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Переадресация dongle на другой астериск

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

oddity
Сообщения: 13
Зарегистрирован: 05 сен 2016, 14:40

Переадресация dongle на другой астериск

Сообщение oddity »

Добрый день!
В нашей фирме есть два одинаковых Freepbx. Один основной, на нём около ста пользователей, на втором с недавнего времени настроен и работает dongle. Исходящие и входящие звонки на нём работают, но появилась необходимость переадресовать звонки с 3g модема на основной фрипбх, что на IP-телефоны не заводить две SIP линии с разных астеров. Буду благодарен если подскажите как правильно это реализовать. Спасибо!
ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Переадресация dongle на другой астериск

Сообщение ded »

1. Соединить оба по IAX2, причём в первом Астериске контекст для IAX2 транка должен быть from-trunk
2. Направить всё, что прибегает с dongle в IAX2 транк на первый Астериск, на котором разрулить входящие через модуль Входящая маршрутизация.
3. Исходящие же по шаблону 89ХХХХХХХХХ отправлять во второй Астериск через IAX2 транк.
oddity
Сообщения: 13
Зарегистрирован: 05 сен 2016, 14:40

Re: Переадресация dongle на другой астериск

Сообщение oddity »

Спасибо большое! Получилось со входящей маршрутизацией. А вот с исходящей говорит "На данный момент все линии заняты". Хотя если звонить по внутренним номерам между астерами, то звонки проходят без проблем в оба направления. Не пойму где я упустил момент.
april22
Сообщения: 2187
Зарегистрирован: 09 июл 2012, 09:47

Re: Переадресация dongle на другой астериск

Сообщение april22 »

в логах ...
Своими вопросами , вы загоняете меня в ГУГЛЬ.
ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Переадресация dongle на другой астериск

Сообщение ded »

,,, все ответы.
whoim
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Re: Переадресация dongle на другой астериск

Сообщение whoim »

я по sip соединял, все работает )
облачные и локальные сервера asterisk/freepbx/a2billing/crm с полной техподдержкой. skype: whoim2, sipuri: whoim@asterisk.ru
oddity
Сообщения: 13
Зарегистрирован: 05 сен 2016, 14:40

Re: Переадресация dongle на другой астериск

Сообщение oddity »

Ребят, сделал всё, как сказал ded, входящая полетела, а исходящие не звонит через донгл. На внутренний номер второго астера звонит, а на МТС никак. Не пойму в чём проблема. Вроде транк между астерами настроил же. правила исходящей тоже. В логах пишет такое:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
[2016-09-12 15:01:07] DEBUG[16109] manager.c: Examining event:
Event: Newexten
Privilege: dialplan,all
Channel: SIP/20202-00001abc
Context: macro-hangupcall
Extension: s
Priority: 3
Application: ExecIf
AppData: 0?Set(CDR(recordingfile)=)
Uniqueid: 1473681665.9636


[2016-09-12 15:01:07] DEBUG[25105][C-00001191] app_macro.c: Executed application: ExecIf
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Evaluating 'ONETOUCH_RECFILE' (from 'ONETOUCH_RECFILE}"!="" & "${CDR(recordingfile)}"=""' len 16)
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Result of 'ONETOUCH_RECFILE' is NULL
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Evaluating 'CDR(recordingfile)' (from 'CDR(recordingfile)}"=""' len 18)
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Function CDR(recordingfile) result is '(null)'
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Expression result is '0'
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Evaluating 'ONETOUCH_RECFILE' (from 'ONETOUCH_RECFILE})' len 16)
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Result of 'ONETOUCH_RECFILE' is NULL
[2016-09-12 15:01:07] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20202-00001abc
Variable: MACRO_DEPTH
Value: 1
Uniqueid: 1473681665.9636


[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Launching 'Hangup'
[2016-09-12 15:01:07] VERBOSE[25105][C-00001191] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/20202-00001abc", "") in new stack
[2016-09-12 15:01:07] DEBUG[16109] manager.c: Examining event:
Event: Newexten
Privilege: dialplan,all
Channel: SIP/20202-00001abc
Context: macro-hangupcall
Extension: s
Priority: 4
Application: Hangup
AppData:
Uniqueid: 1473681665.9636


[2016-09-12 15:01:07] DEBUG[25105][C-00001191] channel.c: Soft-Hanging up channel 'SIP/20202-00001abc'
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] app_macro.c: Spawn extension (macro-hangupcall,s,4) exited non-zero on 'SIP/20202-00001abc' in macro 'hangupcall'
[2016-09-12 15:01:07] VERBOSE[25105][C-00001191] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/20202-00001abc' in macro 'hangupcall'
[2016-09-12 15:01:07] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20202-00001abc
Variable: MACRO_DEPTH
Value: 0
Uniqueid: 1473681665.9636


[2016-09-12 15:01:07] DEBUG[25105][C-00001191] pbx.c: Spawn extension (user-post-LD,h,1) exited non-zero on 'SIP/20202-00001abc'
[2016-09-12 15:01:07] VERBOSE[25105][C-00001191] pbx.c: == Spawn extension (user-post-LD, h, 1) exited non-zero on 'SIP/20202-00001abc'
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] channel.c: Hanging up channel 'SIP/20202-00001abc'
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] chan_sip.c: Hangup call SIP/20202-00001abc, SIP callid 422293984@10.204.10.178
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f528817b208'
[2016-09-12 15:01:07] DEBUG[16109] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/20202-00001abc
Uniqueid: 1473681665.9636
CallerIDNum: 20202
CallerIDName: <unknown>
ConnectedLineNum: +380502180052
ConnectedLineName: CID:0623033641
AccountCode:
Cause: 3
Cause-txt: No route to destination


[2016-09-12 15:01:07] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:07] DEBUG[25105][C-00001191] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:07] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:07] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:07] DEBUG[10730] devicestate.c: No provider found, checking channel drivers for SIP - 20202
[2016-09-12 15:01:07] DEBUG[10730] chan_sip.c: Checking device state for peer 20202
[2016-09-12 15:01:07] DEBUG[10730] devicestate.c: Changing state for SIP/20202 - state 1 (Not in use)
[2016-09-12 15:01:07] DEBUG[10730] devicestate.c: device 'SIP/20202' state '1'
[2016-09-12 15:01:07] DEBUG[10732] devicestate.c: Checking if I can find provider for "Custom" - number: DND20202
[2016-09-12 15:01:07] DEBUG[10732] devicestate.c: Checking provider SLA with Custom
[2016-09-12 15:01:07] DEBUG[10732] devicestate.c: Checking provider Meetme with Custom
[2016-09-12 15:01:07] DEBUG[10777] app_queue.c: Device 'SIP/20202' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2016-09-12 15:01:07] DEBUG[10732] devicestate.c: Checking provider Custom with Custom
[2016-09-12 15:01:07] DEBUG[10732] db.c: Unable to find key 'DND20202' in family 'CustomDevstate'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 422293984@10.204.10.178 (Checking From) --From tag 1204923471 --To-tag as61e4fc30
[2016-09-12 15:01:07] DEBUG[10748][C-00001191] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2016-09-12 15:01:07] DEBUG[10748][C-00001191] chan_sip.c: Stopping retransmission on '422293984@10.204.10.178' of Response 2: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 422293984@10.204.10.178
[2016-09-12 15:01:07] DEBUG[10748] rtp_engine.c: Destroyed RTP instance '0x7f528817b208'
[2016-09-12 15:01:07] DEBUG[10750] chan_iax2.c: ip callno count decremented to 3 for 192.168.211.200
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 10d565db061af2e43b19fc5b5afe060d@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:07] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: SIP call-id changed from '10d565db061af2e43b19fc5b5afe060d@172.22.0.119:5060' to '655278f65546e3a167d3f9211dfbadf6@172.22.255.6:5060'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 655278f65546e3a167d3f9211dfbadf6@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 6d85972e343b97bf54797cc10760b03b@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:07] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: SIP call-id changed from '6d85972e343b97bf54797cc10760b03b@172.22.0.119:5060' to '5df404652ee256fc545a686f02403304@172.22.255.6:5060'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 5df404652ee256fc545a686f02403304@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 655278f65546e3a167d3f9211dfbadf6@172.22.255.6:5060 (Checking To) --From tag as0f78c34d --To-tag 9b542a8029e79d6267bde4c7969c36f8
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Stopping retransmission on '655278f65546e3a167d3f9211dfbadf6@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 655278f65546e3a167d3f9211dfbadf6@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 5df404652ee256fc545a686f02403304@172.22.255.6:5060 (Checking To) --From tag as785cbb4c --To-tag f0687b5cf2c4e6c81c371e9a9c946333
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Stopping retransmission on '5df404652ee256fc545a686f02403304@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 5df404652ee256fc545a686f02403304@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 706aeb3e11529fad335cace9441d839a@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:07] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: SIP call-id changed from '706aeb3e11529fad335cace9441d839a@172.22.0.119:5060' to '5d799af855b2605e2acd0c605aacd49b@172.22.255.6:5060'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 5d799af855b2605e2acd0c605aacd49b@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 6239058f040681fc17b57cae0138a132@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:07] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: SIP call-id changed from '6239058f040681fc17b57cae0138a132@172.22.0.119:5060' to '7c682b8f2ad4b2522764afd1303b9ff6@172.22.255.6:5060'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 7c682b8f2ad4b2522764afd1303b9ff6@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 626380af59cb147e3b3b2ade610d477e@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:07] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: SIP call-id changed from '626380af59cb147e3b3b2ade610d477e@172.22.0.119:5060' to '5f69d3615d61a85c6a13dde94b9972cd@172.22.255.6:5060'
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 5f69d3615d61a85c6a13dde94b9972cd@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 5d799af855b2605e2acd0c605aacd49b@172.22.255.6:5060 (Checking To) --From tag as11b26c7f --To-tag 6be98b41d66dbff7ebb0212c6d15c0e5
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Stopping retransmission on '5d799af855b2605e2acd0c605aacd49b@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 5d799af855b2605e2acd0c605aacd49b@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 7c682b8f2ad4b2522764afd1303b9ff6@172.22.255.6:5060 (Checking To) --From tag as77475d98 --To-tag 645f797b680569f5c4b6e2f4abcaf23a
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Stopping retransmission on '7c682b8f2ad4b2522764afd1303b9ff6@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 7c682b8f2ad4b2522764afd1303b9ff6@172.22.255.6:5060
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: = Looking for Call ID: 5f69d3615d61a85c6a13dde94b9972cd@172.22.255.6:5060 (Checking To) --From tag as7d1e70c4 --To-tag a09a18716ae9828c0878fd122baf6846
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Stopping retransmission on '5f69d3615d61a85c6a13dde94b9972cd@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:07] DEBUG[10748] chan_sip.c: Destroying SIP dialog 5f69d3615d61a85c6a13dde94b9972cd@172.22.255.6:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 7b3a472c1a8ab1065bc7f99e5cd3e60f@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:08] DEBUG[10748] acl.c: For destination '10.204.10.21', our source address is '172.22.255.6'.
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: SIP call-id changed from '7b3a472c1a8ab1065bc7f99e5cd3e60f@172.22.0.119:5060' to '4dab1ac066b11fef1f8e646d61b4b49b@172.22.255.6:5060'
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 4dab1ac066b11fef1f8e646d61b4b49b@172.22.255.6:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.21:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: = Looking for Call ID: 4dab1ac066b11fef1f8e646d61b4b49b@172.22.255.6:5060 (Checking To) --From tag as757168af --To-tag fad0c9aa13fa6b98d31070a3ea1874de
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Stopping retransmission on '4dab1ac066b11fef1f8e646d61b4b49b@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Destroying SIP dialog 4dab1ac066b11fef1f8e646d61b4b49b@172.22.255.6:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: = Looking for Call ID: 73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060 (Checking To) --From tag as1876a1ab --To-tag 2109020990
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Acked pending invite 102
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Stopping retransmission on '73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: SIP response 200 to standard invite
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing session-level SDP o=20001 880926956 1829215281 IN IP4 10.204.10.29... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED OR FAILED.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: Splitting '10.204.10.29' into...
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: ...host '10.204.10.29' and port ''.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing session-level SDP c=IN IP4 10.204.10.29... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Setting payload 8 based on m type on 0x7f53001abeb0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Setting payload 0 based on m type on 0x7f53001abeb0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Setting payload 101 based on m type on 0x7f53001abeb0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f52d4cea9f8'
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Copying payload 0 from 0x7f53001abeb0 to 0x7f52d4ceabc0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Copying payload 8 from 0x7f53001abeb0 to 0x7f52d4ceabc0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] rtp_engine.c: Copying payload 101 from 0x7f53001abeb0 to 0x7f52d4ceabc0
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f52d4cea9f8'
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: We're settling with these formats: (ulaw|alaw)
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: We have an owner, now see if we need to change this call
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Updating call counter for outgoing call
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: build_route: Contact hop: <sip:20001@10.204.10.29:5060>
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: Splitting '10.204.10.29:5060' into...
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: ...host '10.204.10.29' and port '5060'.
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: No provider found, checking channel drivers for SIP - 20001
[2016-09-12 15:01:08] DEBUG[10730] chan_sip.c: Checking device state for peer 20001
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: Changing state for SIP/20001 - state 2 (In use)
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: device 'SIP/20001' state '2'
[2016-09-12 15:01:08] DEBUG[10777] app_queue.c: Device 'SIP/20001' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: Splitting '10.204.10.29:5060' into...
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] netsock2.c: ...host '10.204.10.29' and port '5060'.
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking if I can find provider for "Custom" - number: DND20001
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider SLA with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Meetme with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Custom with Custom
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Trying to put 'ACK sip:200' onto UDP socket destined for 10.204.10.29:5060
[2016-09-12 15:01:08] DEBUG[10732] db.c: Unable to find key 'DND20001' in family 'CustomDevstate'
[2016-09-12 15:01:08] DEBUG[10732] app_queue.c: Extension '20001@ext-local' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2016-09-12 15:01:08] VERBOSE[25087][C-00001190] app_dial.c: -- Connected line update to SIP/20272-00001aba prevented.
[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: ExtensionStatus
Privilege: call,all
Exten: 20001
Context: ext-local
Hint: SIP/20001&Custom:DND20001,CustomPresence:20001
Status: 1


[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: No provider found, checking channel drivers for SIP - 20001
[2016-09-12 15:01:08] DEBUG[10730] chan_sip.c: Checking device state for peer 20001
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: Changing state for SIP/20001 - state 2 (In use)
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: device 'SIP/20001' state '2'
[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/20001-00001abb
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 20001
CallerIDName: Serdyukov
ConnectedLineNum: 20272
ConnectedLineName: Sekretar GP
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[10777] app_queue.c: Device 'SIP/20001' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking if I can find provider for "Custom" - number: DND20001
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider SLA with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Meetme with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Custom with Custom
[2016-09-12 15:01:08] VERBOSE[25087][C-00001190] app_dial.c: -- SIP/20001-00001abb answered SIP/20272-00001aba
[2016-09-12 15:01:08] DEBUG[10732] db.c: Unable to find key 'DND20001' in family 'CustomDevstate'
[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: DIALSTATUS
Value: ANSWER
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: DIALEDPEERNAME
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: DIALEDPEERNUMBER
Value: 20001
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: SIP answering channel: SIP/20272-00001aba
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: No provider found, checking channel drivers for SIP - 20272
[2016-09-12 15:01:08] DEBUG[10730] chan_sip.c: Checking device state for peer 20272
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: Changing state for SIP/20272 - state 2 (In use)
[2016-09-12 15:01:08] DEBUG[10730] devicestate.c: device 'SIP/20272' state '2'
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: Setting framing from config on incoming call
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: ** Our capability: (ulaw|alaw) Video flag: True Text flag: True
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: ** Our prefcodec: (nothing)
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: -- Done with adding codecs to SDP
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw)
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.204.10.190:5060
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] features.c: bridge answer set, chan answer set
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] features.c: Removing dialed interfaces datastore on SIP/20001-00001abb since we're bridging
[2016-09-12 15:01:08] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking if I can find provider for "Custom" - number: DND20272
[2016-09-12 15:01:08] DEBUG[10777] app_queue.c: Device 'SIP/20272' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider SLA with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Meetme with Custom
[2016-09-12 15:01:08] DEBUG[10732] devicestate.c: Checking provider Custom with Custom
[2016-09-12 15:01:08] DEBUG[10732] db.c: Unable to find key 'DND20272' in family 'CustomDevstate'
[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPEER
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPEER
Value: SIP/20272-00001aba
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/20272-00001aba
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 20272
CallerIDName: Sekretar GP
ConnectedLineNum: 20001
ConnectedLineName: Serdyukov
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/20001-00001abb
Uniqueid: 1473681658.9635
AccountCode:
OldAccountCode:


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: SIP/20272-00001aba
Channel2: SIP/20001-00001abb
Uniqueid1: 1473681658.9634
Uniqueid2: 1473681658.9635
CallerID1: 20272
CallerID2: 20001


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPEER
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPVTCALLID
Value: 73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPEER
Value: SIP/20272-00001aba
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPVTCALLID
Value: 20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPEER
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPVTCALLID
Value: 73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPEER
Value: SIP/20272-00001aba
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPVTCALLID
Value: 20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:08] DEBUG[10724] res_odbc.c: odbc_release_obj2(0x7f52d42f7008) called (obj->txf = (nil))
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: 0x7f52882acec0 -- Probation learning mode pass with source address 10.204.10.190:18176
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] chan_sip.c: Oooh, format changed to ulaw
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] channel.c: Set channel SIP/20272-00001aba to read format alaw
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] channel.c: Set channel SIP/20272-00001aba to write format alaw
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160
[2016-09-12 15:01:08] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7f52d4cea9f8'
[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPEER
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPVTCALLID
Value: 73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPEER
Value: SIP/20272-00001aba
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPVTCALLID
Value: 20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPEER
Value: SIP/20001-00001abb
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20272-00001aba
Variable: BRIDGEPVTCALLID
Value: 73f20abe2c24a7111b18db5455a59a45@172.22.255.6:5060
Uniqueid: 1473681658.9634


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPEER
Value: SIP/20272-00001aba
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[16109] manager.c: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/20001-00001abb
Variable: BRIDGEPVTCALLID
Value: 20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190
Uniqueid: 1473681658.9635


[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #2735952))
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.204.10.190:5060
[2016-09-12 15:01:08] DEBUG[10748] chan_sip.c: = Looking for Call ID: 20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190 (Checking From) --From tag 20bbc093e0c90c4497f4a87b-70a27c18 --To-tag as774e32bf
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2016-09-12 15:01:08] DEBUG[10748][C-00001190] chan_sip.c: Stopping retransmission on '20bbc093-e0c9001a-324b7d9d-59eb2e16@10.204.10.190' of Response 102: Match Found
[2016-09-12 15:01:09] DEBUG[25087][C-00001190] res_rtp_asterisk.c: 0x7f52d406c2a0 -- Probation learning mode pass with source address 10.204.10.29:10086
[2016-09-12 15:01:09] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[2016-09-12 15:01:09] DEBUG[25087][C-00001190] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Allocating new SIP dialog for 4400eb466b13f6fc77df2da76f2540de@172.22.0.119:5060 - OPTIONS (No RTP)
[2016-09-12 15:01:09] DEBUG[10748] acl.c: For destination '10.204.10.180', our source address is '172.22.255.6'.
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.22.255.6:5060
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: SIP call-id changed from '4400eb466b13f6fc77df2da76f2540de@172.22.0.119:5060' to '052edd4a7d3e225c68230def3e1e4dbb@172.22.255.6:5060'
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Initializing initreq for method OPTIONS - callid 052edd4a7d3e225c68230def3e1e4dbb@172.22.255.6:5060
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.204.10.180:5060
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: = Looking for Call ID: 052edd4a7d3e225c68230def3e1e4dbb@172.22.255.6:5060 (Checking To) --From tag as0a986017 --To-tag 001563ef4a1d027b1340c081-0c83b24b
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Stopping retransmission on '052edd4a7d3e225c68230def3e1e4dbb@172.22.255.6:5060' of Request 102: Match Found
[2016-09-12 15:01:09] DEBUG[10748] chan_sip.c: Destroying SIP dialog 052edd4a7d3e225c68230def3e1e4dbb@172.22.255.6:5060
[2016-09-12 15:01:10] DEBUG[25114] manager.c: Running action 'Login'
april22
Сообщения: 2187
Зарегистрирован: 09 июл 2012, 09:47

Re: Переадресация dongle на другой астериск

Сообщение april22 »

а что в консоли на том астере . где стоит донгл ?
Своими вопросами , вы загоняете меня в ГУГЛЬ.
oddity
Сообщения: 13
Зарегистрирован: 05 сен 2016, 14:40

Re: Переадресация dongle на другой астериск

Сообщение oddity »

А там никак не реагирует. Вызов не проходит....
april22
Сообщения: 2187
Зарегистрирован: 09 июл 2012, 09:47

Re: Переадресация dongle на другой астериск

Сообщение april22 »

что то я в логе (не привычный)
не нашёл ни чего подобного типа
dial (SIP\что то типа транка в удаленную сторону \${EXTEN})

core set verbose 4
Звонок - лог под споллер .

маршруты точно есть ?! ;-)
Своими вопросами , вы загоняете меня в ГУГЛЬ.
Ответить
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