Содержание
- Установка Asterisk и FreePBX из репозитория в CentOS
- FreeRDP параметры подключения
- Windows Skype в Ubuntu
- При присоединению к домену произошла ошибка: Совпадает ИД безопасности
- 6 комментариев
- Error opening firmware directory var lib asterisk firmware iax
- Re: Asterisk FreePBX
- Re: Asterisk FreePBX
Установка Asterisk и FreePBX из репозитория в CentOS
Приведу самый простой способ установки. Нам не понадобится собирать из исходников и ковыряться в ошибках консоли. Обновляем CentOS:
Первый вариант установки
Переходим к добавлению нужных репозиториев для установки Asterisk
Устанавливаем репозиторий asterisk
Устанавливаем репозиторий digium
и переходим к установке
Второй вариант установки
Находим строчку TTY=9 и комментируем, чтобы получилось #TTY=9
Теперь можно зайти на сервер http://IP-адрес_вашего-сервера
Возможные ошибки
1. Ошибка при скачивании репозитория с помощью команды wget
2. Ошибка при установке asterisk и freepbx (yum -y install freepbx)
нет пакета в репозитории, устанавливайте отдельно или ищите нужный репозиторий
Документация и ссылки по Asterisk и FreePBX для вкуривания:
- http://www.freepbx.org/support/documentation/module-documentation
- http://asteriskonvps.com/category/asterisk/
- http://issues.freepbx.org/secure/Dashboard.jspa
- https://wiki.asterisk.org/wiki/display/AST/Home
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS6%2FRedHatEnterpriseLinux6%29
FreeRDP параметры подключения
Windows Skype в Ubuntu
6 комментариев
Замечательная статья, вот возник только 1 вопросик а как к этому всему можно подключить h.323 чтоб не компилировать его сначала а потом астериску *SORRY*
пока только разбираюсь понемногу и пишу статьи
думаю со временем и к этому доберусь )
😉
Эх когда даёмс /usr/sbin/safe_asterisk то в в ответ получаем :
[root@asterisk
]# [Oct 29 17:09:33] NOTICE[8915]: cdr.c:1622 do_reload: CDR simple logging enabled.
[Oct 29 17:09:33] NOTICE[8915]: loader.c:1192 load_modules: 201 modules will be loaded.
[Oct 29 17:09:33] NOTICE[8915]: res_odbc.c:1889 load_module: res_odbc loaded.
[Oct 29 17:09:33] NOTICE[8915]: res_smdi.c:1418 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Oct 29 17:09:33] NOTICE[8915]: config.c:2355 ast_config_engine_register: Registered Config Engine odbc
[Oct 29 17:09:33] WARNING[8915]: res_musiconhold.c:1107 moh_scan_files: Cannot open dir /var/lib/asterisk/moh or dir does not exist
[Oct 29 17:09:33] WARNING[8915]: res_musiconhold.c:1966 load_module: No music on hold classes configured, disabling music on hold.
[Oct 29 17:09:33] ERROR[8915]: chan_motif.c:2499 custom_connection_handler: Connection ‘local-jabber-account’ configured on endpoint ‘jingle-endpoint’ could not be found
[Oct 29 17:09:33] ERROR[8915]: config_options.c:581 aco_process_var: Error parsing connection=local-jabber-account at line 81 of
[Oct 29 17:09:33] ERROR[8915]: config_options.c:406 process_category: In motif.conf: Processing options for jingle-endpoint failed
[Oct 29 17:09:33] ERROR[8915]: chan_motif.c:2565 load_module: Unable to read config file motif.conf. Not loading module.
[Oct 29 17:09:33] WARNING[8915]: chan_iax2.c:3299 reload_firmware: Error opening firmware directory ‘/var/lib/asterisk/firmware/iax’: No such file or directory
[Oct 29 17:09:33] NOTICE[8915]: chan_skinny.c:7736 config_load: Configuring skinny from skinny.conf
[Oct 29 17:09:33] WARNING[8915]: chan_skinny.c:7765 config_load: Unable to get our IP address, Skinny disabled
[Oct 29 17:09:33] WARNING[8915]: chan_dahdi.c:18577 process_dahdi: Ignoring any changes to ‘userbase’ (on reload) at line 23.
[Oct 29 17:09:33] WARNING[8915]: chan_dahdi.c:18577 process_dahdi: Ignoring any changes to ‘vmsecret’ (on reload) at line 31.
[Oct 29 17:09:33] WARNING[8915]: chan_dahdi.c:18577 process_dahdi: Ignoring any changes to ‘hassip’ (on reload) at line 35.
[Oct 29 17:09:33] WARNING[8915]: chan_dahdi.c:18577 process_dahdi: Ignoring any changes to ‘hasiax’ (on reload) at line 39.
[Oct 29 17:09:33] WARNING[8915]: chan_dahdi.c:18577 process_dahdi: Ignoring any changes to ‘hasmanager’ (on reload) at line 47.
[Oct 29 17:09:33] ERROR[8915]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(«asterisk.kh.dnepro.dom», «(null)», …): Name or service not known
[Oct 29 17:09:33] WARNING[8915]: acl.c:833 resolve_first: Unable to lookup ‘asterisk.kh.dnepro.dom’
[Oct 29 17:09:33] NOTICE[8915]: confbridge/conf_config_parser.c:1315 verify_default_profiles: Adding default_user profile to app_confbridge
[Oct 29 17:09:33] NOTICE[8915]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:164 pbx_load_module: Starting AEL load process.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:177 pbx_load_module: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:180 pbx_load_module: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:187 pbx_load_module: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:192 pbx_load_module: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[Oct 29 17:09:33] NOTICE[8915]: pbx_ael.c:195 pbx_load_module: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
[Oct 29 17:09:33] WARNING[8915]: pbx_dundi.c:4835 set_config: Unable to look up host ‘asterisk.kh.dnepro.dom’
[Oct 29 17:09:33] ERROR[8915]: codec_dahdi.c:623 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory
эх ну невезет так не везет %)
Все нормально стало. Просто я ошибся в 1 месте.
Эм хотелось бы внести маленькую поправку уже есть нормальная не бета версия freepbx-2.11.0 =)
Источник
Error opening firmware directory var lib asterisk firmware iax
Sergey_M » 21 май 2014, 17:59
Простите за мою некомпетентность, в общем образ качал отсюда http://asterisk.org/downloads. Как я понял, это порезаный в хлам CentOS, с необходимыми модулями Астериска.
По поводу конфигурационных файлов, в корне Asterisk’a есть следующие файлы:
— extensions.conf
— extensions_additional.conf
— extensions_custom.conf
— extensions_custom.conf.sample
— extensions_override_freepbx.conf
— iax.conf
— iax_additional.conf
— iax_custom.conf
— iax_custom_post.conf
— iax_general_additional.conf
— iax_general_custom.conf
— iax_registrations.conf
— iax_registrations_custom.conf
Эм, я так понимаю Вам нужно это?
Звонок с номера 40000 (MSK-SOK-ASTER-01) на номер 50000 (MSK-SEM-ASTER-01)
Код: выделить все MSK-SOK-ASTER-01*CLI>
INVITE sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: «Sergey Chenchikov» ;tag=1507904733
To:
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Contact: «Sergey Chenchikov»
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.3.30
Privacy: none
P-Preferred-Identity: «Sergey Chenchikov»
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 379
v=0
o=40000 8000 8000 IN IP4 10.101.10.15
s=SIP Call
c=IN IP4 10.101.10.15
t=0 0
m=audio 5004 RTP/AVP 18 8 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
— (16 headers 18 lines) —
Sending to 10.101.10.15:5060 (no NAT)
Sending to 10.101.10.15:5060 (no NAT)
Using INVITE request as basis request — 494005238-5060-35@BA.BAB.BA.BF
Found peer ‘40000’ for ‘40000’ from 10.101.10.15:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us — (ulaw|alaw), peer — audio=(g723|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined — (alaw)
Non-codec capabilities (dtmf): us — 0x1 (telephone-event|), peer — 0x1 (telephone-event|), combined — 0x1 (telephone-event|)
Peer audio RTP is at port 10.101.10.15:5004
Looking for 50000 in from-internal (domain 10.101.10.251)
list_route: hop:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: «Sergey Chenchikov» ;tag=1507904733
To:
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Length: 0
Audio is at 13746
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: «Sergey Chenchikov» ;tag=1507904733
To: ;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Require: timer
Content-Length: 237
v=0
o=root 1593559795 1593559795 IN IP4 10.101.10.251
s=Asterisk PBX 11.9.0
c=IN IP4 10.101.10.251
t=0 0
m=audio 13746 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
CANCEL sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: «Sergey Chenchikov» ;tag=1507904733
To:
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 CANCEL
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.3.30
Content-Length: 0
— (9 headers 0 lines) —
Sending to 10.101.10.15:5060 (no NAT)
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: «Sergey Chenchikov» ;tag=1507904733
To: ;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: «Sergey Chenchikov» ;tag=1507904733
To: ;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 CANCEL
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
ACK sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: «Sergey Chenchikov» ;tag=1507904733
To: ;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 ACK
Content-Length: 0
— (7 headers 0 lines) —
Really destroying SIP dialog ‘494005238-5060-35@BA.BAB.BA.BF’ Method: ACK
На втором сервере в момент звонка вообще тишина. Но зато там даже без звонка происходит нечто вот такое:
Код: выделить все
Reliably Transmitting (no NAT) to 10.101.10.25:5063:
OPTIONS sip:50000@10.101.10.25:5063 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.252:5060;branch=z9hG4bK19881a4e
Max-Forwards: 70
From: «Unknown» ;tag=as4cb89ad3
To:
Contact:
Call-ID: 3400de917b8e850e36fd0f3345117c8d@10.101.10.252:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.9.0)
Date: Wed, 21 May 2014 13:54:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.10.252:5060;branch=z9hG4bK19881a4e
From: «Unknown» ;tag=as4cb89ad3
To: ;tag=678336553
Call-ID: 3400de917b8e850e36fd0f3345117c8d@10.101.10.252:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T12P 5.60.14.4
Content-Length: 0
— (8 headers 0 lines) —
Really destroying SIP dialog ‘3400de917b8e850e36fd0f3345117c8d@10.101.10.252:5060’ Method: OPTIONS
А теперь что происходит при звонке с номера 5000 (MSK-SEM-ASTER-01) на номер 40000 (MSK-SOK-ASTER-01)
Код: выделить все Reliably Transmitting (no NAT) to 10.101.10.25:5063:
OPTIONS sip:50000@10.101.10.25:5063 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.252:5060;branch=z9hG4bK6b7df0e8
Max-Forwards: 70
From: «Unknown» ;tag=as5f7deb9d
To:
Contact:
Call-ID: 2969cd8e23dfe0df0724e197351fb5b5@10.101.10.252:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.9.0)
Date: Wed, 21 May 2014 13:55:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.10.252:5060;branch=z9hG4bK6b7df0e8
From: «Unknown» ;tag=as5f7deb9d
To: ;tag=138544125
Call-ID: 2969cd8e23dfe0df0724e197351fb5b5@10.101.10.252:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T12P 5.60.14.4
Content-Length: 0
— (8 headers 0 lines) —
Really destroying SIP dialog ‘2969cd8e23dfe0df0724e197351fb5b5@10.101.10.252:5060’ Method: OPTIONS
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.101.10.25:5063;branch=z9hG4bK1595238395;received=10.101.10.25
From: ;tag=1599401414
To: ;tag=as3bf72332
Call-ID: 473865052@10.101.10.25
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
[2014-05-21 17:55:28] WARNING[12841][C-00000005]: channel.c:4840 ast_prod: Prodding channel ‘SIP/50000-00000005’ failed
ACK sip:40000@10.101.10.252 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.25:5063;branch=z9hG4bK1595238395
From: ;tag=1599401414
To: ;tag=as3bf72332
Call-ID: 473865052@10.101.10.25
CSeq: 1 ACK
Content-Length: 0
— (7 headers 0 lines) —
Really destroying SIP dialog ‘473865052@10.101.10.25’ Method: ACK
Замечу еще один нюанс: 10.101.10.15 и 10.101.10.25, это IP адреса телефонов.
Так же лог ничего не выдает на MSK-SOK-ASTER-01
Re: Asterisk FreePBX
april22 » 22 май 2014, 08:51
дебаг конечно интерсно — но я просил просто консоль астериска — что там происходит .
а так настраивайте транк , у вас ни чего не прилетает на второй астериск поэтому и нет звонка .
если делали с веб морды — то и делайте там , не ковыряйте без ОЧЕНЬ большой надобности контексты , а особенно extensions_additional.conf
Re: Asterisk FreePBX
Sergey_M » 22 май 2014, 10:01
Пробовал все настроить через WEB морду, транк на WEB морде показывает поднявшимся, маршрут прописал, указал на данный транк и с одной стороны и с другой, но связь так и не появилась.
При настройке с WEB морды в консоли астера вообще ничего не происходит.
Попробовал перегрузить SIP модуль ( sip reload ), выдал следующее:
Код: выделить все MSK-SOK-ASTER-01*CLI> sip reload
[2014-05-22 10:05:17] ERROR[1753]: phone_message.c:1645 build_dialplan_routing: Unable to build dialplan routing — invalid license
[2014-05-22 10:05:17] ERROR[1753]: phone_users.c:4051 process_message_config: accept_outofcall_message must be enabled in sip.conf for res_digium_phone to function properly
[2014-05-22 10:05:17] ERROR[1753]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(«MSK-SOK-ASTER-01», «(null)», . ): Name or service not known
[2014-05-22 10:05:17] WARNING[1753]: acl.c:833 resolve_first: Unable to lookup ‘MSK-SOK-ASTER-01’
При перезагрузке IAX2:
Код: выделить все MSK-SOK-ASTER-01*CLI> iax2 reload
[2014-05-22 10:06:50] WARNING[32518]: chan_iax2.c:3385 reload_firmware: Error opening firmware directory ‘/var/lib/asterisk/firmware/iax’: No such file or directory
[2014-05-22 10:06:50] NOTICE[32518]: iax2-provision.c:558 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.
А если перегрузить DIALPLAN, то там вообще куча варнингов!
Источник
MSK-SOK-ASTER-01*CLI>
<--- SIP read from UDP:10.101.10.15:5060 --->
INVITE sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Contact: "Sergey Chenchikov" <sip:40000@10.101.10.15:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.3.30
Privacy: none
P-Preferred-Identity: "Sergey Chenchikov" <sip:40000@10.101.10.251>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 379
v=0
o=40000 8000 8000 IN IP4 10.101.10.15
s=SIP Call
c=IN IP4 10.101.10.15
t=0 0
m=audio 5004 RTP/AVP 18 8 4 9 97 2 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 18 lines) ---
Sending to 10.101.10.15:5060 (no NAT)
Sending to 10.101.10.15:5060 (no NAT)
Using INVITE request as basis request - 494005238-5060-35@BA.BAB.BA.BF
Found peer '40000' for '40000' from 10.101.10.15:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(g723|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.101.10.15:5004
Looking for 50000 in from-internal (domain 10.101.10.251)
list_route: hop: <sip:40000@10.101.10.15:5060>
<--- Transmitting (no NAT) to 10.101.10.15:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:50000@10.101.10.251:5060>
Content-Length: 0
<------------>
Audio is at 13746
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 10.101.10.15:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:50000@10.101.10.251:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 237
v=0
o=root 1593559795 1593559795 IN IP4 10.101.10.251
s=Asterisk PBX 11.9.0
c=IN IP4 10.101.10.251
t=0 0
m=audio 13746 RTP/AVP 8 101
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:10.101.10.15:5060 --->
CANCEL sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 CANCEL
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.3.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 10.101.10.15:5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 10.101.10.15:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 INVITE
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.101.10.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;received=10.101.10.15;rport=5060
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 CANCEL
Server: FPBX-2.11.0(11.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:10.101.10.15:5060 --->
ACK sip:50000@10.101.10.251 SIP/2.0
Via: SIP/2.0/UDP 10.101.10.15:5060;branch=z9hG4bK1201722838;rport
From: "Sergey Chenchikov" <sip:40000@10.101.10.251>;tag=1507904733
To: <sip:50000@10.101.10.251>;tag=as17c9a2c8
Call-ID: 494005238-5060-35@BA.BAB.BA.BF
CSeq: 340 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '494005238-5060-35@BA.BAB.BA.BF' Method: ACK
2 zavndw, спасибо за пример.
У меня выходит такая ошибка:
Код:
06-02/09:55:20 INFO [http-bio-9088-exec-94] AMIManager — Reinit..
06-02/09:55:56 INFO [Thread-6] Config — Reload tasks config.
06-02/09:56:08 ERROR [Thread-7] AMIManager — Connection timed out
java.net.ConnectException: Connection timed out
at java.net.PlainSocketImpl.socketConnect(Native Method)
at java.net.AbstractPlainSocketImpl.doConnect(AbstractPlainSocketImpl.java:350)
at java.net.AbstractPlainSocketImpl.connectToAddress(AbstractPlainSocketImpl.java:206)
at java.net.AbstractPlainSocketImpl.connect(AbstractPlainSocketImpl.java:188)
at java.net.SocksSocketImpl.connect(SocksSocketImpl.java:392)
at java.net.Socket.connect(Socket.java:589)
at org.asteriskjava.util.internal.SocketConnectionFacadeImpl.<init>(SocketConnectionFacadeImpl.java:93)
at org.asteriskjava.util.internal.SocketConnectionFacadeImpl.<init>(SocketConnectionFacadeImpl.java:66)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.createSocket(ManagerConnectionImpl.java:765)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.connect(ManagerConnectionImpl.java:744)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.doLogin(ManagerConnectionImpl.java:498)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.login(ManagerConnectionImpl.java:446)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.login(ManagerConnectionImpl.java:431)
at org.asteriskjava.manager.DefaultManagerConnection.login(DefaultManagerConnection.java:294)
at ru.bgcrm.plugin.asterisk.AmiEventListener.run(AmiEventListener.java:60)
06-02/09:56:08 ERROR [http-bio-9088-exec-94] AMIManager — Logoff may only be perfomed when in state CONNECTED or RECONNECTING, but connection is in state DISCONNECTED
java.lang.IllegalStateException: Logoff may only be perfomed when in state CONNECTED or RECONNECTING, but connection is in state DISCONNECTED
at org.asteriskjava.manager.internal.ManagerConnectionImpl.logoff(ManagerConnectionImpl.java:772)
at org.asteriskjava.manager.DefaultManagerConnection.logoff(DefaultManagerConnection.java:305)
at ru.bgcrm.plugin.asterisk.AmiEventListener.logoff(AmiEventListener.java:90)
at ru.bgcrm.plugin.asterisk.AMIManager.init(AMIManager.java:56)
at ru.bgcrm.plugin.asterisk.AMIManager.access$0(AMIManager.java:48)
at ru.bgcrm.plugin.asterisk.AMIManager$1.notify(AMIManager.java:41)
at ru.bgcrm.plugin.asterisk.AMIManager$1.notify(AMIManager.java:1)
at ru.bgcrm.event.EventProcessor.processingEvent(EventProcessor.java:205)
at ru.bgcrm.event.EventProcessor.processEvent(EventProcessor.java:140)
at ru.bgcrm.event.EventProcessor.processEvent(EventProcessor.java:286)
at ru.bgcrm.event.EventProcessor.processEvent(EventProcessor.java:97)
at ru.bgcrm.struts.action.admin.ConfigAction.update(ConfigAction.java:117)
at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method)
at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:62)
at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:43)
at java.lang.reflect.Method.invoke(Method.java:497)
at ru.bgcrm.struts.action.BaseAction$Invoker.invoke(BaseAction.java:532)
at ru.bgcrm.struts.action.BaseAction.dispatchMethod(BaseAction.java:241)
at org.apache.struts.actions.DispatchAction.execute(DispatchAction.java:170)
at org.apache.struts.chain.commands.servlet.ExecuteAction.execute(ExecuteAction.java:58)
at org.apache.struts.chain.commands.AbstractExecuteAction.execute(AbstractExecuteAction.java:67)
at org.apache.struts.chain.commands.ActionCommandBase.execute(ActionCommandBase.java:51)
at org.apache.commons.chain.impl.ChainBase.execute(ChainBase.java:191)
at org.apache.commons.chain.generic.LookupCommand.execute(LookupCommand.java:305)
at org.apache.commons.chain.impl.ChainBase.execute(ChainBase.java:191)
at org.apache.struts.chain.ComposableRequestProcessor.process(ComposableRequestProcessor.java:283)
at org.apache.struts.action.ActionServlet.process(ActionServlet.java:1913)
at org.apache.struts.action.ActionServlet.doPost(ActionServlet.java:462)
at javax.servlet.http.HttpServlet.service(HttpServlet.java:641)
at javax.servlet.http.HttpServlet.service(HttpServlet.java:722)
at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:305)
at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:210)
at ru.bgcrm.servlet.filter.SetRequestParamsFilter.doFilter(SetRequestParamsFilter.java:43)
at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:243)
at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:210)
at ru.bgcrm.servlet.filter.AuthFilter.doFilter(AuthFilter.java:150)
at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:243)
at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:210)
at ru.bgcrm.servlet.filter.SetCharacterEncodingFilter.doFilter(SetCharacterEncodingFilter.java:38)
at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:243)
at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:210)
at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:224)
at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:169)
at org.apache.catalina.authenticator.AuthenticatorBase.invoke(AuthenticatorBase.java:472)
at org.apache.catalina.valves.AccessLogValve.invoke(AccessLogValve.java:928)
at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:168)
at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:98)
at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:118)
at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:407)
at org.apache.coyote.http11.AbstractHttp11Processor.process(AbstractHttp11Processor.java:987)
at org.apache.coyote.AbstractProtocol$AbstractConnectionHandler.process(AbstractProtocol.java:539)
at org.apache.tomcat.util.net.JIoEndpoint$SocketProcessor.run(JIoEndpoint.java:300)
at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1142)
at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:617)
at java.lang.Thread.run(Thread.java:745)
Конфигурация в crm:
Код:
# Типы сообщений
messageType.1.title=Звонки
messageType.1.class=ru.bgcrm.dao.message.MessageTypeCall
# AMI
asterisk:amiManager.1.messageTypeId=1
asterisk:amiManager.1.host=10.1.1.6
asterisk:amiManager.1.port=5038
asterisk:amiManager.1.login=admin
asterisk:amiManager.1.pswd=amp111
createOnStart=ru.bgcrm.plugin.asterisk.AMIManager,ru.bgcrm.event.listener.MessageTypeCallListener
В астериске лог:
Код:
[2016-06-02 09:43:07] WARNING[18796]: pbx_config.c:1837 pbx_load_config: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 127 of extensions.conf
[2016-06-02 09:43:07] WARNING[18796]: pbx.c:8391 ast_context_verify_includes: Context ‘from-internal-xfer’ tries to include nonexistent context ‘from-internal-custom’
…. [много почти одинакоовых строк]
[2016-06-02 09:43:07] WARNING[18796]: pbx.c:8391 ast_context_verify_includes: Context ‘app-cf-toggle’ tries to include nonexistent context ‘app-cf-toggle-custom’
[2016-06-02 09:43:07] WARNING[18796]: res_odbc.c:503 load_odbc_config: The ‘pooling’, ‘shared_connections’, ‘limit’, and ‘idlecheck’ options are deprecated. Please see UPGRADE.txt for information
[2016-06-02 09:43:07] NOTICE[18796]: res_odbc.c:585 load_odbc_config: Registered ODBC class ‘asteriskcdrdb’ dsn->[MySQL-asteriskcdrdb]
[2016-06-02 09:43:07] ERROR[18796]: res_sorcery_config.c:240 sorcery_config_internal_load: Unable to load config file ‘pjproject.conf’
[2016-06-02 09:43:07] ERROR[18804]: phone_message.c:1654 msg_build_dialplan_routing: Unable to build dialplan routing — invalid license
[2016-06-02 09:43:07] NOTICE[18799]: sorcery.c:1374 sorcery_object_load: Type ‘system’ is not reloadable, maintaining previous values
[2016-06-02 09:43:07] WARNING[18796]: res_digium_phone.c:2047 reload: No Valid DPMA License found. Module is loaded but disabled. Please reload module once valid license is installed.
[2016-06-02 09:43:07] WARNING[18796]: iax2/firmware.c:234 iax_firmware_reload: Error opening firmware directory ‘/var/lib/asterisk/firmware/iax’: No such file or directory
[2016-06-02 09:43:07] NOTICE[18796]: iax2/provision.c:562 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.
[2016-06-02 09:43:07] WARNING[18517]: chan_sip.c:31135 build_peer: ‘tcp’ is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used.
[2016-06-02 09:43:07] WARNING[18796]: chan_dahdi.c:18963 process_dahdi: Ignoring any changes to ‘signalling’ (on reload) at line 12.
[2016-06-02 09:43:07] WARNING[18796]: chan_dahdi.c:18963 process_dahdi: Ignoring any changes to ‘rxwink’ (on reload) at line 13.
[2016-06-02 09:43:07] NOTICE[18796]: confbridge/conf_config_parser.c:2076 verify_default_profiles: Adding default_menu menu to app_confbridge
[2016-06-02 09:43:07] NOTICE[18796]: app_queue.c:8742 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.
[2016-06-02 09:43:07] WARNING[18796]: app_flite.c:77 read_config: Flite: Unable to read config file flite.conf. Using default settings
manager.conf создает freepbx:
Код:
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+
[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000
[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=10.1.1.0/255.255.255.0
permit=127.0.0.1/255.255.255.0
read = call
writetimeout = 5000
Затем, спустя некоторое время AMI снова пытается подключиться, и выходит такая ошибка:
Код:
06-02/10:03:22 INFO [Thread-7] ManagerConnectionImpl — Connecting to 10.1.1.6:5038
06-02/10:05:30 ERROR [Thread-7] AMIManager — Connection timed out
java.net.ConnectException: Connection timed out
at java.net.PlainSocketImpl.socketConnect(Native Method)
at java.net.AbstractPlainSocketImpl.doConnect(AbstractPlainSocketImpl.java:350)
at java.net.AbstractPlainSocketImpl.connectToAddress(AbstractPlainSocketImpl.java:206)
at java.net.AbstractPlainSocketImpl.connect(AbstractPlainSocketImpl.java:188)
at java.net.SocksSocketImpl.connect(SocksSocketImpl.java:392)
at java.net.Socket.connect(Socket.java:589)
at org.asteriskjava.util.internal.SocketConnectionFacadeImpl.<init>(SocketConnectionFacadeImpl.java:93)
at org.asteriskjava.util.internal.SocketConnectionFacadeImpl.<init>(SocketConnectionFacadeImpl.java:66)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.createSocket(ManagerConnectionImpl.java:765)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.connect(ManagerConnectionImpl.java:744)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.doLogin(ManagerConnectionImpl.java:498)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.login(ManagerConnectionImpl.java:446)
at org.asteriskjava.manager.internal.ManagerConnectionImpl.login(ManagerConnectionImpl.java:431)
at org.asteriskjava.manager.DefaultManagerConnection.login(DefaultManagerConnection.java:294)
at ru.bgcrm.plugin.asterisk.AmiEventListener.run(AmiEventListener.java:60)
_________________
Сервер: вер. 7.1.1118 / 16.04.2019 22:18:28
os: Linux; java: Java HotSpot(TM) 64-Bit Server VM, v.1.8.0_202
Приведу самый простой способ установки. Нам не понадобится собирать из исходников и ковыряться в ошибках консоли. Обновляем CentOS:
yum -y update
Первый вариант установки
Переходим к добавлению нужных репозиториев для установки Asterisk
Устанавливаем репозиторий asterisk
wget http://packages.asterisk.org/centos/centos-asterisk.repo -O /etc/yum.repos.d/centos-asterisk.repo
Устанавливаем репозиторий digium
wget http://packages.digium.com/centos/centos-digium.repo -O /etc/yum.repos.d/centos-digium.repo
Теперь обновляемся
yum -y update
и переходим к установке
yum -y install freepbx
Второй вариант установки
yum -y install dnsmasq rpm -Uvh http://packages.asterisk.org/centos/6/current/x86_64/RPMS/asterisknow-version-3.0.0-1_centos6.noarch.rpm yum -y update yum -y install asterisk asterisk-configs --enablerepo=asterisk-11 yum -y install dahdi-linux dahdi-tools libpri yum -y install gcc gcc-c++ wget bison mysql-devel mysql-server php php-mysql php-process php-pear php-mbstring tftp-server httpd make ncurses-devel libtermcap-devel sendmail sendmail-cf caching-nameserver sox newt-devel libxml2-devel libtiff-devel php-gd audiofile-devel gtk2-devel subversion nano kernel-devel selinux-policy sqlite-devel openssl-devel yum -y install libtool-ltdl-devel unixODBC unixODBC-devel mysql-connector-odbc rpm -Uvh http://download.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm yum -y install iksemel-devel pear install db yum -y install asterisk-odbc php-pear-DB asterisk-sounds-extra-en-gsm
Находим строчку TTY=9 и комментируем, чтобы получилось #TTY=9
nano /usr/sbin/safe_asterisk
Продолжаем установку
/usr/sbin/safe_asterisk rpm -Uvh http://packages.asterisk.org/centos/6/current/x86_64/RPMS/freepbx-2.11.0beta2-2_centos6.x86_64.rpm amportal chown /etc/dahdi/modules amportal chown /etc/dahdi/system.conf
Теперь можно зайти на сервер http://IP-адрес_вашего-сервера
Возможные ошибки
1. Ошибка при скачивании репозитория с помощью команды wget
-bash: wget: command not found
решение:
yum -y install wget
2. Ошибка при установке asterisk и freepbx (yum -y install freepbx)
Error: Package: freepbx-2.11.0beta2-2_centos6.x86_64 (asterisk-current) Requires: asterisk-core Error: Package: freepbx-2.11.0beta2-2_centos6.x86_64 (asterisk-current) Requires: asterisk-odbc You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest
решение:
нет пакета в репозитории, устанавливайте отдельно или ищите нужный репозиторий
Документация и ссылки по Asterisk и FreePBX для вкуривания:
- http://www.freepbx.org/support/documentation/module-documentation
- http://asteriskonvps.com/category/asterisk/
- http://issues.freepbx.org/secure/Dashboard.jspa
- https://wiki.asterisk.org/wiki/display/AST/Home
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS6%2FRedHatEnterpriseLinux6%29