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спросил 2014-07-07 09:09:01 +0400

Sobsoft Gravatar Sobsoft

Все линии заняты при попытке позвонить

Доброго времени суток. Сразу скажу, что в asterisk'е я полный нуб, так как только начал им заниматься.

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:

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

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.

Все линии заняты при попытке позвонить

Доброго времени суток. Сразу скажу, что в asterisk'е я полный нуб, так как только начал им заниматься.

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:

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

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.

Результат команды: asterisk -rvvvv

Connected to Asterisk 11.10.0 currently running on elastix (pid = 3205)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [89825939633@from-internal:1] Macro("SIP/101-00000009", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-00000009", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000009", "AMPUSERCIDNAME=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000009", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/101-00000009", "CALLERID(all)="KondrenkovEV" <101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/101-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/101-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/101-00000009", "CALLERID(name)=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/101-00000009", "Using CallerID "KondrenkovEV" <101>") in new stack
    -- Executing [89825939633@from-internal:2] NoOp("SIP/101-00000009", "Calling Out Route: outbound") in new stack
    -- Executing [89825939633@from-internal:3] Set("SIP/101-00000009", "MOHCLASS=default") in new stack
    -- Executing [89825939633@from-internal:4] Set("SIP/101-00000009", "_NODEST=") in new stack
    -- Executing [89825939633@from-internal:5] Macro("SIP/101-00000009", "record-enable,101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-00000009", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/101-00000009", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/101-00000009", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/101-00000009", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/101-00000009", "1?MacroExit()") in new stack
    -- Executing [89825939633@from-internal:6] Macro("SIP/101-00000009", "dialout-trunk,2,89825939633,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000009", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000009", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000009", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000009", "DIAL_NUMBER=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000009", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000009", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000009", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000009", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000009", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000009", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000009", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000009", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000009", "TRUNKOUTCID=<3462206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000009", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-00000009", "1?Set(CALLERID(all)=<3462206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000009", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000009", "OUTNUM=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000009", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-00000009", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-00000009", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000009", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-00000009", "SIP/dom1/89825939633,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/dom1/89825939633
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/101-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/101-00000009", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/101-00000009", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000009", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/101-00000009", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/101-00000009", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/101-00000009", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [89825939633@from-internal:7] Macro("SIP/101-00000009", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000009", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000009", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000009", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/101-00000009> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x2aafe00cfb10 -- Probation passed - setting RTP source address to 192.168.3.201:16386
    -- <SIP/101-00000009> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/101-00000009", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-00000009' in macro 'outisbusy'
  == Spawn extension (from-internal, 89825939633, 7) exited non-zero on 'SIP/101-00000009'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000009", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000009", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/101-00000009", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000009", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/101-00000009", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/101-00000009", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/101-00000009", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/101-00000009", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/101-00000009", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/101-00000009", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/101-00000009", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/101-00000009", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/101-00000009", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/101-00000009>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/101-00000009", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/101-00000009' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000009'

Все линии заняты при попытке позвонить

Доброго времени суток. Сразу скажу, что в asterisk'е я полный нуб, так как только начал им заниматься.

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:

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

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.

Результат команды: asterisk -rvvvv

Connected to Asterisk 11.10.0 currently running on elastix (pid = 3205)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [89825939633@from-internal:1] Macro("SIP/101-00000009", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-00000009", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000009", "AMPUSERCIDNAME=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000009", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/101-00000009", "CALLERID(all)="KondrenkovEV" <101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/101-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/101-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/101-00000009", "CALLERID(name)=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/101-00000009", "Using CallerID "KondrenkovEV" <101>") in new stack
    -- Executing [89825939633@from-internal:2] NoOp("SIP/101-00000009", "Calling Out Route: outbound") in new stack
    -- Executing [89825939633@from-internal:3] Set("SIP/101-00000009", "MOHCLASS=default") in new stack
    -- Executing [89825939633@from-internal:4] Set("SIP/101-00000009", "_NODEST=") in new stack
    -- Executing [89825939633@from-internal:5] Macro("SIP/101-00000009", "record-enable,101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-00000009", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/101-00000009", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/101-00000009", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/101-00000009", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/101-00000009", "1?MacroExit()") in new stack
    -- Executing [89825939633@from-internal:6] Macro("SIP/101-00000009", "dialout-trunk,2,89825939633,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000009", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000009", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000009", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000009", "DIAL_NUMBER=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000009", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000009", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000009", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000009", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000009", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000009", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000009", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000009", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000009", "TRUNKOUTCID=<3462206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000009", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-00000009", "1?Set(CALLERID(all)=<3462206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000009", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000009", "OUTNUM=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000009", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-00000009", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-00000009", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000009", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-00000009", "SIP/dom1/89825939633,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/dom1/89825939633
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/101-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/101-00000009", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/101-00000009", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000009", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/101-00000009", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/101-00000009", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/101-00000009", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [89825939633@from-internal:7] Macro("SIP/101-00000009", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000009", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000009", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000009", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/101-00000009> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x2aafe00cfb10 -- Probation passed - setting RTP source address to 192.168.3.201:16386
    -- <SIP/101-00000009> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/101-00000009", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-00000009' in macro 'outisbusy'
  == Spawn extension (from-internal, 89825939633, 7) exited non-zero on 'SIP/101-00000009'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000009", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000009", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/101-00000009", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000009", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/101-00000009", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/101-00000009", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/101-00000009", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/101-00000009", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/101-00000009", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/101-00000009", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/101-00000009", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/101-00000009", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/101-00000009", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/101-00000009>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/101-00000009", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/101-00000009' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000009'

вот дебаг транка/ов:

   elastix*CLI> sip set debug peer ЛОГИН1
SIP Debugging Enabled for IP: IPАДРЕСПРОКСИРОСТЕЛЕКОМ
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
OPTIONS sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK61c3a148;rport
Max-Forwards: 70
From: "Unknown" <sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as31ec0654
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
Contact: <sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 404 Not Found
From: "Unknown"<sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as31ec0654
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>;tag=2115544740
Call-ID: 704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK61c3a148
contact: <sip:belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: OPTIONS
Audio is at 16760
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
INVITE sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK7f971e7a;rport
Max-Forwards: 70
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:47 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 1360211675 1360211675 IN IP4 ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК
s=Asterisk PBX 11.10.0
c=IN IP4 ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК
t=0 0
m=audio 16760 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

---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 100 Trying
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK7f971e7a
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 403 Forbidden
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>;tag=1879402295
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK7f971e7a
contact: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
ACK sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK7f971e7a;rport
Max-Forwards: 70
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>;tag=1879402295
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(11.10.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: INVITE
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
OPTIONS sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK6811a13a;rport
Max-Forwards: 70
From: "Unknown" <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as5b148e0e
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 404 Not Found
From: "Unknown"<sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as5b148e0e
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>;tag=1560659494
Call-ID: 6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK6811a13a
contact: <sip:belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: OPTIONS
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
REGISTER sip:srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK257e1393;rport
Max-Forwards: 70
From: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
CSeq: 116 REGISTER
User-Agent: FPBX-2.8.1(11.10.0)
Authorization: Digest username="ЛОГИН2ТРАНКА", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgwMTMyMTMyOGYzYTQzMWJlNTczMDBmNWUyY2NhYjE2ZmUwYmQzMWNm", response="e86d8519d715db30e7216b6f266c8f70", qop=auth, cnonce="4d0c05ab", nc=00000002
Expires: 120
Contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 100 Trying
From: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK257e1393
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 200 Registration Successful
From: "ЛОГИН2ТРАНКА ЛОГИН2ТРАНКА"<sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=591634824
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK257e1393
contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>;expires=111,<sip:206855@192.168.3.101:5060;transport=UDP>;expires=58
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2277da5418df63f96f64cd9d05f0c15e@127.0.0.1' Method: REGISTER
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
REGISTER sip:srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK275f2e11;rport
Max-Forwards: 70
From: <sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
User-Agent: FPBX-2.8.1(11.10.0)
Authorization: Digest username="ЛОГИН", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgwMTMyMTMyOGYzYTQzMWJlNTczMDBmNWUyY2NhYjE2ZmUwYmQzMWNm", response="ebf9dad3fbd994151626a51d647b22a0", qop=auth, cnonce="121574c5", nc=00000002
Expires: 120
Contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 100 Trying
From: <sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK275f2e11
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 200 Registration Successful
From: "ЛОГИН ЛОГИН"<sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>;tag=1318887433
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK275f2e11
contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>;expires=119,<sip:206906@192.168.3.101:5060;transport=UDP>;expires=43
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

для удобства и безопасности переименовал адреса и логины на текст в верхнем регистре.

Все линии заняты при попытке позвонить

Доброго времени суток. Сразу скажу, что в asterisk'е я полный нуб, так как только начал им заниматься.

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:

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

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.

Результат команды: asterisk -rvvvv

Connected to Asterisk 11.10.0 currently running on elastix (pid = 3205)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [89825939633@from-internal:1] Macro("SIP/101-00000009", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-00000009", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000009", "AMPUSERCIDNAME=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000009", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/101-00000009", "CALLERID(all)="KondrenkovEV" <101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/101-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/101-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/101-00000009", "CALLERID(name)=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/101-00000009", "Using CallerID "KondrenkovEV" <101>") in new stack
    -- Executing [89825939633@from-internal:2] NoOp("SIP/101-00000009", "Calling Out Route: outbound") in new stack
    -- Executing [89825939633@from-internal:3] Set("SIP/101-00000009", "MOHCLASS=default") in new stack
    -- Executing [89825939633@from-internal:4] Set("SIP/101-00000009", "_NODEST=") in new stack
    -- Executing [89825939633@from-internal:5] Macro("SIP/101-00000009", "record-enable,101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-00000009", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/101-00000009", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/101-00000009", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/101-00000009", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/101-00000009", "1?MacroExit()") in new stack
    -- Executing [89825939633@from-internal:6] Macro("SIP/101-00000009", "dialout-trunk,2,89825939633,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000009", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000009", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000009", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000009", "DIAL_NUMBER=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000009", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000009", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000009", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000009", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000009", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000009", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000009", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000009", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000009", "TRUNKOUTCID=<3462206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000009", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-00000009", "1?Set(CALLERID(all)=<3462206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000009", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000009", "OUTNUM=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000009", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-00000009", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-00000009", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000009", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-00000009", "SIP/dom1/89825939633,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/dom1/89825939633
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/101-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/101-00000009", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/101-00000009", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000009", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/101-00000009", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/101-00000009", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/101-00000009", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [89825939633@from-internal:7] Macro("SIP/101-00000009", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000009", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000009", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000009", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/101-00000009> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x2aafe00cfb10 -- Probation passed - setting RTP source address to 192.168.3.201:16386
    -- <SIP/101-00000009> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/101-00000009", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-00000009' in macro 'outisbusy'
  == Spawn extension (from-internal, 89825939633, 7) exited non-zero on 'SIP/101-00000009'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000009", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000009", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/101-00000009", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000009", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/101-00000009", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/101-00000009", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/101-00000009", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/101-00000009", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/101-00000009", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/101-00000009", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/101-00000009", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/101-00000009", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/101-00000009", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/101-00000009>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/101-00000009", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/101-00000009' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000009'

вот дебаг транка/ов:

 elastix*CLI> sip set debug peer ЛОГИН1
dom1
SIP Debugging Enabled for IP: IPАДРЕСПРОКСИРОСТЕЛЕКОМ
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
OPTIONS sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK61c3a148;rport
Max-Forwards: 70
From: "Unknown" <sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as31ec0654
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
Contact: <sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- 62.148.237.152
  == Using SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 404 Not Found
From: "Unknown"<sip:НОМЕРТЕЛФОНАУВТОРОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as31ec0654
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>;tag=2115544740
Call-ID: 704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK61c3a148
contact: <sip:belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying RTP TOS bits 184
  == Using SIP dialog '704bd30041285c63537f6ad32975a89f@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: OPTIONS
RTP CoS mark 5
    -- Executing [980925@from-internal:1] Macro("SIP/100-00000047", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/100-00000047", "AMPUSER=100") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-00000047", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-00000047", "1?Set(REALCALLERIDNUM=100)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/100-00000047", "AMPUSER=100") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/100-00000047", "AMPUSERCIDNAME=Reseption") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000047", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/100-00000047", "AMPUSERCID=100") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/100-00000047", "CALLERID(all)="Reseption" <100>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-00000047", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-00000047", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/100-00000047", "CALLERID(number)=100") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/100-00000047", "CALLERID(name)=Reseption") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/100-00000047", "Using CallerID "Reseption" <100>") in new stack
    -- Executing [980925@from-internal:2] NoOp("SIP/100-00000047", "Calling Out Route: ishodyashie") in new stack
    -- Executing [980925@from-internal:3] Set("SIP/100-00000047", "MOHCLASS=default") in new stack
    -- Executing [980925@from-internal:4] Set("SIP/100-00000047", "_NODEST=") in new stack
    -- Executing [980925@from-internal:5] Macro("SIP/100-00000047", "record-enable,100,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/100-00000047", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/100-00000047", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/100-00000047", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/100-00000047", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/100-00000047", "1?MacroExit()") in new stack
    -- Executing [980925@from-internal:6] Macro("SIP/100-00000047", "dialout-trunk,2,83462980925,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000047", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000047", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000047", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000047", "DIAL_NUMBER=83462980925") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000047", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000047", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000047", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000047", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000047", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000047", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000047", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000047", "0?Set(REALCALLERIDNUM=100)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-00000047", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/100-00000047", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000047", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000047", "TRUNKOUTCID=<206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-00000047", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-00000047", "1?Set(CALLERID(all)=<206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-00000047", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-00000047", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000047", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000047", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000047", "OUTNUM=83462980925") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000047", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000047", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-00000047", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000047", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-00000047", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000047", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000047", "SIP/dom1/83462980925,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 16760
15382
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
62.148.237.152:5060:
INVITE sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 sip:83462980925@62.148.237.152:5060 SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK7f971e7a;rport
192.168.3.6:5060;branch=z9hG4bK53bdda43;rport
Max-Forwards: 70
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
<sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>
<sip:83462980925@62.148.237.152:5060>
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
<sip:dom@192.168.3.6:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
37bab8f3522d427f296bde565d86c709@192.168.3.6:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:47 09:38:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 258
256

v=0
o=root 1360211675 1360211675 474005419 474005419 IN IP4 ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК
192.168.3.6
s=Asterisk PBX 11.10.0
c=IN IP4 ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК
192.168.3.6
t=0 0
m=audio 16760 15382 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 read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
<sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>
<sip:83462980925@62.148.237.152:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
37bab8f3522d427f296bde565d86c709@192.168.3.6:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK7f971e7a
192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK53bdda43
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 UDP:62.148.237.152:5060 --->
SIP/2.0 403 Forbidden
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
<sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>;tag=1879402295
<sip:83462980925@62.148.237.152:5060>;tag=1542634927
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
37bab8f3522d427f296bde565d86c709@192.168.3.6:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK7f971e7a
192.168.3.6:5060;received=188.19.12.85;rport=2048;branch=z9hG4bK53bdda43
contact: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
<sip:83462980925@belngn.usi.ru:5060;maddr=62.148.237.152>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
62.148.237.152:5060:
ACK sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 sip:83462980925@62.148.237.152:5060 SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK7f971e7a;rport
192.168.3.6:5060;branch=z9hG4bK53bdda43;rport
Max-Forwards: 70
From: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as682a2d82
<sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:КУДАЗВОНИМНОМЕРТЕЛЕФОНА@IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060>;tag=1879402295
<sip:83462980925@62.148.237.152:5060>;tag=1542634927
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
<sip:dom@192.168.3.6:5060>
Call-ID: 7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
37bab8f3522d427f296bde565d86c709@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 '7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' '37bab8f3522d427f296bde565d86c709@192.168.3.6:5060' in 6400 ms (Method: INVITE)
Really destroying SIP dialog '7bf535b710e38be57761b36f249e5cd0@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: INVITE
Reliably Transmitting (NAT)   == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/100-00000047", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-00000047", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/100-00000047", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/100-00000047", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/100-00000047", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/100-00000047", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-00000047", "TRUNK Dial failed due to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
OPTIONS sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK6811a13a;rport
Max-Forwards: 70
From: "Unknown" <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as5b148e0e
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
Contact: <sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Call-ID: 6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 06:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/100-00000047", "CALLERID(number)=100") in new stack
    -- Executing [980925@from-internal:7] Macro("SIP/100-00000047", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/100-00000047", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/100-00000047", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/100-00000047", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/100-00000047", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/100-00000047> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x185e0be0 -- Probation passed - setting RTP source address to 192.168.3.200:16390
    -- <SIP/100-00000047> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/100-00000047", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/100-00000047' in macro 'outisbusy'
  == Spawn extension (from-internal, 980925, 7) exited non-zero on 'SIP/100-00000047'
    -- Executing [h@from-internal:1] Macro("SIP/100-00000047", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000047", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000047", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000047", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000047", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000047", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000047", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000047", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000047", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000047", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000047", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000047", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/100-00000047", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/100-00000047>AGI Script hangup.agi completed, returning 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 404 Not Found
From: "Unknown"<sip:НОМЕРТЕЛФОНАУПЕРВОГОТРАНКА@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК>;tag=as5b148e0e
To: <sip:IPАДРЕСПРОКСИРОСТЕЛЕКОМ>;tag=1560659494
Call-ID: 6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK6811a13a
contact: <sip:belngn.usi.ru:5060;maddr=IPАДРЕСПРОКСИРОСТЕЛЕКОМ>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6f0a943d7364aecf5aae25fe506bde3b@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060' Method: OPTIONS
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/100-00000047", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/100-00000047' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000047'
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
62.148.237.152:5060:
REGISTER sip:srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK257e1393;rport
192.168.3.6:5060;branch=z9hG4bK6f5130c8;rport
Max-Forwards: 70
From: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
<sip:dom@srgngn.usi.ru>;tag=as3da83ae4
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>
<sip:dom@srgngn.usi.ru>
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
152108f82a083ac23fec854978e19fd6@127.0.0.1
CSeq: 116 211 REGISTER
User-Agent: FPBX-2.8.1(11.10.0)
Authorization: Digest username="ЛОГИН2ТРАНКА", username="dom", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgwMTMyMTMyOGYzYTQzMWJlNTczMDBmNWUyY2NhYjE2ZmUwYmQzMWNm", response="e86d8519d715db30e7216b6f266c8f70", nonce="MTQwNDgxMjAxMjY0MjU3YzE0Njc5NWEwZjgxMGZkMWE1NGM5YmRiOTU3NTA3", response="ad79da784f7b1f6794b83c49b672f7db", qop=auth, cnonce="4d0c05ab", nc=00000002
cnonce="39b8af96", nc=00000004
Expires: 120
Contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 100 Trying
From: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK257e1393
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 200 Registration Successful
From: "ЛОГИН2ТРАНКА ЛОГИН2ТРАНКА"<sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=as5d40a434
To: <sip:ЛОГИН2ТРАНКА@srgngn.usi.ru>;tag=591634824
Call-ID: 2277da5418df63f96f64cd9d05f0c15e@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK257e1393
contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>;expires=111,<sip:206855@192.168.3.101:5060;transport=UDP>;expires=58
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '2277da5418df63f96f64cd9d05f0c15e@127.0.0.1' Method: REGISTER
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060:
REGISTER sip:srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;branch=z9hG4bK275f2e11;rport
Max-Forwards: 70
From: <sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
User-Agent: FPBX-2.8.1(11.10.0)
Authorization: Digest username="ЛОГИН", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgwMTMyMTMyOGYzYTQzMWJlNTczMDBmNWUyY2NhYjE2ZmUwYmQzMWNm", response="ebf9dad3fbd994151626a51d647b22a0", qop=auth, cnonce="121574c5", nc=00000002
Expires: 120
Contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>
Content-Length: 0


---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 100 Trying
From: <sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK275f2e11
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:IPАДРЕСПРОКСИРОСТЕЛЕКОМ:5060 --->
SIP/2.0 200 Registration Successful
From: "ЛОГИН ЛОГИН"<sip:ЛОГИН@srgngn.usi.ru>;tag=as0c3750e0
To: <sip:ЛОГИН@srgngn.usi.ru>;tag=1318887433
Call-ID: 3739a80c3700cd447404a6b47325ecd7@127.0.0.1
CSeq: 116 REGISTER
Via: SIP/2.0/UDP ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060;received=СТАТИЧЕСКИЙАЙПИВНЕШНИЙ;rport=2048;branch=z9hG4bK275f2e11
contact: <sip:s@ЛОКАЛЬНЫЙАЙПИСЕРВЕРААСТЕРИСК:5060>;expires=119,<sip:206906@192.168.3.101:5060;transport=UDP>;expires=43
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
<sip:s@192.168.3.6:5060>
Content-Length: 0

для удобства и безопасности переименовал адреса и логины на текст в верхнем регистре.

Все линии заняты при попытке позвонить

Доброго времени суток. Сразу скажу, что в asterisk'е я полный нуб, так как только начал им заниматься.

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:

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

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.

Результат команды: asterisk -rvvvv

Connected to Asterisk 11.10.0 currently running on elastix (pid = 3205)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [89825939633@from-internal:1] Macro("SIP/101-00000009", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/101-00000009", "1?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/101-00000009", "AMPUSER=101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/101-00000009", "AMPUSERCIDNAME=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/101-00000009", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/101-00000009", "AMPUSERCID=101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/101-00000009", "CALLERID(all)="KondrenkovEV" <101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/101-00000009", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/101-00000009", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/101-00000009", "CALLERID(name)=KondrenkovEV") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/101-00000009", "Using CallerID "KondrenkovEV" <101>") in new stack
    -- Executing [89825939633@from-internal:2] NoOp("SIP/101-00000009", "Calling Out Route: outbound") in new stack
    -- Executing [89825939633@from-internal:3] Set("SIP/101-00000009", "MOHCLASS=default") in new stack
    -- Executing [89825939633@from-internal:4] Set("SIP/101-00000009", "_NODEST=") in new stack
    -- Executing [89825939633@from-internal:5] Macro("SIP/101-00000009", "record-enable,101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/101-00000009", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/101-00000009", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/101-00000009", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/101-00000009", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/101-00000009", "1?MacroExit()") in new stack
    -- Executing [89825939633@from-internal:6] Macro("SIP/101-00000009", "dialout-trunk,2,89825939633,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/101-00000009", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/101-00000009", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/101-00000009", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/101-00000009", "DIAL_NUMBER=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/101-00000009", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/101-00000009", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/101-00000009", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/101-00000009", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/101-00000009", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/101-00000009", "0?Set(REALCALLERIDNUM=101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/101-00000009", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/101-00000009", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/101-00000009", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/101-00000009", "TRUNKOUTCID=<3462206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/101-00000009", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/101-00000009", "1?Set(CALLERID(all)=<3462206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/101-00000009", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/101-00000009", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/101-00000009", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/101-00000009", "OUTNUM=89825939633") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/101-00000009", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/101-00000009", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/101-00000009", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/101-00000009", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/101-00000009", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/101-00000009", "SIP/dom1/89825939633,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/dom1/89825939633
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/101-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/101-00000009", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/101-00000009", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/101-00000009", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/101-00000009", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/101-00000009", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/101-00000009", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/101-00000009", "CALLERID(number)=101") in new stack
    -- Executing [89825939633@from-internal:7] Macro("SIP/101-00000009", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/101-00000009", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/101-00000009", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/101-00000009", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/101-00000009", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/101-00000009> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x2aafe00cfb10 -- Probation passed - setting RTP source address to 192.168.3.201:16386
    -- <SIP/101-00000009> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/101-00000009", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/101-00000009' in macro 'outisbusy'
  == Spawn extension (from-internal, 89825939633, 7) exited non-zero on 'SIP/101-00000009'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000009", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000009", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/101-00000009", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000009", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/101-00000009", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/101-00000009", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/101-00000009", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/101-00000009", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/101-00000009", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/101-00000009", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/101-00000009", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/101-00000009", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/101-00000009", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/101-00000009>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/101-00000009", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/101-00000009' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000009'

вот дебаг транка/ов:

elastix*CLI> sip set debug peer dom1
SIP Debugging Enabled for IP: 62.148.237.152
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [980925@from-internal:1] Macro("SIP/100-00000047", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/100-00000047", "AMPUSER=100") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-00000047", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-00000047", "1?Set(REALCALLERIDNUM=100)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/100-00000047", "AMPUSER=100") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/100-00000047", "AMPUSERCIDNAME=Reseption") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-00000047", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/100-00000047", "AMPUSERCID=100") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/100-00000047", "CALLERID(all)="Reseption" <100>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/100-00000047", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/100-00000047", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/100-00000047", "CALLERID(number)=100") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/100-00000047", "CALLERID(name)=Reseption") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/100-00000047", "Using CallerID "Reseption" <100>") in new stack
    -- Executing [980925@from-internal:2] NoOp("SIP/100-00000047", "Calling Out Route: ishodyashie") in new stack
    -- Executing [980925@from-internal:3] Set("SIP/100-00000047", "MOHCLASS=default") in new stack
    -- Executing [980925@from-internal:4] Set("SIP/100-00000047", "_NODEST=") in new stack
    -- Executing [980925@from-internal:5] Macro("SIP/100-00000047", "record-enable,100,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/100-00000047", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/100-00000047", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/100-00000047", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/100-00000047", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/100-00000047", "1?MacroExit()") in new stack
    -- Executing [980925@from-internal:6] Macro("SIP/100-00000047", "dialout-trunk,2,83462980925,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/100-00000047", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-00000047", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-00000047", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/100-00000047", "DIAL_NUMBER=83462980925") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/100-00000047", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/100-00000047", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-00000047", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-00000047", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/100-00000047", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-00000047", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-00000047", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-00000047", "0?Set(REALCALLERIDNUM=100)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-00000047", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/100-00000047", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/100-00000047", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/100-00000047", "TRUNKOUTCID=<206906>") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-00000047", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-00000047", "1?Set(CALLERID(all)=<206906>)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/100-00000047", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/100-00000047", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/100-00000047", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/100-00000047", "0?sub-flp-2,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/100-00000047", "OUTNUM=83462980925") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/100-00000047", "custom=SIP/dom1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-00000047", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-00000047", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-00000047", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-00000047", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-00000047", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-00000047", "SIP/dom1/83462980925,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 15382
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=z9hG4bK53bdda43;rport
Max-Forwards: 70
From: <sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:83462980925@62.148.237.152:5060>
Contact: <sip:dom@192.168.3.6:5060>
Call-ID: 37bab8f3522d427f296bde565d86c709@192.168.3.6:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(11.10.0)
Date: Tue, 08 Jul 2014 09:38:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 474005419 474005419 IN IP4 192.168.3.6
s=Asterisk PBX 11.10.0
c=IN IP4 192.168.3.6
t=0 0
m=audio 15382 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 read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: <sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:83462980925@62.148.237.152:5060>
Call-ID: 37bab8f3522d427f296bde565d86c709@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=z9hG4bK53bdda43
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 403 Forbidden
From: <sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:83462980925@62.148.237.152:5060>;tag=1542634927
Call-ID: 37bab8f3522d427f296bde565d86c709@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=z9hG4bK53bdda43
contact: <sip:83462980925@belngn.usi.ru:5060;maddr=62.148.237.152>
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=z9hG4bK53bdda43;rport
Max-Forwards: 70
From: <sip:dom@192.168.3.6>;tag=as5fff8c12
To: <sip:83462980925@62.148.237.152:5060>;tag=1542634927
Contact: <sip:dom@192.168.3.6:5060>
Call-ID: 37bab8f3522d427f296bde565d86c709@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 '37bab8f3522d427f296bde565d86c709@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/100-00000047", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-00000047", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/100-00000047", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/100-00000047", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/100-00000047", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/100-00000047", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/100-00000047", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/100-00000047", "CALLERID(number)=100") in new stack
    -- Executing [980925@from-internal:7] Macro("SIP/100-00000047", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/100-00000047", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/100-00000047", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/100-00000047", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/100-00000047", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/100-00000047> Playing 'all-circuits-busy-now.slin' (language 'ru')
       > 0x185e0be0 -- Probation passed - setting RTP source address to 192.168.3.200:16390
    -- <SIP/100-00000047> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/100-00000047", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/100-00000047' in macro 'outisbusy'
  == Spawn extension (from-internal, 980925, 7) exited non-zero on 'SIP/100-00000047'
    -- Executing [h@from-internal:1] Macro("SIP/100-00000047", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000047", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000047", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000047", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000047", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000047", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000047", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000047", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000047", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000047", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000047", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000047", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/100-00000047", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/100-00000047>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/100-00000047", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/100-00000047' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000047'
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=z9hG4bK6f5130c8;rport
Max-Forwards: 70
From: <sip:dom@srgngn.usi.ru>;tag=as3da83ae4
To: <sip:dom@srgngn.usi.ru>
Call-ID: 152108f82a083ac23fec854978e19fd6@127.0.0.1
CSeq: 211 REGISTER
User-Agent: FPBX-2.8.1(11.10.0)
Authorization: Digest username="dom", realm="Realm", algorithm=MD5, uri="sip:srgngn.usi.ru", nonce="MTQwNDgxMjAxMjY0MjU3YzE0Njc5NWEwZjgxMGZkMWE1NGM5YmRiOTU3NTA3", response="ad79da784f7b1f6794b83c49b672f7db", qop=auth, cnonce="39b8af96", nc=00000004
Expires: 120
Contact: <sip:s@192.168.3.6:5060>
Content-Length: 0

описание транка

host=62.148.237.152
port=5060
username=dom
secret=пароль
fromuser=dom
type=friend
context=from_external
dtmfmode=rfc2833
canreinvite=no
nat=yes
insecure=port,invite
qualify=200
disallow=all
allow=all

fromdomain не выставлял - не думал что оно обязательно

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.