asterisk.ru

виртуальный сервер
CENTOS 6.4
ASTERISK 1.8.15
QTECH QVI 2132, QVI2196
1000 внутрених абонентов и SIP транкгруппа на SoftX3000
Раз 1 или 2 недели астер виснет.
при звонке и внутри и снаружи тишина.
в косоли тишина даже при debug 5 и verbose 5.
так как на астере 1000 абонентов то приходиться быстрее перегружать, что б дать связь.
какими способами можно выявить причину зависания?
есть подозрения что зависания вызываються умирающими периодически шлюзами QTECH
остановка была предположительно в период между 13-10 и 13-38


log debug
[Oct 7 13:33:20] DEBUG[17037] taskprocessor.c: destroying taskprocessor 'ast_msg_queue'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9156-0000d921'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9895-0000d92c'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9840-0000d8e0'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9336-0000d92e'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9458-0000d8df'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9160-0000d927'
[Oct 7 13:33:20] DEBUG[16993] channel.c: Didn't get a frame from channel: SIP/9156-0000d921
[Oct 7 13:33:20] DEBUG[16999] channel.c: Didn't get a frame from channel: SIP/9895-0000d92c
[Oct 7 13:33:20] DEBUG[16993] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16999] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d90d'
[Oct 7 13:33:20] DEBUG[16993] channel.c: Bridge stops bridging channels SIP/9156-0000d921 and SIP/ngn-0000d922
[Oct 7 13:33:20] DEBUG[16999] channel.c: Bridge stops bridging channels SIP/9796-0000d92b and SIP/9895-0000d92c
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d908'
[Oct 7 13:33:20] DEBUG[17000] channel.c: Set channel SIP/9336-0000d92e to write format ulaw
[Oct 7 13:33:20] DEBUG[17000] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d922'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d923'
[Oct 7 13:33:20] DEBUG[16958] channel.c: Didn't get a frame from channel: SIP/9458-0000d8df
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d924'
[Oct 7 13:33:20] DEBUG[16958] channel.c: Bridge stops bridging channels SIP/9458-0000d8df and SIP/9840-0000d8e0
[Oct 7 13:33:20] DEBUG[17000] app_macro.c: Spawn extension (macro-dialling_internal,s,9) exited non-zero on 'SIP/9336-0000d92e' in macro 'dialling_internal'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9836-0000d919'
[Oct 7 13:33:20] DEBUG[17000] pbx.c: Spawn extension (phones,9663,5) exited non-zero on 'SIP/9336-0000d92e'
[Oct 7 13:33:20] DEBUG[16997] channel.c: Didn't get a frame from channel: SIP/9160-0000d927
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9042-0000d91b'
[Oct 7 13:33:20] DEBUG[16994] channel.c: Set channel SIP/ngn-0000d923 to write format alaw
[Oct 7 13:33:20] DEBUG[16997] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[17000] channel.c: Soft-Hanging up channel 'SIP/9336-0000d92e'
[Oct 7 13:33:20] DEBUG[16994] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d91d'
[Oct 7 13:33:20] DEBUG[16997] channel.c: Bridge stops bridging channels SIP/9160-0000d927 and SIP/9421-0000d928
[Oct 7 13:33:20] DEBUG[16980] channel.c: Didn't get a frame from channel: SIP/ngn-0000d908
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d8fc'
[Oct 7 13:33:20] DEBUG[16982] channel.c: Didn't get a frame from channel: SIP/ngn-0000d90d
[Oct 7 13:33:20] DEBUG[16980] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16982] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16980] channel.c: Bridge stops bridging channels SIP/9888-0000d907 and SIP/ngn-0000d908
[Oct 7 13:33:20] DEBUG[16982] channel.c: Bridge stops bridging channels SIP/9548-0000d90c and SIP/ngn-0000d90d
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9548-0000d90c'
[Oct 7 13:33:20] DEBUG[16991] channel.c: Didn't get a frame from channel: SIP/ngn-0000d91d
[Oct 7 13:33:20] DEBUG[16991] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9888-0000d907'
[Oct 7 13:33:20] DEBUG[16991] channel.c: Bridge stops bridging channels SIP/9986-0000d91c and SIP/ngn-0000d91d
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9357-0000d913'
[Oct 7 13:33:20] DEBUG[16973] channel.c: Set channel SIP/ngn-0000d8fc to write format alaw
[Oct 7 13:33:20] DEBUG[16973] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9154-0000d912'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d8eb'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9421-0000d928'
[Oct 7 13:33:20] DEBUG[16989] channel.c: Didn't get a frame from channel: SIP/9836-0000d919
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9081-0000d90e'
[Oct 7 13:33:20] DEBUG[16994] app_macro.c: Spawn extension (macro-dialling_internal,s,9) exited non-zero on 'SIP/ngn-0000d923' in macro 'dialling_internal'
[Oct 7 13:33:20] DEBUG[16989] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16989] channel.c: Bridge stops bridging channels SIP/9115-0000d918 and SIP/9836-0000d919
[Oct 7 13:33:20] DEBUG[16974] channel.c: Set channel SIP/9042-0000d91b to write format ulaw
[Oct 7 13:33:20] DEBUG[16985] channel.c: Didn't get a frame from channel: SIP/9357-0000d913
[Oct 7 13:33:20] DEBUG[16974] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Oct 7 13:33:20] DEBUG[16985] channel.c: Bridge stops bridging channels SIP/9154-0000d912 and SIP/9357-0000d913
[Oct 7 13:33:20] DEBUG[16994] pbx.c: Spawn extension (medgor-menu,9521,5) exited non-zero on 'SIP/ngn-0000d923'
[Oct 7 13:33:20] DEBUG[16983] channel.c: Didn't get a frame from channel: SIP/9081-0000d90e
[Oct 7 13:33:20] DEBUG[16974] app_macro.c: Spawn extension (macro-dialling_internal,s,9) exited non-zero on 'SIP/9042-0000d91b' in macro 'dialling_internal'
[Oct 7 13:33:20] DEBUG[16983] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16994] channel.c: Soft-Hanging up channel 'SIP/ngn-0000d923'
[Oct 7 13:33:20] DEBUG[16983] channel.c: Bridge stops bridging channels SIP/9081-0000d90e and SIP/9923-0000d910
[Oct 7 13:33:20] DEBUG[16964] channel.c: Didn't get a frame from channel: SIP/ngn-0000d8eb
[Oct 7 13:33:20] DEBUG[16964] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Oct 7 13:33:20] DEBUG[16964] channel.c: Bridge stops bridging channels SIP/9536-0000d8ea and SIP/ngn-0000d8eb
[Oct 7 13:33:20] DEBUG[16974] pbx.c: Spawn extension (phones,9613,5) exited non-zero on 'SIP/9042-0000d91b'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9923-0000d910'
[Oct 7 13:33:20] DEBUG[16974] channel.c: Soft-Hanging up channel 'SIP/9042-0000d91b'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9802-0000d8e3'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9811-0000d90f'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9796-0000d92b'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9087-0000d8e2'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9811-0000d930'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9942-0000d929'
[Oct 7 13:33:20] DEBUG[16960] channel.c: Didn't get a frame from channel: SIP/9802-0000d8e3
[Oct 7 13:33:20] DEBUG[16960] channel.c: Bridge stops bridging channels SIP/9087-0000d8e2 and SIP/9802-0000d8e3
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9115-0000d918'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9986-0000d91c'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9302-0000d926'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9689-0000d925'
[Oct 7 13:33:20] DEBUG[17037] channel.c: Soft-Hanging up channel 'SIP/9536-0000d8ea'
[Oct 7 13:33:20] DEBUG[16998] channel.c: Set channel SIP/9942-0000d929 to write format ulaw
[Oct 7 13:33:20] DEBUG[16998] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Oct 7 13:33:20] DEBUG[16998] app_macro.c: Spawn extension (macro-dialling_internal,s,9) exited non-zero on 'SIP/9942-0000d929' in macro 'dialling_internal'
[Oct 7 13:33:20] DEBUG[16982] channel.c: Hanging up channel 'SIP/ngn-0000d90d'
[Oct 7 13:33:20] DEBUG[16982] chan_sip.c: Hangup call SIP/ngn-0000d90d, SIP callid 17f4454d34b5c9581964d82002f5c703@10.163.160.58
[Oct 7 13:33:20] DEBUG[16993] channel.c: Hanging up channel 'SIP/ngn-0000d922'
[Oct 7 13:33:20] DEBUG[16993] chan_sip.c: Hangup call SIP/ngn-0000d922, SIP callid 5cbdc63113a0a4bd64d4302d30d704ad@10.163.160.58
[Oct 7 13:33:20] DEBUG[16982] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f21b400ab78'
[Oct 7 13:33:20] DEBUG[16996] channel.c: Didn't get a frame from channel: SIP/9689-0000d925
[Oct 7 13:33:20] DEBUG[16996] channel.c: Bridge stops bridging channels SIP/9689-0000d925 and SIP/9302-0000d926
[Oct 7 13:33:20] DEBUG[16993] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f21ac006a58'
[Oct 7 13:33:20] DEBUG[16998] pbx.c: Spawn extension (phones,9179,5) exited non-zero on 'SIP/9942-0000d929'
[Oct 7 13:33:20] DEBUG[16958] channel.c: Hanging up channel 'SIP/9840-0000d8e0'
[Oct 7 13:33:20] DEBUG[16958] chan_sip.c: Hangup call SIP/9840-0000d8e0, SIP callid 3cadd7b1059e1f754bb0ec6c55279204@10.163.160.58
[Oct 7 13:33:20] DEBUG[16958] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f2198003458'
[Oct 7 13:33:20] DEBUG[17000] channel.c: Hanging up channel 'SIP/9336-0000d92e'
[Oct 7 13:33:20] DEBUG[16998] channel.c: Soft-Hanging up channel 'SIP/9942-0000d929'
[Oct 7 13:33:24] DEBUG[17045] xmldoc.c: Cannot find variable 'description' in tree 'MessageSend'
[Oct 7 13:33:24] DEBUG[17045] xmldoc.c: Cannot find variable 'description' in tree 'AGENT'
[Oct 7 13:33:24] ERROR[17045] netsock2.c: getaddrinfo("vlg-medgorod-pbx01", "(null)", ...): Name or service not known
[Oct 7 13:33:24] DEBUG[17045] xmldoc.c: Cannot find variable 'description' in tree 'SIPPEER'
[Oct 7 13:33:24] DEBUG[17045] xmldoc.c: Cannot find variable 'description' in tree 'SIPCHANINFO'
[Oct 7 13:34:14] DEBUG[17045] taskprocessor.c: destroying taskprocessor 'ast_msg_queue'
[Oct 7 13:34:24] DEBUG[17174] xmldoc.c: Cannot find variable 'description' in tree 'MessageSend'
[Oct 7 13:34:24] DEBUG[17174] xmldoc.c: Cannot find variable 'description' in tree 'AGENT'
[Oct 7 13:34:24] ERROR[17174] netsock2.c: getaddrinfo("vlg-medgorod-pbx01", "(null)", ...): Name or service not known
[Oct 7 13:34:24] DEBUG[17174] xmldoc.c: Cannot find variable 'description' in tree 'SIPPEER'
[Oct 7 13:34:24] DEBUG[17174] xmldoc.c: Cannot find variable 'description' in tree 'SIPCHANINFO'

log messages

[Oct 7 12:11:21] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9292-0000b001' for lack of RTP activity in 61 seconds
[Oct 7 12:14:14] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9601-0000d13c' for lack of RTP activity in 61 seconds
[Oct 7 12:17:14] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9977-0000d183' for lack of RTP activity in 61 seconds
[Oct 7 12:18:13] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9087-0000d1a9' for lack of RTP activity in 61 seconds
[Oct 7 12:20:35] WARNING[16035] chan_sip.c: Purely numeric hostname (3898), and not a peer--rejecting!
[Oct 7 12:20:35] WARNING[16035] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct 7 12:24:17] WARNING[16104] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct 7 12:29:28] NOTICE[50391] chan_sip.c: Unable to create/find SIP channel for this INVITE
[Oct 7 12:29:28] NOTICE[50391] chan_sip.c: Unable to create/find SIP channel for this INVITE
[Oct 7 12:30:20] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9505-0000d325' for lack of RTP activity in 61 seconds
[Oct 7 12:33:40] WARNING[50391] chan_sip.c: Retransmission timeout reached on transmission 404ab0ab11dfcb8f446faa8abf9f2729@10.163.137.49 for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Oct 7 12:34:26] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9042-0000d389' for lack of RTP activity in 61 seconds
[Oct 7 12:35:18] WARNING[50391] chan_sip.c: Retransmission timeout reached on transmission 4d873a25960b0707eba2ec99f29cfb49@10.163.137.52 for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32001ms with no response
[Oct 7 12:47:47] NOTICE[50391] chan_sip.c: Unable to create/find SIP channel for this INVITE
[Oct 7 13:00:21] NOTICE[50391] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:02:56] NOTICE[50391] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:06:30] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9840-0000d74d' for lack of RTP activity in 61 seconds
[Oct 7 13:06:56] WARNING[16774] file.c: Failed to write frame
[Oct 7 13:07:20] WARNING[16783] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct 7 13:07:42] NOTICE[50391] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:08:54] NOTICE[50391] chan_sip.c: Got OK on REFER Notify message
[Oct 7 13:08:59] NOTICE[50391] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:09:18] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9071-0000d7aa' for lack of RTP activity in 61 seconds
[Oct 7 13:12:09] NOTICE[50391] chan_sip.c: Disconnecting call 'SIP/9069-0000d80d' for lack of RTP activity in 61 seconds
[Oct 7 13:14:31] NOTICE[50391] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:15:51] WARNING[16967] app_dial.c: Unable to write frametype: 2
[Oct 7 13:33:24] NOTICE[17045] cdr.c: CDR simple logging enabled.
[Oct 7 13:33:24] NOTICE[17045] loader.c: 142 modules will be loaded.
[Oct 7 13:33:24] NOTICE[17045] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Oct 7 13:33:24] ERROR[17045] netsock2.c: getaddrinfo("vlg-medgorod-pbx01", "(null)", ...): Name or service not known
[Oct 7 13:33:24] WARNING[17045] acl.c: Unable to lookup 'vlg-medgorod-pbx01'
[Oct 7 13:33:25] WARNING[17072] pbx.c: PBX requires Asterisk to be fully booted
[Oct 7 13:33:25] WARNING[17072] chan_sip.c: Failed to start PBX
[Oct 7 13:33:45] NOTICE[17072] chan_sip.c: Peer 'ngn' is now Reachable. (42ms / 100ms)
[Oct 7 13:34:24] NOTICE[17174] cdr.c: CDR simple logging enabled.
[Oct 7 13:34:24] NOTICE[17174] loader.c: 142 modules will be loaded.
[Oct 7 13:34:24] NOTICE[17174] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Oct 7 13:34:24] ERROR[17174] netsock2.c: getaddrinfo("vlg-medgorod-pbx01", "(null)", ...): Name or service not known
[Oct 7 13:34:24] WARNING[17174] acl.c: Unable to lookup 'vlg-medgorod-pbx01'
[Oct 7 13:34:24] NOTICE[17201] chan_sip.c: Call from '9890' (10.163.137.10:5060) to extension '9056' rejected because extension not found in context 'phones'.
[Oct 7 13:34:45] NOTICE[17201] chan_sip.c: Peer 'ngn' is now Reachable. (35ms / 100ms)
[Oct 7 13:35:22] NOTICE[17201] chan_sip.c: SIP Transfer attempted with no appropriate bridged calls to transfer
[Oct 7 13:38:59] WARNING[17353] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)

подскажите как и куда копать?

На вскидку вы действительно имеете траблы с сетью. Наблюдал именно зависания из-за шлюзов - астер поругается на сеть но держится почти неделю, а потом .... просто всех откидывает и наглухо виснет.

Ну а уж насчет виртуалки - тем более

киляйте его с коркой а потом с gdb на ту корку.

Я готов завесить вглухую ваш Астериск при наличии толстого канала ошибочным сценарием - создав петлю вглухую.
Проверяйте сеть и ДНС даже после рестарта. Вот рестарт -
[Oct 7 13:34:24] NOTICE[17174] loader.c: 142 modules will be loaded.
и сразу после этого
[Oct 7 13:34:24] ERROR[17174] netsock2.c: getaddrinfo("vlg-medgorod-pbx01", "(null)", ...): Name or service not known
[Oct 7 13:34:24] WARNING[17174] acl.c: Unable to lookup 'vlg-medgorod-pbx01'

Что у вас с ДНСом там? 8.8.8.8 как обычно?

киляйте его с коркой а потом с gdb на ту корку.

я новичек можно пояснить?
завершить астер с сохранением файла ядра и выложить сам файл или неправильно?

Что у вас с ДНСом там? 8.8.8.8 как обычно?

смотреть SIP.conf?
звонки не ходят как из и наружу так и внутри астера, так может быть при петле?
извиняюсь за вопрос- я новичек.

какими то командами можно проявить симптомы зависания из-за петли или по другой причине?

какой раз убеждаюсь ...
как ? доверяют админить телефонию на 1000 номеров когда знаний по нулям, ну или близко ...

Вот так с ходу никто причину не подскажет, может угадают, а может и нет - мне больше интересно почему она раз в 2 недели виснет, утечка ?

Новичок, при 1000 внутрених абонентов и SIP транкгруппа на SoftX3000?
Когда звонки не ходят как из и наружу так и внутри астера, так может быть при отсутствии ответов на ДНС запросы. Кто у вас на этом сервере ДНС? Написано в /etc/resolv.conf
Прверять в такой момент командой dig например.
Если ответа мгновенно нет, то каррамба, SIP загибается. Это проблема №53
Читать
http://asterisk.ru/news/53

resolv.conf

nameserver 10.163.140.2
nameserver 10.63.158.2

в связи давно, астериск стал читать год назад, сервер с 1000 абонентами встал под работу 2 месяца назад, тот кто начинал проект вынужден был уйти в том числе и из-за проблем запуска астера