

exten => _1866NXXXXXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1877NXXXXXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1888NXXXXXX,1,Dial(ZAP/g1/${EXTEN},60)

exten => _1NXXNXXXXXX,1,Dial(IAX2/402 at voipjet/${EXTEN},60)      ;voipjet
NANPA

exten => _011.,1,Dial(IAX2/402 at voipjet/${EXTEN},60)             ;voipjet
WORLD

 

[bpns-external]

exten => s,1,Playback,bpnsmenu

exten => 1,1,Dial(SIP/1003,20,tr)

exten => 1,2,Voicemail,u1003

exten => 1,102,Voicemail,b1003

exten => 2,1,Dial(SIP/1001,20,tr)

exten => 2,2,Voicemail,u1001

exten => 2,102,Voicemail,b1001

exten => 3,1,Dial(SIP/1002,20,tr)

exten => 3,2,VOicemail,u1002

exten => 3,102,Voicemail,b1002

exten => 1001,1,Dial(SIP/1001,20,tr)

exten => 1001,2,Voicemail,u1001

exten => 1001,102,VOicemail,b1002

exten => 1002,1,Dial(SIP/1002,20,tr)

exten => 1002,2,Voicemail,u1002

exten => 1002,102,Voicemail,b1002

exten => 1003,1,Dial(SIP/1003,20,tr)

exten => 1003,2,Voicemail,u1003

exten => 1003,102,Voicemail,b1003

exten => 8500,1,VoicemailMain

exten => t,1,Hangup

 

[bpns-internal]

include => outgoing-calls

exten => 1001,1,Dial(SIP/1001,20,tr)

exten => 1001,2,Voicemail,u1002

exten => 1001,102,Voicemail,b1002

exten => 1002,1,Dial(SIP/1002,20,tr)

exten => 1002,2,Voicemail,u1002

exten => 1002,102,Voicemail,u1002

exten => 1003,1,Dial(SIP/1003,20,tr)

exten => 1003,2,Voicemail,u1003

exten => 1003,103,Voicemail,b1003

exten => 1767,1,Dial(SIP/1001,20,tr)

exten => 1767,2,Voicemail,u1001

exten => 1767,102,Voicemail,b1001

exten => 8500,1,VoicemailMain

 

[demo1-external]

exten => s,1,Dial(SIP/1010,20,tr)

exten => s,2,Voicemail,u1010

exten => s,102,Voicemail,b1010

exten => 8500,1,VoicemailMain

 

[demo1-internal]

include => demo1-external

include => bpns-internal

include => outgoing-calls

 

[demo2-external]

exten => s,1,Dial(SIP/1030,20,tr)

exten => s,2,Voicemail,u1030

exten => s,102,Voicemail,b1030

exten => 8500,1,VoicemailMain

 

[demo2-internal]

include => demo2-external

include => bpns-internal

include => outgoing-calls

 

[demo3-external]

exten => s,1,Dial(SIP/2000,20,tr)

exten => s,2,Voicemail,u2000

exten => s,102,Voicemail,b2000

exten => 8500,1,VoicemailMain

 

[demo3-internal]

include => demo3-external

include => bpns-internal

include => outgoing-calls

 

It doesn't work. I have a couple asterisk guru friends who swear it
should work. Here is what asterisk tells me in verbose mode:

 

 

    -- Starting simple switch on 'Zap/1-1'

Jan 30 20:46:02 WARNING[7140]: chan_zap.c:5586 ss_thread: CallerID
returned with error on channel 'Zap/1-1'

  == Starting Zap/1-1 at incoming-calls,s,1 failed so falling back to
exten 's'

  == Starting Zap/1-1 at incoming-calls,s,1 still failed so falling back
to context 'default'

Jan 30 20:46:02 WARNING[7140]: pbx.c:1942 ast_pbx_run: Channel 'Zap/1-1'
sent into invalid extension 's' in context 'default', but no invalid
handler

*       Hungup 'Zap/1-1'  

 

Now I understand it is looking for the startup point. I don't understand
why. 2 other asterisk guys I know swear it's supposed to work, although
they are using sip/iax and not zap for input.

 

Anyone have any ideas?

 

Thanks

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