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

Vardan hvardan71 at gmail.com
Tue May 11 02:48:42 CDT 2010


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<hvardan71 at gmail.com>:
>> 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=0000890
>>
>> 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 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<alexrecarey at gmail.com>:
>>>>> 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:
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>>>>
>>>
>>>
>>>
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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