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Прерывается вызов при анонсе в очереди

Проблемы и их решения Asterisk как такового

Модератор: april22

Прерывается вызов при анонсе в очереди

Сообщение spirt » 09 ноя 2021, 03:32

Заметил несколько неожиданное для меня поведение очереди при проигрывании анонса - как только начинается проигрывание анонса в линию (вы являетесь первым в очереди и т.д.) дозвон до участников прекращается... После окончания анонса вызов возобновляется. Т.е. если участник очереди берёт рубку во время анонса, то соединения не происходит.
Воспроизвёл это на тестовом астере со следующими конфигами:
Extensions.conf
[Показать] Спойлер:
[incomming]
exten = _X.,1,Answer()
same = n,Background(tt-weasels)
same = n,Queue(sales,ct);
same = n,Hangup()

Queues.conf
[Показать] Спойлер:
[general]
music = default

[sales]
music = default
wrapuptime=1
autofill = yes
autopause=no
autopausebusy=no
autopausedelay=0
monitor-type = MixMonitor
strategy = ringall
retry=0
ringinuse=no
timeout=15
announce-frequency = 15
queue-youarenext = queue-youarenext ; ("You are now first in line.")
; announce = queue-periodic-announce
queue-thankyou = silence/1
context = from_external
setqueuevar=yes
member = SIP/901
member = SIP/902

В консоли наблюдается следующее:
[Показать] Спойлер:
== Using SIP RTP CoS mark 5
-- Executing [687798@incomming:1] Answer("SIP/trunk-0000000f", "") in new stack
-- Executing [687798@incomming:2] BackGround("SIP/trunk-0000000f", "tt-weasels") in new stack
-- <SIP/trunk-0000000f> Playing 'tt-weasels.alaw' (language 'ru')
-- Executing [687798@incomming:3] Queue("SIP/trunk-0000000f", "sales,ct") in new stack
-- Started music on hold, class 'default', on channel 'SIP/trunk-0000000f'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-00000010 connected line has changed. Saving it until answer for SIP/trunk-0000000f
-- SIP/901-00000010 is ringing
-- Nobody picked up in 15000 ms
-- Stopped music on hold on SIP/trunk-0000000f
-- <SIP/trunk-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
-- Told SIP/trunk-0000000f in sales their queue position (which was 1)
-- <SIP/trunk-0000000f> Playing 'silence/1.alaw' (language 'ru')
-- Started music on hold, class 'default', on channel 'SIP/trunk-0000000f'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-00000011 connected line has changed. Saving it until answer for SIP/trunk-0000000f
-- SIP/901-00000011 is ringing
-- Stopped music on hold on SIP/trunk-0000000f
== Spawn extension (incomming, 687798, 3) exited non-zero on 'SIP/trunk-0000000f'

Перечитал уже кучу инфы по очередям и перепробовал различные опции в queue.conf, но пока всё тщетно. Прошу подсказать как добиться постоянного вызова участника при включенном анонсе.
spirt
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение ded » 09 ноя 2021, 11:32

Судя по конфигу у вас установлен таймаут нахождения позвонившего в очереди = 15 сек.
timeout=15
и в логе видно, что никто за 15 сек трубку не взял
-- Nobody picked up in 15000 ms

Опция 'с' в параметрах вызова очереди
same = n,Queue(sales,ct)
указывает на продолжение выполнения диал-плана, то есть после
-- Nobody picked up in 15000 ms
выполняется Hangup.

Посмотрите ещё кучу информации -
http://asterisk.ru/knowledgebase/Asterisk+cmd+Queue
ded
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение Wapo » 09 ноя 2021, 11:57

ded вы просто не поняли - ТС жалуется то пока очередь проигрывает анонсы на агентов вызовы не идут т.е. пока вы слышите в трубке "вы следующий" аппараты агентов молчат
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Wapo
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение ded » 09 ноя 2021, 12:43

Написал о том, что вижу в логе:
-- SIP/901-00000010 is ringing - агент очереди, оператор на номере 901, звонит его телефон
-- Nobody picked up in 15000 ms он не ответил в течение 15 сек.
-- Stopped music on hold on SIP/trunk-0000000f остановилась музыка в ожидании
-- <SIP/trunk-0000000f> Playing 'queue-youarenext.alaw' (language 'ru') начал воспроизводиться анонс (вы являетесь первым в очереди и т.д.)
-- Told SIP/trunk-0000000f in sales their queue position (which was 1) анонс рассказывает позвонившему клиенту, что он первый в очереди. Но телефонный аппарат у агента 901 молчит, потому что вышел таймаут 15 сек.
ded
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение spirt » 09 ноя 2021, 18:49

Поставил timeout = 60. Теперь вызов длится 60 секунд, потом останавливается, проговаривается анонс и снова начинается вызов.
[Показать] Спойлер:
== Using SIP RTP CoS mark 5
-- Executing [687798@incomming:1] Answer("SIP/trunk-0000001d", "") in new stack
-- Executing [687798@incomming:2] BackGround("SIP/trunk-0000001d", "tt-weasels") in new stack
-- <SIP/trunk-0000001d> Playing 'tt-weasels.alaw' (language 'ru')
-- Executing [687798@incomming:3] Queue("SIP/trunk-0000001d", "sales,ct") in new stack
-- Started music on hold, class 'default', on channel 'SIP/trunk-0000001d'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-0000001e connected line has changed. Saving it until answer for SIP/trunk-0000001d
-- SIP/901-0000001e is ringing
-- Nobody picked up in 60000 ms
-- Stopped music on hold on SIP/trunk-0000001d
-- <SIP/trunk-0000001d> Playing 'queue-youarenext.alaw' (language 'ru')
-- Told SIP/trunk-0000001d in sales their queue position (which was 1)
-- <SIP/trunk-0000001d> Playing 'silence/1.alaw' (language 'ru')
-- Started music on hold, class 'default', on channel 'SIP/trunk-0000001d'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-0000001f connected line has changed. Saving it until answer for SIP/trunk-0000001d
-- SIP/901-0000001f is ringing
-- Stopped music on hold on SIP/trunk-0000001d
== Spawn extension (incomming, 687798, 3) exited non-zero on 'SIP/trunk-0000001d'

Как сделать чтобы анонс начинался через 15 секунд после начала вызова?
spirt
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение ded » 09 ноя 2021, 19:25

Опции -
http://www.asteriskdocs.org/en/3rd_Edit ... ming_id001

А у вас как попадают звонящие в очередь? Через какой транк? ЧТо это за линия в физическом смысле?
ded
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение spirt » 09 ноя 2021, 19:43

SIP линия от Zadarma. Использую в тестовых целях. Вот sip.conf:
[Показать] Спойлер:
[trunk]
host=sip.zadarma.com
insecure=invite
type=friend
fromdomain=sip.zadarma.com
disallow=all
allow=alaw
allow=ulaw
dtmfmode=auto
secret=XXXXXXX
defaultuser=687798
trunkname=687798
fromuser=687798
callbackextension=687798
context=incomming
qualify=400
directmedia=no
nat=force_rport,comedia
spirt
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение dimondack » 09 ноя 2021, 19:57

Тут попробовал с Вашими настройками
вроде решается все со своим musiconhold

[dialplan]
exten => 999,1,Answer()
;same => n,Background(sh_1)
same => n,Queue(sales,ct)
same => n,HangUp()

======================
queues.conf

[general]
music = paul

[sales]
music = paul
wrapuptime=1
autofill = yes
autopause=no
autopausebusy=no
autopausedelay=0
monitor-type = MixMonitor
strategy = ringall
retry=0
ringinuse=no
timeout=15
announce-frequency = 15
queue-youarenext = queue-youarenext ; ("You are now first in line.")
; announce = queue-periodic-announce
queue-thankyou = silence/1
context = users
setqueuevar=yes
member = SIP/1745
member = SIP/1746

====================
musiconhold.conf

[paul]
mode=files
directory=sounds/paul

=======================

https://disk.yandex.ru/d/rmMxg9SO-V9PXw
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Re: Прерывается вызов при анонсе в очереди

Сообщение spirt » 10 ноя 2021, 01:15

Какая у вас версия астера? Интересно было бы увидеть содержимое вашей консоли при звонке.
Попробовал. Закинул aquadelic.alaw в /var/lib/asterisk/moh/queues/. Поднастроил конфиги:
[Показать] Спойлер:
musiconhold.conf

[queues-moh]
mode=files
directory=/var/lib/asterisk/moh/queues
sort=alpha

========================
queues.conf

[general]
music = queues-moh

[sales]
music = queues-moh
wrapuptime = 1
autofill = yes
autopause = no
autopausebusy = no
autopausedelay = 0
monitor-type = MixMonitor
strategy = ringall
retry = 0
ringinuse = no
timeout = 60
announce-frequency = 15
min-announce-frequency = 15
;periodic-announce-frequency = 15
queue-youarenext = queue-youarenext ; ("You are now first in line.")
announce = queue-periodic-announce
queue-thankyou = silence/1
context = incomming
setqueuevar = yes
member = SIP/901
member = SIP/902

Результат прежний.
У меня астер 16.19.1.
spirt
 
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Re: Прерывается вызов при анонсе в очереди

Сообщение dimondack » 10 ноя 2021, 04:25

Asterisk 18.3.0
=================

Тут я уже с параметрами пробовал
Queue(sales,ct,,,120)
announce-frequency = 25

Код: выделить все
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] pbx.c: Executing [999@users:1] Answer("SIP/1748-0000000b", "") in new stack
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] res_rtp_asterisk.c: 0x809393000 -- Strict RTP switching to RTP target address 192.168.88.253:47226 as source
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] pbx.c: Executing [999@users:2] Queue("SIP/1748-0000000b", "sales,ct,,,120") in new stack
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1748-0000000b'
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: Called SIP/1746
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: Called SIP/1745
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: SIP/1745-0000000d connected line has changed. Saving it until answer for SIP/1748-0000000b
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: SIP/1746-0000000c connected line has changed. Saving it until answer for SIP/1748-0000000b
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: SIP/1746-0000000c is ringing
[Nov  9 19:46:51] VERBOSE[100945][C-00000004] app_queue.c: SIP/1745-0000000d is ringing
[Nov  9 19:46:55] VERBOSE[100897][C-00000004] res_rtp_asterisk.c: 0x808ec0000 -- Strict RTP learning after remote address set to: 192.168.88.254:4042
[Nov  9 19:46:55] VERBOSE[100945][C-00000004] app_queue.c: SIP/1746-0000000c connected line has changed. Saving it until answer for SIP/1748-0000000b
[Nov  9 19:46:55] VERBOSE[100945][C-00000004] app_queue.c: SIP/1746-0000000c answered SIP/1748-0000000b
[Nov  9 19:46:55] VERBOSE[100945][C-00000004] res_musiconhold.c: Stopped music on hold on SIP/1748-0000000b
[Nov  9 19:46:55] VERBOSE[100946][C-00000004] bridge_channel.c: Channel SIP/1746-0000000c joined 'simple_bridge' basic-bridge <3b865349-9f8c-4c7b-9b98-b96eec8f7603>
[Nov  9 19:46:55] VERBOSE[100945][C-00000004] bridge_channel.c: Channel SIP/1748-0000000b joined 'simple_bridge' basic-bridge <3b865349-9f8c-4c7b-9b98-b96eec8f7603>
[Nov  9 19:46:55] VERBOSE[100946][C-00000004] res_rtp_asterisk.c: 0x808ec0000 -- Strict RTP switching to RTP target address 192.168.88.254:4042 as source
[Nov  9 19:46:56] VERBOSE[100945][C-00000004] res_rtp_asterisk.c: 0x809393000 -- Strict RTP learning complete - Locking on source address 192.168.88.253:47226
[Nov  9 19:47:00] VERBOSE[100946][C-00000004] res_rtp_asterisk.c: 0x808ec0000 -- Strict RTP learning complete - Locking on source address 192.168.88.254:4042
[Nov  9 19:47:18] VERBOSE[100885] chan_iax2.c: Accepting AUTHENTICATED call from 192.168.88.254:53196:
       > requested format = Unknown,
       > requested prefs = (),
       > actual format = ulaw,
       > host prefs = (alaw|ulaw),
       > priority = mine
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] pbx.c: Executing [999@users:1] Answer("IAX2/1725-11705", "") in new stack
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] pbx.c: Executing [999@users:2] Queue("IAX2/1725-11705", "sales,ct,,,120") in new stack
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] res_musiconhold.c: Started music on hold, class 'paul', on channel 'IAX2/1725-11705'
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] app_queue.c: Called SIP/1745
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] app_queue.c: SIP/1745-0000000e connected line has changed. Saving it until answer for IAX2/1725-11705
[Nov  9 19:47:18] VERBOSE[100952][C-00000005] app_queue.c: SIP/1745-0000000e is ringing
[Nov  9 19:47:19] VERBOSE[100897][C-00000005] res_rtp_asterisk.c: 0x809244000 -- Strict RTP learning after remote address set to: 192.168.88.254:10038
[Nov  9 19:47:19] VERBOSE[100952][C-00000005] app_queue.c: SIP/1745-0000000e connected line has changed. Saving it until answer for IAX2/1725-11705
[Nov  9 19:47:19] VERBOSE[100952][C-00000005] app_queue.c: SIP/1745-0000000e answered IAX2/1725-11705
[Nov  9 19:47:19] VERBOSE[100952][C-00000005] res_musiconhold.c: Stopped music on hold on IAX2/1725-11705
[Nov  9 19:47:19] VERBOSE[100954][C-00000005] bridge_channel.c: Channel SIP/1745-0000000e joined 'simple_bridge' basic-bridge <4925d0e1-046b-4250-9be3-eb73a70af985>
[Nov  9 19:47:19] VERBOSE[100952][C-00000005] bridge_channel.c: Channel IAX2/1725-11705 joined 'simple_bridge' basic-bridge <4925d0e1-046b-4250-9be3-eb73a70af985>
[Nov  9 19:47:19] VERBOSE[100954][C-00000005] res_rtp_asterisk.c: 0x809244000 -- Strict RTP switching to RTP target address 192.168.88.254:10038 as source
[Nov  9 19:47:24] VERBOSE[100954][C-00000005] res_rtp_asterisk.c: 0x809244000 -- Strict RTP learning complete - Locking on source address 192.168.88.254:10038
[Nov  9 19:47:25] VERBOSE[100897][C-00000006] res_rtp_asterisk.c: 0x808d5d000 -- Strict RTP learning after remote address set to: 192.168.88.254:5062
[Nov  9 19:47:25] VERBOSE[100966][C-00000006] pbx.c: Executing [999@users:1] Answer("SIP/1701-0000000f", "") in new stack
[Nov  9 19:47:25] VERBOSE[100966][C-00000006] pbx.c: Executing [999@users:2] Queue("SIP/1701-0000000f", "sales,ct,,,120") in new stack
[Nov  9 19:47:25] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:47:25] VERBOSE[100966][C-00000006] res_rtp_asterisk.c: 0x808d5d000 -- Strict RTP switching to RTP target address 192.168.88.254:5062 as source
[Nov  9 19:47:30] VERBOSE[100966][C-00000006] res_rtp_asterisk.c: 0x808d5d000 -- Strict RTP learning complete - Locking on source address 192.168.88.254:5062
[Nov  9 19:47:30] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:47:30] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
[Nov  9 19:47:35] VERBOSE[100966][C-00000006] app_queue.c: Told SIP/1701-0000000f in sales their queue position (which was 1)
[Nov  9 19:47:35] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'silence/1.alaw' (language 'ru')
[Nov  9 19:47:37] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:47:57] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:47:57] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
[Nov  9 19:48:02] VERBOSE[100966][C-00000006] app_queue.c: Told SIP/1701-0000000f in sales their queue position (which was 1)
[Nov  9 19:48:02] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'silence/1.alaw' (language 'ru')
[Nov  9 19:48:03] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:48:23] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:48:23] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
[Nov  9 19:48:28] VERBOSE[100966][C-00000006] app_queue.c: Told SIP/1701-0000000f in sales their queue position (which was 1)
[Nov  9 19:48:28] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'silence/1.alaw' (language 'ru')
[Nov  9 19:48:29] VERBOSE[100897] chan_sip.c: Saved useragent "MicroSIP/3.20.5" for peer 1745
[Nov  9 19:48:29] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:48:49] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:48:49] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
[Nov  9 19:48:54] VERBOSE[100966][C-00000006] app_queue.c: Told SIP/1701-0000000f in sales their queue position (which was 1)
[Nov  9 19:48:54] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'silence/1.alaw' (language 'ru')
[Nov  9 19:48:55] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:49:15] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:49:15] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'queue-youarenext.alaw' (language 'ru')
[Nov  9 19:49:21] VERBOSE[100966][C-00000006] app_queue.c: Told SIP/1701-0000000f in sales their queue position (which was 1)
[Nov  9 19:49:21] VERBOSE[100966][C-00000006] file.c: <SIP/1701-0000000f> Playing 'silence/1.alaw' (language 'ru')
[Nov  9 19:49:22] VERBOSE[100966][C-00000006] res_musiconhold.c: Started music on hold, class 'paul', on channel 'SIP/1701-0000000f'
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] res_musiconhold.c: Stopped music on hold on SIP/1701-0000000f
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [999@users:3] Hangup("SIP/1701-0000000f", "") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Spawn extension (users, 999, 3) exited non-zero on 'SIP/1701-0000000f'
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [h@users:1] NoOp("SIP/1701-0000000f", "*****SEND to Telegram*************") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [h@users:2] NoOp("SIP/1701-0000000f", "***DIALSTATUS=*************") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [h@users:3] GotoIf("SIP/1701-0000000f", "0?end") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [h@users:4] System("SIP/1701-0000000f", "/usr/local/share/asterisk/agi-bin/scriptFINAL.sh  ") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Executing [h@users:5] Hangup("SIP/1701-0000000f", "") in new stack
[Nov  9 19:49:27] VERBOSE[100966][C-00000006] pbx.c: Spawn extension (users, h, 5) exited non-zero on 'SIP/1701-0000000f'
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] bridge_channel.c: Channel IAX2/1725-11705 left 'simple_bridge' basic-bridge <4925d0e1-046b-4250-9be3-eb73a70af985>
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Executing [h@users:1] NoOp("IAX2/1725-11705", "*****SEND to Telegram*************") in new stack
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Executing [h@users:2] NoOp("IAX2/1725-11705", "***DIALSTATUS=*************") in new stack
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Executing [h@users:3] GotoIf("IAX2/1725-11705", "0?end") in new stack
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Executing [h@users:4] System("IAX2/1725-11705", "/usr/local/share/asterisk/agi-bin/scriptFINAL.sh  ") in new stack
[Nov  9 19:49:38] VERBOSE[100954][C-00000005] bridge_channel.c: Channel SIP/1745-0000000e left 'simple_bridge' basic-bridge <4925d0e1-046b-4250-9be3-eb73a70af985>
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Executing [h@users:5] Hangup("IAX2/1725-11705", "") in new stack
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] pbx.c: Spawn extension (users, h, 5) exited non-zero on 'IAX2/1725-11705'
[Nov  9 19:49:38] VERBOSE[100952][C-00000005] chan_iax2.c: Hungup 'IAX2/1725-11705'
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] bridge_channel.c: Channel SIP/1748-0000000b left 'simple_bridge' basic-bridge <3b865349-9f8c-4c7b-9b98-b96eec8f7603>
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Executing [h@users:1] NoOp("SIP/1748-0000000b", "*****SEND to Telegram*************") in new stack
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Executing [h@users:2] NoOp("SIP/1748-0000000b", "***DIALSTATUS=*************") in new stack
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Executing [h@users:3] GotoIf("SIP/1748-0000000b", "0?end") in new stack
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Executing [h@users:4] System("SIP/1748-0000000b", "/usr/local/share/asterisk/agi-bin/scriptFINAL.sh  ") in new stack
[Nov  9 19:49:42] VERBOSE[100946][C-00000004] bridge_channel.c: Channel SIP/1746-0000000c left 'simple_bridge' basic-bridge <3b865349-9f8c-4c7b-9b98-b96eec8f7603>
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Executing [h@users:5] Hangup("SIP/1748-0000000b", "") in new stack
[Nov  9 19:49:42] VERBOSE[100945][C-00000004] pbx.c: Spawn extension (users, h, 5) exited non-zero on 'SIP/1748-0000000b'
[Nov  9 19:57:14] VERBOSE[100897] chan_sip.c: Saved useragent "SIPPER for PhonerLite" for peer 1701
[Nov  9 19:58:35] VERBOSE[100897] chan_sip.c: Saved useragent "Grandstream Wave 1.0.3.34" for peer 1748
[Nov  9 21:01:15] NOTICE[100897] chan_sip.c: Peer '1701' is now Lagged. (407ms / 300ms)
[Nov  9 21:01:25] NOTICE[100897] chan_sip.c: Peer '1701' is now Reachable. (1ms / 300ms)
Последний раз редактировалось dimondack 10 ноя 2021, 04:40, всего редактировалось 1 раз.
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