Which Asterisk version do you use ? And what is your exact question ? What do you expect queue_log file to hold ?
2016-03-15 10:33 GMT+01:00 Антон Сацкий <[email protected]>: > > Hi list need your help > i have call in queue it shows that it was answered by 4003 > ============================ > [root@asterisk ~]# grep --color "1456128646.157422" > /var/log/asterisk/queue_log-20160228 > > 1456128688|1456128646.157422|800|NONE|ENTERQUEUE||0967145750|2 > 1456128717|1456128646.157422|800|SIP/4003|CONNECT|29|1456128688.157426|28 > 1456128817|1456128646.157422|800|SIP/4003|COMPLETECALLER|29|100|2 > > ============================ > BUT IN FACT call was PICK UPPED by 4001 using features > > [root@asterisk ~]# grep --color "1456128646.157422" > /var/log/asterisk/full-20160228 > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:2] MSet("SIP/3590640-000209b9", > "CDR(recordingfile)=3590640_1456128646.157422") in new stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:3] MixMonitor("SIP/3590640-000209b9", > "3590640_1456128646.157422.wav,b") in new stack > > > > [root@asterisk ~]# grep --color "C-0000f165" > /var/log/asterisk/full-20160228 > [Feb 22 10:10:46] VERBOSE[2070][C-0000f165] netsock2.c: == Using SIP RTP > CoS mark 5 > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:1] Set("SIP/3590640-000209b9", "CALLERID(name)=RU") in > new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:2] GotoIfTime("SIP/3590640-000209b9", > "9:00-19:30,mon-fri,*,*?4") in new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Goto > (incoming,3590640,4) > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:4] Goto("SIP/3590640-000209b9", "working") in new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Goto > (incoming,3590640,13) > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:13] Progress("SIP/3590640-000209b9", "") in new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:14] MSet("SIP/3590640-000209b9", "EXT=3590640") in new > stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:15] Set("SIP/3590640-000209b9", "CHANNEL(language)=ru") > in new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:16] Playback("SIP/3590640-000209b9", > "01_HELLO/01_HELLO") in new stack > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] res_rtp_asterisk.c: > > 0x7f9b1c19d490 -- Probation passed - setting RTP source address to > 95.67.3.3:14380 > [Feb 22 10:10:46] VERBOSE[9760][C-0000f165] file.c: -- > <SIP/3590640-000209b9> Playing '01_HELLO/01_HELLO.slin' (language 'ru') > [Feb 22 10:10:49] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:17] Wait("SIP/3590640-000209b9", "2") in new stack > [Feb 22 10:10:51] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@incoming:18] BackGround("SIP/3590640-000209b9", > "02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES") in new stack > [Feb 22 10:10:51] VERBOSE[9760][C-0000f165] file.c: -- > <SIP/3590640-000209b9> Playing > '02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES.slin' (language 'ru') > [Feb 22 10:10:55] DTMF[9760][C-0000f165] channel.c: DTMF begin '2' > received on SIP/3590640-000209b9 > [Feb 22 10:10:55] DTMF[9760][C-0000f165] channel.c: DTMF begin ignored '2' > on SIP/3590640-000209b9 > [Feb 22 10:10:55] DTMF[9760][C-0000f165] channel.c: DTMF end '2' received > on SIP/3590640-000209b9, duration 260 ms > [Feb 22 10:10:55] DTMF[9760][C-0000f165] channel.c: DTMF end passthrough > '2' on SIP/3590640-000209b9 > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: == CDR updated on > SIP/3590640-000209b9 > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [2@incoming:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua") in new > stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [2@incoming:2] Set("SIP/3590640-000209b9", "CALLERID(name)=UA") in new > stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [2@incoming:3] Goto("SIP/3590640-000209b9", "ua_start,3590640,1") in new > stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Goto > (ua_start,3590640,1) > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_start:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua") > in new stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_start:2] Set("SIP/3590640-000209b9", "TIMEOUT(digit)=3") in > new stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] func_timeout.c: -- Digit > timeout set to 3.000 > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_start:3] BackGround("SIP/3590640-000209b9", > "01_QUALITY_OF_THE_SERVICE/01_QUALITY_OF_THE_SERVICE") in new stack > [Feb 22 10:11:00] VERBOSE[9760][C-0000f165] file.c: -- > <SIP/3590640-000209b9> Playing > '01_QUALITY_OF_THE_SERVICE/01_QUALITY_OF_THE_SERVICE.slin' (language 'ua') > [Feb 22 10:11:22] DTMF[9760][C-0000f165] channel.c: DTMF begin '3' > received on SIP/3590640-000209b9 > [Feb 22 10:11:22] DTMF[9760][C-0000f165] channel.c: DTMF begin ignored '3' > on SIP/3590640-000209b9 > [Feb 22 10:11:22] DTMF[9760][C-0000f165] channel.c: DTMF end '3' received > on SIP/3590640-000209b9, duration 240 ms > [Feb 22 10:11:22] DTMF[9760][C-0000f165] channel.c: DTMF end passthrough > '3' on SIP/3590640-000209b9 > [Feb 22 10:11:25] VERBOSE[9760][C-0000f165] pbx.c: == CDR updated on > SIP/3590640-000209b9 > [Feb 22 10:11:25] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3@ua_start:1] Goto("SIP/3590640-000209b9", "ua_step1_3,3590640,1") in > new stack > [Feb 22 10:11:25] VERBOSE[9760][C-0000f165] pbx.c: -- Goto > (ua_step1_3,3590640,1) > [Feb 22 10:11:25] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_step1_3:1] BackGround("SIP/3590640-000209b9", > "10_STAY_ONLINE_PLEASE/10_STAY_ONLINE_PLEASE") in new stack > [Feb 22 10:11:25] VERBOSE[9760][C-0000f165] file.c: -- > <SIP/3590640-000209b9> Playing > '10_STAY_ONLINE_PLEASE/10_STAY_ONLINE_PLEASE.slin' (language 'ua') > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_step1_3:2] Gosub("SIP/3590640-000209b9", > "mix,~~s~~,1(3590640)") in new stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:1] MSet("SIP/3590640-000209b9", "LOCAL(EXT)=3590640") in new > stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:2] MSet("SIP/3590640-000209b9", > "CDR(recordingfile)=3590640_1456128646.157422") in new stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:3] MixMonitor("SIP/3590640-000209b9", > "3590640_1456128646.157422.wav,b") in new stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [~~s~~@mix:4] Return("SIP/3590640-000209b9", "") in new stack > [Feb 22 10:11:28] VERBOSE[9775][C-0000f165] app_mixmonitor.c: == Begin > MixMonitor Recording SIP/3590640-000209b9 > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] pbx.c: -- Executing > [3590640@ua_step1_3:3] Queue("SIP/3590640-000209b9", "800,Xxt") in new > stack > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] res_musiconhold.c: -- > Started music on hold, class 'default', on SIP/3590640-000209b9 > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] netsock2.c: == Using SIP RTP > CoS mark 5 > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] app_queue.c: -- Called > SIP/4003 > [Feb 22 10:11:28] VERBOSE[9760][C-0000f165] app_queue.c: -- > SIP/4003-000209bd is ringing > [Feb 22 10:11:57] VERBOSE[9760][C-0000f165] app_queue.c: -- > *SIP/4001-000209c3 > answered SIP/3590640-000209b9* > [Feb 22 10:11:57] VERBOSE[9760][C-0000f165] res_musiconhold.c: -- > Stopped music on hold on SIP/3590640-000209b9 > [Feb 22 10:13:37] VERBOSE[9760][C-0000f165] pbx.c: == Spawn extension > (ua_step1_3, 3590640, 3) exited non-zero on 'SIP/3590640-000209b9' > [Feb 22 10:13:37] VERBOSE[9775][C-0000f165] app_mixmonitor.c: == > MixMonitor close filestream (mixed) > [Feb 22 10:13:37] VERBOSE[9775][C-0000f165] app_mixmonitor.c: == End > MixMonitor Recording SIP/3590640-000209b9 > > > > > > My features > *8 PICKUP > > > -- > Best regards > Antony > моб (066) 919-75-33 > моб (063) 656-43-40 > [email protected] <mail%[email protected]> > > -- > _____________________________________________________________________ > -- 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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