RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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RE: [Asterisk-Users] Help with extensions
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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Re: [Asterisk-Users] Help with extensions
You read my mind. I was about to compose a response asking
why you'd strip digits when this message arrived.
Thanks.
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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Re: [Asterisk-Users] Help with extensions
Nope. Didn't work. When the call hits extensions.conf no matching
extension definition exists. I tried replacing line #2,
exten => s,2,VoiceMail2(${EXTEN}) with exten =>
_u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
in the sip debug. It seems like this pattern matching should solve
the problem. It is unclear why the pattern is not matched. Is the
syntax correct?
-Steve
David J Carter wrote:
Steve,
Sorry bum information. Line 2 should read: -
exten => s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a "u" or "b" based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]
If I hard code the following rules then calls get forwarded
as expected.
exten => u67501,1,VoiceMail2(${EXTEN})
exten => #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
exten => #,2,Hangup
Thanks
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RE: [Asterisk-Users] Help with extensions
I have had the same issue when receiving a call from an IAX provider.
Here is what I did to solve it.
[from-proxy]
exten => .,1,Goto(voicemail-direct,s,1)
[voicemail-direct]
exten => s,1,Answer
exten => s,2,VoiceMail2(${EXTEN:1})
exten => 3,3,Hangup
Not sure it does not pattern match like it should be the above worked
for me.
Robert
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Blair
> Sent: Sunday, February 06, 2005 8:43 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help with extensions
>
>
> Nope. Didn't work. When the call hits extensions.conf no
> matching extension definition exists. I tried replacing line
> #2, exten => s,2,VoiceMail2(${EXTEN}) with exten =>
> _u[3678].,2,VoiceMail2(${EXTEN}) without success. Same error
> in the sip debug. It seems like this pattern matching should
> solve the problem. It is unclear why the pattern is not
> matched. Is the syntax correct?
>
> -Steve
>
> David J Carter wrote:
>
> >Steve,
> >
> >Sorry bum information. Line 2 should read: -
> >
> >exten => s,2,VoiceMail2(${EXTEN})
> >
> >Don't need to strip the first digit as this is either u or b
> already,
> >(Unobtainable or Busy).
> >
> >Regards
> >
> >Dave
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of David J
> >Carter
> >Sent: 06 February 2005 12:30
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: RE: [Asterisk-Users] Help with extensions
> >
> >
> >Steve,
> >
> >I haven't tried this but can't you do something like.
> >
> >[from-proxy]
> >exten => s,1,Answer
> >exten => s,2,VoiceMail2(${EXTEN:1})
> >exten => 3,3,Hangup
> >
> >Regards
> >
> >Dave
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of Steve
> >Blair
> >Sent: 06 February 2005 12:14
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: [Asterisk-Users] Help with extensions
> >
> >
> >
> >Hello:
> >
> > I'd like some help with defining extension rules. I want calls
> >arriving at Asterisk from my SIP proxy to be sent directly to
> >voicemail. I'd also like the appropriate greeting played
> when the call
> >gets to voicemail.
> >
> > My proxy prefixes the extension with a "u" or "b" based on SIP
> >response codes before relaying to Asterisk. So when the call
> arrives it
> >is in the format [u|b][3|6|7|8]
> >
> > If I hard code the following rules then calls get forwarded as
> >expected.
> > exten => u67501,1,VoiceMail2(${EXTEN})
> > exten => #,2,Hangup
> >
> > However to save on typing I'd like a general rule. I've tried the
> >following but Asterisk cannot find the extension with this set of
> >rules. Can someone explain how what I want can be accomplished?
> >
> >exten => _[ub][3678].,2,VoiceMail2(${EXTEN})
> >exten => #,2,Hangup
> >
> >Thanks
> >___
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> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] Help with extensions - can't dial 700
Just a shot in the dark: do you have features.conf with the following lines in it? Could be that asterisk's sample config file got in the way... Call Parking defaults to using extension 700: [general] parkext => 700 Mojo Chris Mason wrote: I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. From a SIP phone I can dial 700 and all other extensions. From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I dial 700. Extensions.conf [office] exten => 700,1,Dial(SIP/700,20) exten => 700,2,VoiceMail,u700 exten => 700,102,VoiceMail,b700 exten => 701,1,Dial(SIP/701,20) exten => 701,2,VoiceMail,u701 exten => 701,102,VoiceMail,b701 exten => 702,1,Dial(SIP/702,20) exten => 702,2,VoiceMail,u702 exten => 702,102,VoiceMail,b702 exten => 703,1,Dial(SIP/703,20) exten => 703,2,VoiceMail,u703 exten => 703,102,VoiceMail,b703 ... [zap-in] exten => s,1,NoOp(Starting Zap-in) include => office == zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes context=zap-in group=1 signalling=fxo_ks faxdetect=incoming channel => 1-12 Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions - can't dial 700
Thanks, I removed that and will test. I don't have an analog extension here, I am testing using SIP remotely, will have to go to the resort to test. Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions - can't dial 700
Check features.conf for parking extensions. There default is 700 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason Sent: Monday, May 16, 2005 8:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Help with extensions - can't dial 700 I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. >From a SIP phone I can dial 700 and all other extensions. >From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I dial 700. Extensions.conf [office] exten => 700,1,Dial(SIP/700,20) exten => 700,2,VoiceMail,u700 exten => 700,102,VoiceMail,b700 exten => 701,1,Dial(SIP/701,20) exten => 701,2,VoiceMail,u701 exten => 701,102,VoiceMail,b701 exten => 702,1,Dial(SIP/702,20) exten => 702,2,VoiceMail,u702 exten => 702,102,VoiceMail,b702 exten => 703,1,Dial(SIP/703,20) exten => 703,2,VoiceMail,u703 exten => 703,102,VoiceMail,b703 .. [zap-in] exten => s,1,NoOp(Starting Zap-in) include => office == zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes context=zap-in group=1 signalling=fxo_ks faxdetect=incoming channel => 1-12 Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with extensions - can't dial 700
Chris, Don't forget that a change in features.conf requires a restart of asterisk (or the modual features.c) - you can't get away with just a reload. On 5/17/05, Chris Mason <[EMAIL PROTECTED]> wrote: > Thanks, I removed that and will test. I don't have an analog extension here, > I am testing using SIP remotely, will have to go to the resort to test. > > Chris Mason > US Number: (646)722-0001 US Fax (815)301-9759 > Skype: netconcepts > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
