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<hvarda...@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<alexreca...@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: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- _____________________________________________________________________ -- 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
