Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread CW_ASN - Gus 
How? Is written in CDR?

Regards,

Gus

- Original Message - 
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1


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On Friday 30 January 2004 00:57, Eric Wieling wrote:
> Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
> being translated into Asterisk Cause Codes and being passed back to
> app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
> variable)?

HANGUPCAUSE is working fine here (cvs).

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread Tais M. Hansen 
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On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
>> HANGUPCAUSE is working fine here (cvs).
> How? Is written in CDR?

CDRs contain BUSY when busy and NO ANSWER on the rest.

extensions.conf:

[provider-out]
...
exten => _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten => _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)

[provider-out-failed]
exten => c1,1,Hangup()

exten => c2,1,Busy()

exten => c3,1,Answer()
exten => c3,2,ResetCDR()
exten => c3,3,Playtones(info)
exten => c3,4,Wait(60)
exten => c3,5,Hangup()

exten => c4,1,Congestion()

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread CW_ASN - Gus 
Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .

Regards,

Gus

- Original Message -
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 30, 2004 9:48 AM
Subject: Re: [Asterisk-Users] HANGUPCAUSE


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On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
>> HANGUPCAUSE is working fine here (cvs).
> How? Is written in CDR?

CDRs contain BUSY when busy and NO ANSWER on the rest.

extensions.conf:

[provider-out]
...
exten => _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten => _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)

[provider-out-failed]
exten => c1,1,Hangup()

exten => c2,1,Busy()

exten => c3,1,Answer()
exten => c3,2,ResetCDR()
exten => c3,3,Playtones(info)
exten => c3,4,Wait(60)
exten => c3,5,Hangup()

exten => c4,1,Congestion()

- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread Tais M. Hansen 
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On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
> Ok, but is not working as expected... we can't see clear ISUP causes. We
> can't make different treatments or store other causes than busy (cause=17)
> in cdr's .

You could use my approach and combine it with the CDR userfield. Personally I 
would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread Eric Wieling 
Personally I would like AST_CAUSE to be the Asterisk cause code (which
should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE,
SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people.

On Fri, 2004-01-30 at 09:57, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
> > Ok, but is not working as expected... we can't see clear ISUP causes. We
> > can't make different treatments or store other causes than busy (cause=17)
> > in cdr's .
> 
> You could use my approach and combine it with the CDR userfield. Personally I 
> would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.
> 
> - -- 
> Regards,
> Tais M. Hansen
> ComX Networks
> Tel: +45-70257474
> Fax: +45-70257374
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the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread Eric Wieling 
See Bug Number 890 on bugs.digium.com.

--Eric

> From: "Tais M. Hansen" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 30, 2004 9:20 AM
> Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1
> 
> 
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> On Friday 30 January 2004 00:57, Eric Wieling wrote:
> > Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
> > being translated into Asterisk Cause Codes and being passed back to
> > app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
> > variable)?
> 
> HANGUPCAUSE is working fine here (cvs).


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Re: [Asterisk-Users] HANGUPCAUSE

2004-01-30 Thread CW_ASN - Gus 
It would to be good in any way... :)


- Original Message -
From: "Tais M. Hansen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 30, 2004 12:57 PM
Subject: Re: [Asterisk-Users] HANGUPCAUSE


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On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
> Ok, but is not working as expected... we can't see clear ISUP causes. We
> can't make different treatments or store other causes than busy (cause=17)
> in cdr's .

You could use my approach and combine it with the CDR userfield. Personally
I
would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.

- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-02-02 Thread Tais M. Hansen 
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On Friday 30 January 2004 17:57, Eric Wieling wrote:
> Personally I would like AST_CAUSE to be the Asterisk cause code (which
> should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE,
> SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people.

I didn't know SIP or IAX actually defined causes for hangups. I any case one 
of us should make this a feature request on bugs.digium.com.

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Re: [Asterisk-Users] HANGUPCAUSE

2004-02-02 Thread Eric Wieling 
Over the weekend the problem with HANGUPCAUSE was fixed.  HANGUPCAUSE
now contains the Asterisk cause code.  README.variables has been
updated.  The problem with Dial not seeing a busy on PRI lines have been
fixed.  I have a version of app_dial.c from 0.7.1 that's been patched
with these two fixes.  The fixes are also in CVS.

Yay!

On Mon, 2004-02-02 at 03:57, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> On Friday 30 January 2004 17:57, Eric Wieling wrote:
> > Personally I would like AST_CAUSE to be the Asterisk cause code (which
> > should be the same for all technologies), TECH_CAUSE (IAX2_CAUSE,
> > SIP_CAUSE, PRI_CAUSE) would be interesting and useful to some people.
> 
> I didn't know SIP or IAX actually defined causes for hangups. I any case one 
> of us should make this a feature request on bugs.digium.com.
> 
> - -- 
> Regards,
> Tais M. Hansen
> ComX Networks
> Tel: +45-70257474
> Fax: +45-70257374
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Go to http://www.digium.com/index.php?menu=documentation and look at
the "Unofficial Links" section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
"Asterisk Resource Pages".

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] HANGUPCAUSE macro..

2004-10-26 Thread Eric Wieling 
IAX does not correctly set the HANGUPCAUSE for a LOT of things.  Look at 
DIALSTATUS or look at the dial-result macro on 
http://www.fnords.org/~eric/asterisk/downloads/macros.inc

Matt Schulte wrote:
I am connecting Asterisk to Asterisk to PSTN (Either by SIP or PRI) and
am having some issues dealing with busy signals. I have the HANGUPCAUSE
dial result macro in place to generate my hangup causes. I get a
hangupcause on my "gateway" machine with a code of 34, here's the code:
... -snip-
exten => hangupcause+34,1,Busy
It does in fact pass this on to the "IAD" asterisk machine as it saw the
following:
-- IAX2/x.x.x.x:4569/1 is busy
-- Hungup 'IAX2/x.x.x.x:4569/1'
  == Everyone is busy/congested at this time
-- Executing Macro("Zap/2-1", "dial-result") in new stack
-- Executing NoOp("Zap/2-1", "HANGUPCAUSE is 0") in new stack
The last line doesn't make sense to me, if it knows it's a busy signal
why on earth would it use code 0? (code 0 is "NOTDEFINED") Is there
another
way to handle these calls?
This is in my extensions.conf, as an example:
exten => _.,1,Dial(IAX2/:[EMAIL PROTECTED]/${EXTEN})
exten => _.,2,Macro(dial-result)
On priority 2 I want to avoid using "Busy", makes sense?

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RE: [Asterisk-Users] HANGUPCAUSE macro..

2004-10-26 Thread Matt Schulte 
Interesting, would this be considered a bug or is it rather intentional?
Or is that a dumb question ;-)

-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED] 

IAX does not correctly set the HANGUPCAUSE for a LOT of things.  Look at

DIALSTATUS or look at the dial-result macro on 
http://www.fnords.org/~eric/asterisk/downloads/macros.inc

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Re: [Asterisk-Users] HANGUPCAUSE macro..

2004-10-26 Thread Eric Wieling 
*I* consider it a bug.  Mark (if I recall correctly) considers it "just 
the way it works".

Matt Schulte wrote:
Interesting, would this be considered a bug or is it rather intentional?
Or is that a dumb question ;-)
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED] 

IAX does not correctly set the HANGUPCAUSE for a LOT of things.  Look at
DIALSTATUS or look at the dial-result macro on 
http://www.fnords.org/~eric/asterisk/downloads/macros.inc
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Re: [asterisk-users] ${HANGUPCAUSE} in CDR

2011-02-08 Thread faisal 



 ${HANGUPCAUSE} value is available on h extension.

-Original Message-
From: "Shariq Khan" 
Sent: Tuesday, February 8, 2011 8:30am
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] ${HANGUPCAUSE} in CDR

Hello Gurus,

Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I 
want to add the Hangup reason of call in userfield of CDR.

Regards,
Shariq Khan
0333-3501125
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Re: [asterisk-users] ${HANGUPCAUSE} in CDR

2011-02-08 Thread Steve Howes 
On 8 Feb 2011, at 13:30, Shariq Khan wrote:
> Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I 
> want to add the Hangup reason of call in userfield of CDR.

http://www.google.com/search?q=asterisk+hangupcause+cdr

Top result... Should do it

Steve


Steve Howes
SMTP to Google proxy Inc
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RE: [Asterisk-Users] Hangupcause is not enough on PRI

2006-04-04 Thread Peeramate @ SIPPhone Thailand 








Do not send me any more

 



Best Regards,

Mr.Peeramate Rochanasmita

Project Manager/General Manager

SIPphone (Thailand) Co., Ltd.
644/19 Moo 1 Klong Kum,
Bung Kum Bangkok Thailand 10230
SIP No.100888
SIP Call Center No.888
Tel. +66
2690 3999
Fax. +66
2690 3535
Mobile. +66
1423 1423
Email : [EMAIL PROTECTED]
MSN : [EMAIL PROTECTED]

Website :
www.sipphone.co.th











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pibix
Sent: Tuesday, April 04, 2006 7:48
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hangupcause is not enough on PRI



 

Hi,

 

I’m using Asterisk and a TE110P E1 PRI in Chile.

When I call to a disconnected number or any not operational
number, the telco sends the Hangupcause disconnection code and an audio message
notifying the disconnection cause to the user.

Asterisk does not allow the user to hear the audio message
form the telco, instead it cuts the call. Any other legacies PRI PBX I’ve
tested allow the user to hear the audio message from the telco.

A few months ago I was dealing with this problem (making the
user hear the disconnection cause message from the telco) and someone suggested
using the Hangupcause code (http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html),
and it solved the problem momentarily. Now, when I call to a not operational
number, depending on the Hangupcause variable, Asterisk plays an internal audio
message notifying the user about the disconnection cause, but my client is not
satisfied with that, he expect to hear the real audio messager form the telco.

 

I would like to know if somebody solved this issue letting
the user hear the real disconnection cause message form the telco.

 

Thank you!

 

Javier Ergas

CEO

Pibix.cl

 

 

 






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Re: [Asterisk-Users] Hangupcause is not enough on PRI

2006-04-04 Thread Andrew Kohlsmith 
On Tuesday 04 April 2006 08:47, Pibix wrote:
> Asterisk does not allow the user to hear the audio message form the telco,
> instead it cuts the call. Any other legacies PRI PBX I've tested allow the
> user to hear the audio message from the telco.

Can we see a pri debug trace for a call that does this?  I think this might be 
more of a signaling issue, as my T1 PRI (Bell Canada) does not do this at 
all.  Asterisk doesn't disconnect the call until it encounters something that 
issues a Hangup(), or the telco sends the DISCONNECT message.

-A.
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RE: [Asterisk-Users] Hangupcause is not enough on PRI

2006-04-04 Thread Wai Wu 
Interesting about your tellco. A the tellco I have dealt with sent the
DISCONNECT message when a non-operational number is called. The usual
messages will come in the this order

1. proceeding 
2. one or more progressing
3. disconnect with the cause value(if number is non-operational, or the
network or destination is busy)

On the other hand, upon receiving disconnect from tellco, Asterisk does
not really need to disconnect right away if listening to the telco
message is desired. (there might be a switch you can set for this).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, April 04, 2006 9:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Hangupcause is not enough on PRI

On Tuesday 04 April 2006 08:47, Pibix wrote:
> Asterisk does not allow the user to hear the audio message form the 
> telco, instead it cuts the call. Any other legacies PRI PBX I've 
> tested allow the user to hear the audio message from the telco.

Can we see a pri debug trace for a call that does this?  I think this
might be more of a signaling issue, as my T1 PRI (Bell Canada) does not
do this at all.  Asterisk doesn't disconnect the call until it
encounters something that issues a Hangup(), or the telco sends the
DISCONNECT message.

-A.
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Re: [Asterisk-Users] Hangupcause is not enough on PRI

2006-04-06 Thread Mojo with Horan & Company, LLC 

Record the telco's message and play it as your own :P

Pibix wrote:

Hi,

 


I’m using Asterisk and a TE110P E1 PRI in Chile.

When I call to a disconnected number or any not operational number, the 
telco sends the Hangupcause disconnection code and an audio message 
notifying the disconnection cause to the user.


Asterisk does not allow the user to hear the audio message form the 
telco, instead it cuts the call. Any other legacies PRI PBX I’ve tested 
allow the user to hear the audio message from the telco.


A few months ago I was dealing with this problem (making the user hear 
the disconnection cause message from the telco) and someone suggested 
using the Hangupcause code 
(http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html), 
and it solved the problem momentarily. Now, when I call to a not 
operational number, depending on the Hangupcause variable, Asterisk 
plays an internal audio message notifying the user about the 
disconnection cause, but my client is not satisfied with that, he expect 
to hear the real audio messager form the telco.


 

I would like to know if somebody solved this issue letting the user hear 
the real disconnection cause message form the telco.


 


Thank you!

 


Javier Ergas

CEO

Pibix.cl

 

 

 





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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-04-21 Thread Alejandro Recarey 
> However, as I can see by the verbose command, ${HANGUPCAUSE} is always
> 0. I thought it was a channel variable that contained the hangupcause?

Just an update, if the call is established, then there is a
hangupcause received.

The above problem only happens if the caller hangs up before pickup.

This is usualy a cause 16, not 0.

Alex

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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-10 Thread Zhang Shukun 
hi , all

i want to wtite hangupcause to cdr, but both caller hangup and
callee hangup result in hangupcause code 16.

how would i know whether caller or callee or system error hangup the phone?

please help.

thanks!

2010/4/22 Alejandro Recarey :
>> However, as I can see by the verbose command, ${HANGUPCAUSE} is always
>> 0. I thought it was a channel variable that contained the hangupcause?
>
> Just an update, if the call is established, then there is a
> hangupcause received.
>
> The above problem only happens if the caller hangs up before pickup.
>
> This is usualy a cause 16, not 0.
>
> Alex
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks for your supporting,
have a nice day.
Sucan

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Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-11 Thread Vardan 
Asterisk variable hangupcause
Page Contents

 * Asterisk variable Hangupcause
   o Recommended SIP <-> ISDN Cause codes (from RFC3398):
   o PRI Hangup Codes
   o Version notes
   o Tip
   o Examples
 + Example 1
 + Example 2
 + Example 3: Macro for handling hangupcause
 + Example 4: Set the hangup cause text to a variable
   o See also


Asterisk variable Hangupcause
Hangupcause is the latest PRI hangup return code on a zap channel 
connected to a PRI interface. Note that this also works on SIP channels, 
maybe other channels as well.
Tip: The packet isdnutils contains a utility called isdncause that 
provides a textual explanation of the error number that you feed it with 
(watch the entry format).

Previous to CVS 2004-08-12:

 From causes.h:
  #define AST_CAUSE_NOTDEFINED0
  #define AST_CAUSE_NORMAL1
  #define AST_CAUSE_BUSY  2
  #define AST_CAUSE_FAILURE   3
  #define AST_CAUSE_CONGESTION4
  #define AST_CAUSE_UNALLOCATED   5


For CVS head releases after 2004-08-12:

  /* Causes for disconnection (from Q.931) */
  #define AST_CAUSE_UNALLOCATED 1
  #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
  #define AST_CAUSE_NO_ROUTE_DESTINATION 3
  #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
  #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
  #define AST_CAUSE_NORMAL_CLEARING 16
  #define AST_CAUSE_USER_BUSY 17
  #define AST_CAUSE_NO_USER_RESPONSE 18
  #define AST_CAUSE_NO_ANSWER 19
  #define AST_CAUSE_CALL_REJECTED 21
  #define AST_CAUSE_NUMBER_CHANGED 22
  #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
  #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
  #define AST_CAUSE_FACILITY_REJECTED 29
  #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
  #define AST_CAUSE_NORMAL_UNSPECIFIED 31
  #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
  #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
  #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
  #define AST_CAUSE_SWITCH_CONGESTION 42
  #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
  #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
  #define AST_CAUSE_PRE_EMPTED 45
  #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
  #define AST_CAUSE_OUTGOING_CALL_BARRED  52
  #define AST_CAUSE_INCOMING_CALL_BARRED  54
  #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
  #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
  #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
  #define AST_CAUSE_CHAN_NOT_IMPLEMENTED  66
  #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED  69
  #define AST_CAUSE_INVALID_CALL_REFERENCE 81
  #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
  #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
  #define AST_CAUSE_MANDATORY_IE_MISSING 96
  #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
  #define AST_CAUSE_WRONG_MESSAGE 98
  #define AST_CAUSE_IE_NONEXIST 99
  #define AST_CAUSE_INVALID_IE_CONTENTS 100
  #define AST_CAUSE_WRONG_CALL_STATE 101
  #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
  #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
  #define AST_CAUSE_PROTOCOL_ERROR 111
  #define AST_CAUSE_INTERWORKING 127
  /* Special Asterisk aliases */
  #define AST_CAUSE_BUSY  AST_CAUSE_USER_BUSY
  #define AST_CAUSE_FAILURE  AST_CAUSE_NETWORK_OUT_OF_ORDER
  #define AST_CAUSE_NORMAL  AST_CAUSE_NORMAL_CLEARING
  #define AST_CAUSE_NOANSWER   AST_CAUSE_NO_ANSWER
  #define AST_CAUSE_CONGESTION   AST_CAUSE_NORMAL_CIRCUIT_CONGESTION
  #define AST_CAUSE_NOTDEFINED  0



Note: This does not work in 0.7.1 (maybe other versions) See: 
http://bugs.digium.com/bug_view_page.php?bug_id=890

Recommended SIP <-> ISDN Cause codes (from RFC3398):

   ISUP Cause valueSIP response
   
   1  unallocated number   404 Not Found
   2  no route to network  404 Not found
   3  no route to destination  404 Not found
   16 normal call clearing --- (*)
   17 user busy486 Busy here
   18 no user responding   408 Request Timeout
   19 no answer from the user  480 Temporarily unavailable
   20 subscriber absent480 Temporarily unavailable
   21 call rejected403 Forbidden (+)
   22 number changed (w/o diagnostic)  410 Gone
   22 number changed (w/ diagnostic)   301 Moved Permanently
   23 redirection to new destination   410 Gone
   26 non-selected user clearing   404 Not Found (=)
   27 destination out of order 502 Bad Gateway
   28 address incomplete   484 Address incomplete


Zhang Shukun wrote:
> hi , all
>
>  i want to wtite hangupcause to cdr, but both caller hangup and
> callee hangup result in hangupcause code 16.
>
> how would i know whether caller or callee or system error hangup the phone?
>
> please help.
>
> thanks!
>
> 2010/4/22 Alejandro Recarey:
>>> However, as I can see by the verbose command, ${HANGUPCAUSE} is a

Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-11 Thread Zhang Shukun 
thank you for reply.

but hangupcause cant different whether caller hangup or callee hangup?

above two situation both return 16.

2010/5/11 Vardan :
> Asterisk variable hangupcause
> Page Contents
>
>     * Asterisk variable Hangupcause
>           o Recommended SIP <-> ISDN Cause codes (from RFC3398):
>           o PRI Hangup Codes
>           o Version notes
>           o Tip
>           o Examples
>                 + Example 1
>                 + Example 2
>                 + Example 3: Macro for handling hangupcause
>                 + Example 4: Set the hangup cause text to a variable
>           o See also
>
>
> Asterisk variable Hangupcause
> Hangupcause is the latest PRI hangup return code on a zap channel
> connected to a PRI interface. Note that this also works on SIP channels,
> maybe other channels as well.
> Tip: The packet isdnutils contains a utility called isdncause that
> provides a textual explanation of the error number that you feed it with
> (watch the entry format).
>
> Previous to CVS 2004-08-12:
>
>  From causes.h:
>  #define AST_CAUSE_NOTDEFINED    0
>  #define AST_CAUSE_NORMAL        1
>  #define AST_CAUSE_BUSY          2
>  #define AST_CAUSE_FAILURE       3
>  #define AST_CAUSE_CONGESTION    4
>  #define AST_CAUSE_UNALLOCATED   5
>
>
> For CVS head releases after 2004-08-12:
>
>  /* Causes for disconnection (from Q.931) */
>  #define AST_CAUSE_UNALLOCATED 1
>  #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>  #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>  #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>  #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>  #define AST_CAUSE_NORMAL_CLEARING 16
>  #define AST_CAUSE_USER_BUSY 17
>  #define AST_CAUSE_NO_USER_RESPONSE 18
>  #define AST_CAUSE_NO_ANSWER 19
>  #define AST_CAUSE_CALL_REJECTED 21
>  #define AST_CAUSE_NUMBER_CHANGED 22
>  #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>  #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>  #define AST_CAUSE_FACILITY_REJECTED 29
>  #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>  #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>  #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>  #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>  #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>  #define AST_CAUSE_SWITCH_CONGESTION 42
>  #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
>  #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
>  #define AST_CAUSE_PRE_EMPTED 45
>  #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
>  #define AST_CAUSE_OUTGOING_CALL_BARRED      52
>  #define AST_CAUSE_INCOMING_CALL_BARRED      54
>  #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
>  #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL     58
>  #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
>  #define AST_CAUSE_CHAN_NOT_IMPLEMENTED      66
>  #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED      69
>  #define AST_CAUSE_INVALID_CALL_REFERENCE 81
>  #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
>  #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
>  #define AST_CAUSE_MANDATORY_IE_MISSING 96
>  #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
>  #define AST_CAUSE_WRONG_MESSAGE 98
>  #define AST_CAUSE_IE_NONEXIST 99
>  #define AST_CAUSE_INVALID_IE_CONTENTS 100
>  #define AST_CAUSE_WRONG_CALL_STATE 101
>  #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
>  #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
>  #define AST_CAUSE_PROTOCOL_ERROR 111
>  #define AST_CAUSE_INTERWORKING 127
>  /* Special Asterisk aliases */
>  #define AST_CAUSE_BUSY  AST_CAUSE_USER_BUSY
>  #define AST_CAUSE_FAILURE  AST_CAUSE_NETWORK_OUT_OF_ORDER
>  #define AST_CAUSE_NORMAL  AST_CAUSE_NORMAL_CLEARING
>  #define AST_CAUSE_NOANSWER   AST_CAUSE_NO_ANSWER
>  #define AST_CAUSE_CONGESTION   AST_CAUSE_NORMAL_CIRCUIT_CONGESTION
>  #define AST_CAUSE_NOTDEFINED  0
>
>
>
> Note: This does not work in 0.7.1 (maybe other versions) See:
> http://bugs.digium.com/bug_view_page.php?bug_id=890
>
> Recommended SIP <-> ISDN Cause codes (from RFC3398):
>
>   ISUP Cause value                        SIP response
>                           
>   1  unallocated number                   404 Not Found
>   2  no route to network                  404 Not found
>   3  no route to destination              404 Not found
>   16 normal call clearing                 --- (*)
>   17 user busy                            486 Busy here
>   18 no user responding                   408 Request Timeout
>   19 no answer from the user              480 Temporarily unavailable
>   20 subscriber absent                    480 Temporarily unavailable
>   21 call rejected                        403 Forbidden (+)
>   22 number changed (w/o diagnostic)      410 Gone
>   22 number changed (w/ diagnostic)       301 Moved Permanently
>   23 redirection to new destination       410 Gone
>   26 non-selected user clearing           404 Not Found (=)
>   27 destination out of order             502 Bad Gateway
>   28 address incomplete                   484 Address incomplete
>
>
> Zhang Shukun wrote:
>> hi , all
>>
>>      i want to wtite hangupcause to cdr, but

Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-11 Thread Vardan 
Can you show your dialplan part for that call and log also please

Thanks

Zhang Shukun wrote:
> thank you for reply.
>
> but hangupcause cant different whether caller hangup or callee hangup?
>
> above two situation both return 16.
>
> 2010/5/11 Vardan:
>> Asterisk variable hangupcause
>> Page Contents
>>
>>  * Asterisk variable Hangupcause
>>o Recommended SIP<->  ISDN Cause codes (from RFC3398):
>>o PRI Hangup Codes
>>o Version notes
>>o Tip
>>o Examples
>>  + Example 1
>>  + Example 2
>>  + Example 3: Macro for handling hangupcause
>>  + Example 4: Set the hangup cause text to a variable
>>o See also
>>
>>
>> Asterisk variable Hangupcause
>> Hangupcause is the latest PRI hangup return code on a zap channel
>> connected to a PRI interface. Note that this also works on SIP channels,
>> maybe other channels as well.
>> Tip: The packet isdnutils contains a utility called isdncause that
>> provides a textual explanation of the error number that you feed it with
>> (watch the entry format).
>>
>> Previous to CVS 2004-08-12:
>>
>>   From causes.h:
>>   #define AST_CAUSE_NOTDEFINED0
>>   #define AST_CAUSE_NORMAL1
>>   #define AST_CAUSE_BUSY  2
>>   #define AST_CAUSE_FAILURE   3
>>   #define AST_CAUSE_CONGESTION4
>>   #define AST_CAUSE_UNALLOCATED   5
>>
>>
>> For CVS head releases after 2004-08-12:
>>
>>   /* Causes for disconnection (from Q.931) */
>>   #define AST_CAUSE_UNALLOCATED 1
>>   #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>>   #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>>   #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>>   #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>>   #define AST_CAUSE_NORMAL_CLEARING 16
>>   #define AST_CAUSE_USER_BUSY 17
>>   #define AST_CAUSE_NO_USER_RESPONSE 18
>>   #define AST_CAUSE_NO_ANSWER 19
>>   #define AST_CAUSE_CALL_REJECTED 21
>>   #define AST_CAUSE_NUMBER_CHANGED 22
>>   #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>>   #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>>   #define AST_CAUSE_FACILITY_REJECTED 29
>>   #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>>   #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>>   #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>>   #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>>   #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>>   #define AST_CAUSE_SWITCH_CONGESTION 42
>>   #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
>>   #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
>>   #define AST_CAUSE_PRE_EMPTED 45
>>   #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
>>   #define AST_CAUSE_OUTGOING_CALL_BARRED  52
>>   #define AST_CAUSE_INCOMING_CALL_BARRED  54
>>   #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
>>   #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58
>>   #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
>>   #define AST_CAUSE_CHAN_NOT_IMPLEMENTED  66
>>   #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED  69
>>   #define AST_CAUSE_INVALID_CALL_REFERENCE 81
>>   #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
>>   #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
>>   #define AST_CAUSE_MANDATORY_IE_MISSING 96
>>   #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
>>   #define AST_CAUSE_WRONG_MESSAGE 98
>>   #define AST_CAUSE_IE_NONEXIST 99
>>   #define AST_CAUSE_INVALID_IE_CONTENTS 100
>>   #define AST_CAUSE_WRONG_CALL_STATE 101
>>   #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102
>>   #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103
>>   #define AST_CAUSE_PROTOCOL_ERROR 111
>>   #define AST_CAUSE_INTERWORKING 127
>>   /* Special Asterisk aliases */
>>   #define AST_CAUSE_BUSY  AST_CAUSE_USER_BUSY
>>   #define AST_CAUSE_FAILURE  AST_CAUSE_NETWORK_OUT_OF_ORDER
>>   #define AST_CAUSE_NORMAL  AST_CAUSE_NORMAL_CLEARING
>>   #define AST_CAUSE_NOANSWER   AST_CAUSE_NO_ANSWER
>>   #define AST_CAUSE_CONGESTION   AST_CAUSE_NORMAL_CIRCUIT_CONGESTION
>>   #define AST_CAUSE_NOTDEFINED  0
>>
>>
>>
>> Note: This does not work in 0.7.1 (maybe other versions) See:
>> http://bugs.digium.com/bug_view_page.php?bug_id=890
>>
>> Recommended SIP<->  ISDN Cause codes (from RFC3398):
>>
>>ISUP Cause valueSIP response
>>
>>1  unallocated number   404 Not Found
>>2  no route to network  404 Not found
>>3  no route to destination  404 Not found
>>16 normal call clearing --- (*)
>>17 user busy486 Busy here
>>18 no user responding   408 Request Timeout
>>19 no answer from the user  480 Temporarily unavailable
>>20 subscriber absent480 Temporarily unavailable
>>21 call rejected403 Forbidden (+)
>>22 number changed (w/o diagnostic)  410 Gone
>>22 number changed (w/ diagnostic)   301 Moved Permanently
>>23 redirection to new desti

Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-11 Thread Zhang Shukun 
this is dialplan:

exten => 123,1,Dial(SIP/1000,10,L(1))
exten => 123,2,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})

this is the log which hangup by caller:
== Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial("SIP/1001-0031",
"SIP/1000,10,L(1)") in new stack
-- Setting call duration limit to 10.000 seconds.
  == Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0032 is ringing
-- SIP/1000-0032 answered SIP/1001-0031
-- Executing [...@95040:1] Playback("SIP/1001-0031",
"vm-goodbye") in new stack
[May 11 17:23:16] WARNING[4258]: file.c:750 ast_readaudio_callback:
Failed to write frame
--  Playing 'vm-goodbye.gsm' (language 'en')
[May 11 17:23:16] WARNING[4258]: app_playback.c:471 playback_exec:
ast_streamfile failed on SIP/1001-0031 for vm-goodbye
-- Executing [...@95040:2] NoOp("SIP/1001-0031", "HANGUPCAUSE is
16") in new stack
  == Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0031'



this is the log which hangup by callee:

   == Using SIP RTP CoS mark 5
-- Executing [...@95040:1] Dial("SIP/1001-0033",
"SIP/1000,10,L(1)") in new stack
-- Setting call duration limit to 10.000 seconds.
  == Using SIP RTP CoS mark 5
-- Called 1000
-- SIP/1000-0034 is ringing
-- SIP/1000-0034 answered SIP/1001-0033
-- Executing [...@95040:1] Playback("SIP/1001-0033",
"vm-goodbye") in new stack
--  Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [...@95040:2] NoOp("SIP/1001-0033", "HANGUPCAUSE is
16") in new stack
  == Spawn extension (95040, 123, 1) exited non-zero on 'SIP/1001-0033'


2010/5/11 Vardan :
> Can you show your dialplan part for that call and log also please
>
> Thanks
>
> Zhang Shukun wrote:
>> thank you for reply.
>>
>> but hangupcause cant different whether caller hangup or callee hangup?
>>
>> above two situation both return 16.
>>
>> 2010/5/11 Vardan:
>>> Asterisk variable hangupcause
>>> Page Contents
>>>
>>>      * Asterisk variable Hangupcause
>>>            o Recommended SIP<->  ISDN Cause codes (from RFC3398):
>>>            o PRI Hangup Codes
>>>            o Version notes
>>>            o Tip
>>>            o Examples
>>>                  + Example 1
>>>                  + Example 2
>>>                  + Example 3: Macro for handling hangupcause
>>>                  + Example 4: Set the hangup cause text to a variable
>>>            o See also
>>>
>>>
>>> Asterisk variable Hangupcause
>>> Hangupcause is the latest PRI hangup return code on a zap channel
>>> connected to a PRI interface. Note that this also works on SIP channels,
>>> maybe other channels as well.
>>> Tip: The packet isdnutils contains a utility called isdncause that
>>> provides a textual explanation of the error number that you feed it with
>>> (watch the entry format).
>>>
>>> Previous to CVS 2004-08-12:
>>>
>>>   From causes.h:
>>>   #define AST_CAUSE_NOTDEFINED    0
>>>   #define AST_CAUSE_NORMAL        1
>>>   #define AST_CAUSE_BUSY          2
>>>   #define AST_CAUSE_FAILURE       3
>>>   #define AST_CAUSE_CONGESTION    4
>>>   #define AST_CAUSE_UNALLOCATED   5
>>>
>>>
>>> For CVS head releases after 2004-08-12:
>>>
>>>   /* Causes for disconnection (from Q.931) */
>>>   #define AST_CAUSE_UNALLOCATED 1
>>>   #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2
>>>   #define AST_CAUSE_NO_ROUTE_DESTINATION 3
>>>   #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6
>>>   #define AST_CAUSE_CALL_AWARDED_DELIVERED 7
>>>   #define AST_CAUSE_NORMAL_CLEARING 16
>>>   #define AST_CAUSE_USER_BUSY 17
>>>   #define AST_CAUSE_NO_USER_RESPONSE 18
>>>   #define AST_CAUSE_NO_ANSWER 19
>>>   #define AST_CAUSE_CALL_REJECTED 21
>>>   #define AST_CAUSE_NUMBER_CHANGED 22
>>>   #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27
>>>   #define AST_CAUSE_INVALID_NUMBER_FORMAT 28
>>>   #define AST_CAUSE_FACILITY_REJECTED 29
>>>   #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY 30
>>>   #define AST_CAUSE_NORMAL_UNSPECIFIED 31
>>>   #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34
>>>   #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38
>>>   #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41
>>>   #define AST_CAUSE_SWITCH_CONGESTION 42
>>>   #define AST_CAUSE_ACCESS_INFO_DISCARDED 43
>>>   #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL 44
>>>   #define AST_CAUSE_PRE_EMPTED 45
>>>   #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED   50
>>>   #define AST_CAUSE_OUTGOING_CALL_BARRED      52
>>>   #define AST_CAUSE_INCOMING_CALL_BARRED      54
>>>   #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57
>>>   #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL     58
>>>   #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65
>>>   #define AST_CAUSE_CHAN_NOT_IMPLEMENTED      66
>>>   #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED      69
>>>   #define AST_CAUSE_INVALID_CALL_REFERENCE 81
>>>   #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88
>>>   #define AST_CAUSE_INVALID_MSG_UNSPECIFIED   95
>>>   #define AST_CAUSE_MANDATORY_IE_MISSING 96
>>>   #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97
>>>  

Re: [asterisk-users] ${HANGUPCAUSE} is always 0 in the h extension

2010-05-11 Thread Vardan 
That is normal.
In any case, if the one of call party is hangup, then you take the 
hangupcase 16 - normal call clearing.

But if you want to see who is was terminatet the call, you must look in 
h in boot leg of the call, incoming and outgoing. and so you can see who 
is hangup the call.
I think you can understand what I have write. In this case I can see who 
has hang up call, and the billsec for both side.
This is with FailOver call.


Look my dial plan in ael, I use this with this call out files:
==
Channel: local/calltocusto...@c2c_c2customerviavendors/n
Callerid: 18185402021
WaitTime: 45
MaxRetries: 0
RetryTime: 10
Account: AGENT1
Context: c2c_C2AgentsVIAVendors
Extension: 18185402021
Priority: 1
Setvar: CUSTOMERPEER="ToVendor3-ToVendor4"
Setvar: AGENTPEER="ToVendor1-ToVendor2"
Setvar: CUSTOMERPHONE=37410219709
Setvar: AGENTPHONE=18185402021
Setvar: AGENTTECH=SIP
Setvar: CUSTOMERTECH=SIP
Archive: yes



Dialplan in AEL:
===
context c2c_C2CustomerVIASIP {

 _CallToCustomer => {
 Dial(SIP/${custom...@${exten});
 Hangup;
 };

};

context c2c_C2CustomerVIAVendors {

 _CallToCustomer => {
 CALLERID(all)=${AGENTPHONE}; Noop(${CALLERID(number)});
 
&FailOverDial(${CUSTOMERTECH},${CUSTOMERPEER},${CUSTOMERPHONE},${CONTEXT});
 Noop(CustomerVIAVerdors == return from macro);
 Hangup;
 };

 h => { Noop(Hangup in c2c_C2CustomerVIAVendors );
 Noop(: Duration of the call once it was answered.${CDR(billsec)});
 };
};


context c2c_C2AgentsVIASIP {

 _CallToCustomer => {
 Dial(SIP/${age...@${exten});
 Hangup;
 };

};

context c2c_C2AgentsVIAVendors {

 _CallToCustomer => {
 CALLERID(all)=${CUSTOMERPHONE}; Noop(${CALLERID(number)});
 &FailOverDial(${AGENTTECH},${AGENTPEER},${AGENTPHONE},${CONTEXT});
 Noop(AgentsVIAVendors === return from macro);
 Hangup;
 };
 _X. => {
 CALLERID(all)=${CUSTOMERPHONE}; Noop(${CALLERID(number)});
 &FailOverDial(${AGENTTECH},${AGENTPEER},${AGENTPHONE},${CONTEXT});
 Noop(AgentsVIAVendors === return from macro);
 Hangup;
 };

 h => { Noop(Hangup in c2c_AgentsVIAVendors );
 Noop(: Duration of the call once it was answered.${CDR(billsec)});
 Hangup;
 };

};


macro FailOverDial (tech,peers,phone,cont) {
 i=1; focount=0;
 Set(j=${CUT(peers,,${i})});
 leng=${LEN(${j})};
 noop(Tech:${tech}); noop(Peers:${peers}); noop(Phone:${phone}); 
noop(J:${j});

 while ( ${leng} > 0 ) {
 Noop(${tech}/${pho...@${j});
 Dial(${tech}/${pho...@${j});
 Noop(Dial status:${DIALSTATUS});
 Noop(End dial command);

 switch(${DIALSTATUS}) {
 case BUSY:
 Noop(Busy);
 return;

 case CHANUNAVAIL:
 Noop(Channel unavailble);
 i=${i}+1;
 Set(j=${CUT(peers,,${i})});
 leng=${LEN(${j})};
 noop(J:${j});
 break;

 case NOANSWER:
 Noop(No answer);
 return;

 case CANCEL:
 Noop(Cancel);
 return;

 case CONGESTION:
 Noop(Congestion);
 i=${i}+1;
 Set(j=${CUT(peers,,${i})});
 leng=${LEN(${j})};
 break;

 case ANSWER:
 Noop( Answer);
 return;

 default:
 Noop(Defaul);
 return;
 };
 };
 catch h {
 Noop(Hagup in macro);
 Noop(: Duration of the call once it was answered.${CDR(billsec)});
 Noop(Hangup for context:${cont});
 &ShowCDRDetails();
 Hangup;
 return;
 };
 return;
};

macro ShowCDRDetails () {


 Noop(Hangup in macro);
 Noop(: The channel's account code:${CDR(accountcode)});
 Noop(: DOCUMENTATION, BILL, IGNORE etc:${CDR(amaflags)});
 Noop(: Time the call was answered:${CDR(answer)});
 Noop(: Duration of the call once it was answered.${CDR(billsec)});
 Noop(: Channel name:${CDR(channel)});
 Noop(: Caller ID:${CDR(clid)});
 Noop(: Destination context:${CDR(dcontext)});
 Noop(: ANSWERED, NO ANSWER, BUSY:${CDR(disposition)});
 Noop(: Destination:${CDR(dst)});
 Noop(: Destination channel:${CDR(dstchannel)});
 Noop(: Duration of the call:${CDR(duration)});
 Noop(: Time the call ended:${CDR(end)});
 Noop(: Last app executed:${CDR(lastapp)});
 Noop(: Last app's arguments:${CDR(lastdata)});
 Noop(: Source:${CDR(src)});
 Noop(: Time the call started:${CDR(start)});
 Noop(: The channel's unique id:${CDR(uniqueid)});
 return;
};




Zhang Shukun wrote:
> this is dialplan:
>