Hello comunity,
I would like to ask you for help with Asterisk log parsing. I was trying to find some information about each attribute in my log, but I am not getting everything. Please help me match what is “communication method, source port, destination port, message” and any other attributes.
My log looks like:
asterisk[19738]: VERBOSE[18717]: app_dial.c:1331 in wait_for_answer: – SIP/1C17D340FF32_0-0000b1b1 is ringing
asterisk[19738]: WARNING[18717]: app_dial.c:2341 in dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:41] Dial(“SIP/trunk-1-0000b1b0”, “SIP/1C17D340FF32_0&SIP/sochor,hktw”) in new stack
asterisk[19738]: VERBOSE[18717]: netsock2.c:505 in ast_set_qos: == Using SIP RTP TOS bits 184
asterisk[19738]: VERBOSE[18717]: app_dial.c:2486 in dial_exec_full: – Called SIP/1C17D340FF32_0
asterisk[19738]: VERBOSE[18717]: netsock2.c:527 in ast_set_qos: == Using SIP RTP CoS mark 5
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:24] Set(“SIP/trunk-1-0000b1b0”, “GROUP_ID_DST=289”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:12] GotoIf(“SIP/trunk-1-0000b1b0”, “0?forward”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:9853 in pbx_builtin_goto: – Goto (phone-ext,374,18)
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:22] Set(“SIP/trunk-1-0000b1b0”, “EXT_ID_DST=1991”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:25] Set(“SIP/trunk-1-0000b1b0”, “CONNECTEDLINE(name,i)=Sochor Martin”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:11] Set(“SIP/trunk-1-0000b1b0”, “FORWARD_REASON=cfu”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:23] Set(“SIP/trunk-1-0000b1b0”, “CALL_DST=1”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [374@phone-ext:18] GotoIf(“SIP/trunk-1-0000b1b0”, “1?cdr”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:4240 in pbx_extension_helper: – Executing [record@functions:8] GotoIf(“SIP/trunk-1-0000b1b0”, “0?record”) in new stack
asterisk[19738]: VERBOSE[18717]: pbx.c:9853 in pbx_builtin_goto: – Goto (functions,record,5)
asterisk[19738]: VERBOSE[18717]: app_verbose.c:109 in verbose_exec: ** Processing call recording for extension ID 1991 **
thanks for help