Все линии заняты при попытке позвонить
Установил elastix, сразу обновил. Зашел в PBX. Добавил двух абонентов с внутренними номерами. Телефон взят за основу CISCO CP-3905. Так вот сделал номера 100 и 101. По внутренней связи звонки проходят отлично.
Добавил транк. Результат команд:
| Код: |
| elastix*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 100/100 192.168.3.200 D Yes Yes A 5060 OK (8 ms) 101/100 192.168.3.201 D Yes Yes A 5060 OK (8 ms) имя/логин 62.148.237.152 Yes Yes 5060 OK (50 ms) 3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline] |
и
| Код: |
| elastix*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time srgngn.usi.ru:5060 N логин 103 Registered Mon, 07 Jul 2014 10:38:18 1 SIP registrations. |
добавил к транку outbound и inbound маршруты. у outbound в Dial Patterns that will use this Route добавил 8. в match pattern в inbound DID Number номер без 8, но с кодом города. В Set Destination Extensions на 1 из аппаратов. При звонке на любой номер начинающийся с 8 выдает "На данный момент все линии заняты...". Ну а если набирать не с 8 соответственно говорит что недоступно, но это понятно - не указано правило. Подскажите почему такое сообщение выводится? На роутере порты открывал, даже ДМЗ включал. Все равно результат тот же. Ещё был сервер Ip телефонии на Oktell, но и его временно отключал, никаких изменений.
| Код: |
| Connected to Asterisk 11.10.0 currently running on elastix (pid = 3174) -- Remote UNIX connection -- Remote UNIX connection disconnected == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [83462980925@from-internal:1] Macro("SIP/100-00000002", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/100-00000002", "AMPUSER=100") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-00000002", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-00000002", "1?Set(REALCALLERIDNUM=100)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/100-00000002", "AMPUSER=100") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/100-00000002", "AMPUSERCIDNAME=Reseption") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000002", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/100-00000002", "AMPUSERCID=100") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/100-00000002", "CALLERID(all)="Reseption" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-00000002", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-00000002", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/100-00000002", "CALLERID(number)=100") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/100-00000002", "CALLERID(name)=Reseption") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/100-00000002", "Using CallerID "Reseption" ") in new stack -- Executing [83462980925@from-internal:2] NoOp("SIP/100-00000002", "Calling Out Route: outbound") in new stack -- Executing [83462980925@from-internal:3] Set("SIP/100-00000002", "MOHCLASS=default") in new stack -- Executing [83462980925@from-internal:4] Set("SIP/100-00000002", "_NODEST=") in new stack -- Executing [83462980925@from-internal:5] Macro("SIP/100-00000002", "record-enable,100,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/100-00000002", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/100-00000002", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/100-00000002", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/100-00000002", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("SIP/100-00000002", "1?MacroExit()") in new stack -- Executing [83462980925@from-internal:6] Macro("SIP/100-00000002", "dialout-trunk,2,83462980925,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000002", "DIAL_TRUNK=2") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000002", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000002", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000002", "DIAL_NUMBER=83462980925") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000002", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000002", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000002", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000002", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000002", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000002", "outbound-callerid,2") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000002", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000002", "0?Set(REALCALLERIDNUM=100)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-00000002", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/100-00000002", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000002", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000002", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-00000002", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-00000002", "1?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-00000002", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-00000002", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000002", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000002", "0?sub-flp-2,s,1") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000002", "OUTNUM=83462980925") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000002", "custom=SIP/dom1") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000002", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-00000002", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000002", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-00000002", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000002", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000002", "SIP/dom1/83462980925,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/dom1/83462980925 == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/100-00000002", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack -- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-00000002", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/100-00000002", "RC=21") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/100-00000002", "21,1") in new stack -- Goto (macro-dialout-trunk,21,1) -- Executing [21@macro-dialout-trunk:1] Goto("SIP/100-00000002", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/100-00000002", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-00000002", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/100-00000002", "CALLERID(number)=100") in new stack -- Executing [83462980925@from-internal:7] Macro("SIP/100-00000002", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/100-00000002", "") in new stack -- Executing [s@macro-outisbusy:2] GotoIf("SIP/100-00000002", "0?emergency,1") in new stack -- Executing [s@macro-outisbusy:3] GotoIf("SIP/100-00000002", "0?intracompany,1") in new stack -- Executing [s@macro-outisbusy:4] Playback("SIP/100-00000002", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack -- Playing 'all-circuits-busy-now.slin' (language 'ru') > 0x1d347e10 -- Probation passed - setting RTP source address to 192.168.3.200:16388 -- Playing 'pls-try-call-later.slin' (language 'ru') -- Executing [s@macro-outisbusy:5] Congestion("SIP/100-00000002", "20") in new stack == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/100-00000002' in macro 'outisbusy' == Spawn extension (from-internal, 83462980925, 7) exited non-zero on 'SIP/100-00000002' -- Executing [h@from-internal:1] Macro("SIP/100-00000002", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000002", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000002", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000002", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000002", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000002", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000002", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000002", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000002", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000002", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000002", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000002", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/100-00000002", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/100-00000002", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/100-00000002' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000002' |
строка
| Код: |
| -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/100-00000002", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack |
и
| Код: |
| -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-00000002", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack |
явно как бы намекают, что ошибка в них, но либо я плохо гуглил, либо.. вообщем не совсем могу понять почему так
Added after 2 hours 50 minutes:
появились входящие. исходящих нету.
| Код: |
| -- Called SIP/dom1/83462980925 == Everyone is busy/congested at this time (1:0/0/1) |
Обратитесь к вашему оператору и уточните формат набора номера. Скорее всего вы неправильно набираете номер.
| Код: |
| -- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000002", "SIP/dom1/83462980925,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/dom1/83462980925 == Everyone is busy/congested at this time (1:0/0/1) |
Вот теперь сделайте sip set debug ip - вашего прова и внимательно прочтите лог ваших обменов с ним.
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P4 3.0 + 1Gb CentOS 5.8 Aster 11.25.1
Не люблю gui-сборки: натуральный продукт вкуснее.
И еще: я ПРОФИ так как НЕ ЛЕНЮСЬ читать литературу.
ну наверно потому, что новички нифига не понимают по логам. и боятся упустить часть кода.
я поэтому выложу всё что дала команда - и надеюсь что это будет последний раз
| Код: |
| elastix*CLI> sip set debug ip 62.148.237.152 SIP Debugging Enabled for IP: 62.148.237.152 Reliably Transmitting (NAT) to 62.148.237.152:5060: OPTIONS sip:62.148.237.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK5cbbe1a4;rport Max-Forwards: 70 From: "Unknown" ;tag=as675e042f To: Contact: Call-ID: 0b00e83b2004b34401f307a53a7cf66a@192.168.3.6:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.8.1(11.10.0) Date: Tue, 08 Jul 2014 09:43:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- Reliably Transmitting (NAT) to 62.148.237.152:5060: OPTIONS sip:62.148.237.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK36ab33a5;rport Max-Forwards: 70 From: "Unknown" ;tag=as2bf83a16 To: Contact: Call-ID: 1c582cc06b2523c92524877457d9fadb@192.168.3.6:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.8.1(11.10.0) Date: Tue, 08 Jul 2014 09:43:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- SIP/2.0 404 Not Found From: "Unknown";tag=as675e042f To: ;tag=331891035 Call-ID: 0b00e83b2004b34401f307a53a7cf66a@192.168.3.6:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK5cbbe1a4 contact: supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join Content-Length: 0 --- (9 headers 0 lines) --- Really destroying SIP dialog '0b00e83b2004b34401f307a53a7cf66a@192.168.3.6:5060' Method: OPTIONS SIP/2.0 404 Not Found From: "Unknown";tag=as2bf83a16 To: ;tag=1177763289 Call-ID: 1c582cc06b2523c92524877457d9fadb@192.168.3.6:5060 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK36ab33a5 contact: supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join Content-Length: 0 --- (9 headers 0 lines) --- Really destroying SIP dialog '1c582cc06b2523c92524877457d9fadb@192.168.3.6:5060' Method: OPTIONS == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [980925@from-internal:1] Macro("SIP/100-00000049", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/100-00000049", "AMPUSER=100") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-00000049", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-00000049", "1?Set(REALCALLERIDNUM=100)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/100-00000049", "AMPUSER=100") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/100-00000049", "AMPUSERCIDNAME=Reseption") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000049", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/100-00000049", "AMPUSERCID=100") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/100-00000049", "CALLERID(all)="Reseption" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-00000049", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-00000049", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/100-00000049", "CALLERID(number)=100") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/100-00000049", "CALLERID(name)=Reseption") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/100-00000049", "Using CallerID "Reseption" ") in new stack -- Executing [980925@from-internal:2] NoOp("SIP/100-00000049", "Calling Out Route: ishodyashie") in new stack -- Executing [980925@from-internal:3] Set("SIP/100-00000049", "MOHCLASS=default") in new stack -- Executing [980925@from-internal:4] Set("SIP/100-00000049", "_NODEST=") in new stack -- Executing [980925@from-internal:5] Macro("SIP/100-00000049", "record-enable,100,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/100-00000049", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/100-00000049", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/100-00000049", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/100-00000049", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("SIP/100-00000049", "1?MacroExit()") in new stack -- Executing [980925@from-internal:6] Macro("SIP/100-00000049", "dialout-trunk,2,83462980925,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000049", "DIAL_TRUNK=2") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000049", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000049", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000049", "DIAL_NUMBER=83462980925") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000049", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000049", "OUTBOUND_GROUP=OUT_2") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000049", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000049", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000049", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000049", "outbound-callerid,2") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000049", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000049", "0?Set(REALCALLERIDNUM=100)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-00000049", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/100-00000049", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000049", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000049", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-00000049", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-00000049", "1?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-00000049", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-00000049", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000049", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000049", "0?sub-flp-2,s,1") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000049", "OUTNUM=83462980925") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000049", "custom=SIP/dom1") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000049", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-00000049", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000049", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-00000049", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000049", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000049", "SIP/dom1/83462980925,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 17850 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 62.148.237.152:5060: INVITE sip:83462980925@62.148.237.152:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK3e45c33e;rport Max-Forwards: 70 From: ;tag=as241e70d7 To: Contact: Call-ID: 7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(11.10.0) Date: Tue, 08 Jul 2014 09:43:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 258 v=0 o=root 1929113078 1929113078 IN IP4 192.168.3.6 s=Asterisk PBX 11.10.0 c=IN IP4 192.168.3.6 t=0 0 m=audio 17850 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/dom1/83462980925 SIP/2.0 100 Trying From: ;tag=as241e70d7 To: Call-ID: 7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK3e45c33e Content-Length: 0 --- (7 headers 0 lines) --- SIP/2.0 403 Forbidden From: ;tag=as241e70d7 To: ;tag=450421407 Call-ID: 7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK3e45c33e contact: supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join Content-Length: 0 --- (9 headers 0 lines) --- Transmitting (NAT) to 62.148.237.152:5060: ACK sip:83462980925@62.148.237.152:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK3e45c33e;rport Max-Forwards: 70 From: ;tag=as241e70d7 To: ;tag=450421407 Contact: Call-ID: 7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(11.10.0) Content-Length: 0 --- Scheduling destruction of SIP dialog '7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060' in 6400 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/100-00000049", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack -- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-00000049", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/100-00000049", "RC=21") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/100-00000049", "21,1") in new stack -- Goto (macro-dialout-trunk,21,1) -- Executing [21@macro-dialout-trunk:1] Goto("SIP/100-00000049", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/100-00000049", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-00000049", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/100-00000049", "CALLERID(number)=100") in new stack -- Executing [980925@from-internal:7] Macro("SIP/100-00000049", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/100-00000049", "") in new stack -- Executing [s@macro-outisbusy:2] GotoIf("SIP/100-00000049", "0?emergency,1") in new stack -- Executing [s@macro-outisbusy:3] GotoIf("SIP/100-00000049", "0?intracompany,1") in new stack -- Executing [s@macro-outisbusy:4] Playback("SIP/100-00000049", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack -- Playing 'all-circuits-busy-now.slin' (language 'ru') > 0x185e0be0 -- Probation passed - setting RTP source address to 192.168.3.200:16392 -- Playing 'pls-try-call-later.slin' (language 'ru') -- Executing [s@macro-outisbusy:5] Congestion("SIP/100-00000049", "20") in new stack == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/100-00000049' in macro 'outisbusy' == Spawn extension (from-internal, 980925, 7) exited non-zero on 'SIP/100-00000049' -- Executing [h@from-internal:1] Macro("SIP/100-00000049", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000049", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000049", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000049", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000049", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000049", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000049", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000049", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000049", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000049", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000049", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000049", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/100-00000049", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/100-00000049", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/100-00000049' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000049' Really destroying SIP dialog '7003a5e361a7c7f57421d09f7a4a7e19@192.168.3.6:5060' Method: INVITE REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 62.148.237.152:5060: REGISTER sip:srgngn.usi.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK786f1d92;rport Max-Forwards: 70 From: ;tag=as3da83ae4 To: Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 216 REGISTER User-Agent: FPBX-2.8.1(11.10.0) Authorization: Digest username="dom", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgxMjMyMzIwMWUyMzJkZWU5OTNkMDkzMjNlMzQ5ODU1NDdjYTgzNDVk", response="1e32c237ed65fa4af0b406f4cecf4ffd", qop=auth, cnonce="2cf0e2f5", nc=00000004 Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 100 Trying From: ;tag=as3da83ae4 To: Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 216 REGISTER Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK786f1d92 Content-Length: 0 --- (7 headers 0 lines) --- SIP/2.0 407 Proxy Authentication Required From: ;tag=as3da83ae4 To: ;tag=1925028415 Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 216 REGISTER Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK786f1d92 supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join proxy-authenticate: Digest realm="Realm",nonce="MTQwNDgxMjYzMjU4MzA0NjU4NTIyMmI2OGYyOTQxMzI4MGVhOThkMTc5NGYy",stale=true,algorithm=MD5,qop="auth,auth-int" Content-Length: 0 --- (9 headers 0 lines) --- Responding to challenge, registration to domain/host name srgngn.usi.ru REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 62.148.237.152:5060: REGISTER sip:srgngn.usi.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK5f65dd3b;rport Max-Forwards: 70 From: ;tag=as3da83ae4 To: Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 217 REGISTER User-Agent: FPBX-2.8.1(11.10.0) Proxy-Authorization: Digest username="dom", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgxMjYzMjU4MzA0NjU4NTIyMmI2OGYyOTQxMzI4MGVhOThkMTc5NGYy", response="12a959631c362aca5c18590595c2a0c7", qop=auth, cnonce="3f0f1a80", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 100 Trying From: ;tag=as3da83ae4 To: Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 217 REGISTER Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK5f65dd3b Content-Length: 0 --- (7 headers 0 lines) --- SIP/2.0 200 Registration Successful From: "dom dom";tag=as3da83ae4 To: ;tag=100416184 Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1 CSeq: 217 REGISTER Via: SIP/2.0/UDP 192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK5f65dd3b contact: ;expires=118,;expires=33 supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join Content-Length: 0 --- (9 headers 0 lines) --- Really destroying SIP dialog '152108f82a083ac23fec854978e19fd6@127.0.0.1' Method: REGISTER elastix*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups [root@elastix ~]# |
Последний раз редактировалось: Sobsoft (Вт Июл 08, 2014 11:52)
спойлер
Во-вторых, вам нужно обратиться к своему оператору:
| Код: |
| SIP/2.0 403 Forbidden From: ;tag=as241e70d7 |
Так как у вас либо неправильные настройки "транка", либо вы посылаете номер не в том формате.
у провайдера спросить "правила набора номера для asterisk"?
Лично я бы сформулировал вопрос так: почему не проходят исходящие звонки.
Не знаю кто ваш оператор, но обычно еще "вися" на трубке ТП просит сделать пару исходящих вызовов, для того что бы диагностировать проблему со своей стороны,
и выдает заключение.
Это я к тому, что проблема может быть и не в правилах набора. А, например, в неправильной посылке вызывающего номера. В любом случае -- правильно диагностировать проблему сможет только ТП вашего оператора.
| xelas @ Вт Июл 08, 2014 17:37 писал(а): |
| Ну, если вы уверены в остальных настройках, то можете спросить и так. Лично я бы сформулировал вопрос так: почему не проходят исходящие звонки. Не знаю кто ваш оператор, но обычно еще "вися" на трубке ТП просит сделать пару исходящих вызовов, для того что бы диагностировать проблему со своей стороны, и выдает заключение. Это я к тому, что проблема может быть и не в правилах набора. А, например, в неправильной посылке вызывающего номера. В любом случае -- правильно диагностировать проблему сможет только ТП вашего оператора. |
оператор Ростелеком и это очень печалит. я не могу даже до персонального менеджера дозвониться - тактично не берет трубку и не перезванивает.
понял попытаю провайдера, если найду там решение отпишусь.