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Jason Brown

2005-Jan-30 18:59 UTC

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

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@voipjet/${EXTEN},60)      ;voipjet
NANPA

exten => _011.,1,Dial(IAX2/402@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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el Flynn

2005-Jan-30 19:09 UTC

head link

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

Jason Brown wrote:> 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.
> <snip>

Just a guess about your problems, but if you have a PRI line incoming,
wouldn't
you need to connect it to a PRI card and not the TDM400P (which is for analog 
POTS lines)??

Flynn

Kevin P. Fleming

2005-Jan-30 19:09 UTC

head link

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

Jason Brown wrote:
> 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.
And why would you think those would act similarly? They don't.

Zap channels without ISDN or R2 signaling don't have a target extension 
to deliver to. They have no idea what phone number was called to make 
that channel ring, they only know the channel is ringing. They send the 
call to the "s" extension in whatever context you direct them to.

For now, you can simulate what you want by making each channel on the 
TDM400 go to a separate inbound context, and then using Goto() to go 
from there to the desired "incoming number" in your
"incoming" context.
With a PRI (which is what you would likely install in a real application 
of this type), you would actually receive the dialed number from the 
telco, and could route the call based on that.

david

2005-Jan-30 19:18 UTC

head link

[Asterisk-Users] Processing incoming calls with multiple contextstover PRI

Hi,Jason,
    The TDM400P card failed to get the Callee number or DID, so the * don't
know how to route the call. There are something difference between the analog
line and the PRI line.

    Regards.

    David
    http://www.iaxtalk.com

  ----- Original Message ----- 
  From: Jason Brown 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, January 31, 2005 9:59 AM
  Subject: [Asterisk-Users] Processing incoming calls with multiple
contextstover PRI


  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@voipjet/${EXTEN},60)      ;voipjet
NANPA

  exten => _011.,1,Dial(IAX2/402@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

  n       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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Lyle Giese

2005-Jan-30 19:20 UTC

head link

[Asterisk-Users] Processing incoming calls with multiple contextstover PRI

In zaptel.conf, put the line associated with 8350 in the context bpns-external
and when an external call comes in on 8350, it will drop to the s step in
bpns-external.  I would suggest that you do something with the call if they
don't bother to dial an extension, like send to a general voice mail box or
ring all phones then drop int the general voice mail.

Lyle

  ----- Original Message ----- 
  From: Jason Brown 
  To: asterisk-users@lists.digium.com 
  Sent: Sunday, January 30, 2005 7:59 PM
  Subject: [Asterisk-Users] Processing incoming calls with multiple
contextstover PRI


  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@voipjet/${EXTEN},60)      ;voipjet
NANPA

  exten => _011.,1,Dial(IAX2/402@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

  n       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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