Прерывается вызов при анонсе в очереди

[dimondack]

Все же
announce-frequency = 20

[spirt]

Поставил я астер 18.8.0. Ситуация не изменилась.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

== Using SIP RTP CoS mark 5
-- Executing [687798@incomming:1] Answer("SIP/trunk-00000005", "") in new stack
-- Executing [687798@incomming:2] Queue("SIP/trunk-00000005", "sales,ct") in new stack
-- Started music on hold, class 'queues-moh', on channel 'SIP/trunk-00000005'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-00000006 connected line has changed. Saving it until answer for SIP/trunk-00000005
-- SIP/901-00000006 is ringing
-- Nobody picked up in 25000 ms
-- Stopped music on hold on SIP/trunk-00000005
-- <SIP/trunk-00000005> Playing 'queue-youarenext.alaw' (language 'ru')
-- Told SIP/trunk-00000005 in sales their queue position (which was 1)
-- <SIP/trunk-00000005> Playing 'silence/1.alaw' (language 'ru')
-- Started music on hold, class 'queues-moh', on channel 'SIP/trunk-00000005'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-00000007 connected line has changed. Saving it until answer for SIP/trunk-00000005
-- SIP/901-00000007 is ringing
-- Nobody picked up in 25000 ms
-- Stopped music on hold on SIP/trunk-00000005
-- <SIP/trunk-00000005> Playing 'queue-youarenext.alaw' (language 'ru')
-- Told SIP/trunk-00000005 in sales their queue position (which was 1)
-- <SIP/trunk-00000005> Playing 'silence/1.alaw' (language 'ru')
-- Started music on hold, class 'queues-moh', on channel 'SIP/trunk-00000005'
== Using SIP RTP CoS mark 5
-- Called SIP/901
-- SIP/901-00000008 connected line has changed. Saving it until answer for SIP/trunk-00000005
-- SIP/901-00000008 is ringing
-- Stopped music on hold on SIP/trunk-00000005
== Spawn extension (incomming, 687798, 2) exited non-zero on 'SIP/trunk-00000005'

Изменения в MOH и частота анонса не влияют на ситуацию.
По логике где-то в queues.conf должна быть опция типа "stop_dialing_on_announcement = no", но что-то она мне никак не попадается

[dimondack]

announce-frequency = 0

[dimondack]

не знаю как вы умудрялись пытаться ответить во время проигрывания анонса
Playing 'queue-youarenext.ulaw' (language 'ru')

Но у меня в это время на софтфонах 1745 и 1746 окошки для ответа/сброса пропадают.
И нет возможности ответить на звонок, в данный короткий интервал времени.


https://disk.yandex.ru/i/5xZG0dGDitfbNw

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

vv*CLI>
-- Accepting AUTHENTICATED call from 192.168.88.254:4569:
-- > requested format = Unknown,
-- > requested prefs = (),
-- > actual format = ulaw,
-- > host prefs = (alaw|ulaw),
-- > priority = mine
-- Executing [999@users:1] Answer("IAX2/1725-14140", "") in new stack
-- Executing [999@users:2] Queue("IAX2/1725-14140", "sales,ct,,,120") in new stack
-- Started music on hold, class 'paul', on channel 'IAX2/1725-14140'
-- Called SIP/1746
-- Called SIP/1745
-- SIP/1745-0000000b connected line has changed. Saving it until answer for IAX2/1725-14140
-- SIP/1746-0000000a connected line has changed. Saving it until answer for IAX2/1725-14140
-- SIP/1746-0000000a is ringing
-- SIP/1745-0000000b is ringing
-- Nobody picked up in 15000 ms
-- Nobody picked up in 15000 ms
-- Stopped music on hold on IAX2/1725-14140
-- <IAX2/1725-14140> Playing 'queue-youarenext.ulaw' (language 'ru')
-- Told IAX2/1725-14140 in sales their queue position (which was 1)
-- <IAX2/1725-14140> Playing 'silence/1.ulaw' (language 'ru')
-- Started music on hold, class 'paul', on channel 'IAX2/1725-14140'
-- Called SIP/1746
-- Called SIP/1745
-- SIP/1745-0000000d connected line has changed. Saving it until answer for IAX2/1725-14140
-- SIP/1746-0000000c connected line has changed. Saving it until answer for IAX2/1725-14140
-- SIP/1746-0000000c is ringing
-- SIP/1745-0000000d is ringing
> 0x809204000 -- Strict RTP learning after remote address set to: 192.168.88.254:4006
-- SIP/1746-0000000c connected line has changed. Saving it until answer for IAX2/1725-14140
-- SIP/1746-0000000c answered IAX2/1725-14140
-- Stopped music on hold on IAX2/1725-14140
-- Channel SIP/1746-0000000c joined 'simple_bridge' basic-bridge <55a0be66-5868-4499-980d-a55eeed603b7>
> 0x809204000 -- Strict RTP switching to RTP target address 192.168.88.254:4006 as source
-- Channel IAX2/1725-14140 joined 'simple_bridge' basic-bridge <55a0be66-5868-4499-980d-a55eeed603b7>
> 0x809204000 -- Strict RTP learning complete - Locking on source address 192.168.88.254:4006
-- Channel IAX2/1725-14140 left 'simple_bridge' basic-bridge <55a0be66-5868-4499-980d-a55eeed603b7>
-- Executing [h@users:1] NoOp("IAX2/1725-14140", "*****SEND to Telegram*************") in new stack
-- Executing [h@users:2] NoOp("IAX2/1725-14140", "***DIALSTATUS=*************") in new stack
-- Executing [h@users:3] GotoIf("IAX2/1725-14140", "0?end") in new stack
-- Executing [h@users:4] System("IAX2/1725-14140", "/usr/local/share/asterisk/agi-bin/scriptFINAL.sh ") in new stack
-- Channel SIP/1746-0000000c left 'simple_bridge' basic-bridge <55a0be66-5868-4499-980d-a55eeed603b7>
-- Executing [h@users:5] Hangup("IAX2/1725-14140", "") in new stack
== Spawn extension (users, h, 5) exited non-zero on 'IAX2/1725-14140'
-- Hungup 'IAX2/1725-14140'
vv*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
vv@vv:/usr/local/etc/asterisk# asterisk -rvvvvvv
Asterisk 18.3.0, Copyright (C) 1999 - 2021, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 18.3.0 currently running on vv (pid = 2251)
vv*CLI>

более того, подключил стационарный телефон через Tau -1.IP с номером 1702
и повторил ..

во время проигрывания Playing 'queue-youarenext.ulaw' (language 'ru')

телефон не звонит и нет возможности ответить на звонок.
специально снимал трубку во время анонса, - тишина....

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

vv*CLI>
-- Accepting AUTHENTICATED call from 192.168.88.254:4569:
-- > requested format = Unknown,
-- > requested prefs = (),
-- > actual format = ulaw,
-- > host prefs = (alaw|ulaw),
-- > priority = mine
-- Executing [999@users:1] Answer("IAX2/1725-7007", "") in new stack
-- Executing [999@users:2] Queue("IAX2/1725-7007", "sales,ct,,,120") in new stack
-- Started music on hold, class 'paul', on channel 'IAX2/1725-7007'
-- Called SIP/1702
-- Called SIP/1746
-- Called SIP/1745
-- SIP/1745-00000003 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1746-00000002 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1746-00000002 is ringing
-- SIP/1702-00000001 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1702-00000001 is ringing
-- SIP/1745-00000003 is ringing
-- Nobody picked up in 15000 ms
-- Nobody picked up in 15000 ms
-- Nobody picked up in 15000 ms
-- Stopped music on hold on IAX2/1725-7007
-- <IAX2/1725-7007> Playing 'queue-youarenext.ulaw' (language 'ru')
> Saved useragent "SIPPER for PhonerLite" for peer 1701
-- Told IAX2/1725-7007 in sales their queue position (which was 1)
-- <IAX2/1725-7007> Playing 'silence/1.ulaw' (language 'ru')
-- Started music on hold, class 'paul', on channel 'IAX2/1725-7007'
-- Called SIP/1702
-- Called SIP/1746
-- Called SIP/1745
-- SIP/1745-00000006 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1746-00000005 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1746-00000005 is ringing
-- SIP/1702-00000004 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1702-00000004 is ringing
-- SIP/1745-00000006 is ringing
> 0x80922f000 -- Strict RTP learning after remote address set to: 192.168.88.130:23020
-- SIP/1702-00000004 connected line has changed. Saving it until answer for IAX2/1725-7007
-- SIP/1702-00000004 answered IAX2/1725-7007
-- Stopped music on hold on IAX2/1725-7007
-- Channel SIP/1702-00000004 joined 'simple_bridge' basic-bridge <dd445786-2551-4ba3-894e-57014fe4b4c3>
-- Channel IAX2/1725-7007 joined 'simple_bridge' basic-bridge <dd445786-2551-4ba3-894e-57014fe4b4c3>
> 0x80922f000 -- Strict RTP switching to RTP target address 192.168.88.130:23020 as source
-- Channel SIP/1702-00000004 left 'simple_bridge' basic-bridge <dd445786-2551-4ba3-894e-57014fe4b4c3>
-- Channel IAX2/1725-7007 left 'simple_bridge' basic-bridge <dd445786-2551-4ba3-894e-57014fe4b4c3>
-- Executing [999@users:3] Hangup("IAX2/1725-7007", "") in new stack
== Spawn extension (users, 999, 3) exited non-zero on 'IAX2/1725-7007'
-- Executing [h@users:1] NoOp("IAX2/1725-7007", "*****SEND to Telegram*************") in new stack
-- Executing [h@users:2] NoOp("IAX2/1725-7007", "***DIALSTATUS=*************") in new stack
-- Executing [h@users:3] GotoIf("IAX2/1725-7007", "0?end") in new stack
-- Executing [h@users:4] System("IAX2/1725-7007", "/usr/local/share/asterisk/agi-bin/scriptFINAL.sh ") in new stack
-- Executing [h@users:5] Hangup("IAX2/1725-7007", "") in new stack
== Spawn extension (users, h, 5) exited non-zero on 'IAX2/1725-7007'
-- Hungup 'IAX2/1725-7007'
vv*CLI>

затем поднял трубку на 1702 после анонса





Может так и не понял ваш вопрос.

[spirt]

телефон не звонит и нет возможности ответить на звонок.
специально снимал трубку во время анонса, - тишина....

Вот в этом, собственно, и проблема, что пока проигрывается анонс, дозвон до участника очереди приостанавливается.
Это крайне неудобно для участника очереди. Допустим не успел он сразу схватить трубку, подходит к телефону, снимает трубку, а там тишина потому что начался анонс... надо ждать когда снова зазвенит.
По моему логично было бы чтобы анонс шел параллельно с вызовом. Не могу поверить, что астер так не умеет...

[ded]

Попробуйте другую стратегию вызова.

[spirt]

Попробовал random и linear.
Ничего не изменилось.

[dimondack]

Тогда ARI

[dimondack]

С этим разберетесь...
https://disk.yandex.ru/d/vmpbcgyHeySx0Q

'000' => 1. Noop(*****___CRM___******) [extensions.conf:387]
2. Answer() [extensions.conf:388]
3. Stasis(dialer) [extensions.conf:389]
4. HangUp() [extensions.conf:390]

Там конечно WEB интерфейс для операторов нужен хороший, но я не готов такое написать пока))
А функционал должен работать

Это мой пилот...

Попробуйте... может Вам поможет

и вообще http://subnets.ru/forum/viewtopic.php?f=13&t=658

[spirt]

ARI это для меня сложно. Нету времени разбираться. Может позже.