Ivan Demkovitch
2018-Nov-15 17:26 UTC
[asterisk-users] asterisk-users Digest, Vol 171, Issue 9
Sebastian,
Well, this can't be problem with trunk because:1. Call coming from outside,
so trunk works2. sip show registry shows it registered.
Trunk allows for 2 channels which is not a problem here either
It's just weird that out of 4 queue member only 2 being called and log
doesn't show anything else.
From: "asterisk-users-request at lists.digium.com"
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To: asterisk-users at lists.digium.com
Sent: Thursday, November 15, 2018 11:20 AM
Subject: asterisk-users Digest, Vol 171, Issue 9
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Today's Topics:
1. Queue not dialing out to cell phone for some reason
(Ivan Demkovitch)
2. Re: Queue not dialing out to cell phone for some reason
(Sebastian Nielsen)
3. Re: Queue not dialing out to cell phone for some reason
(Ivan Demkovitch)
4. Re: Queue not dialing out to cell phone for some reason
(Sebastian Nielsen)
----------------------------------------------------------------------
Message: 1
Date: Thu, 15 Nov 2018 16:53:38 +0000 (UTC)
From: Ivan Demkovitch <idemkovitch at yahoo.com>
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] Queue not dialing out to cell phone for some
reason
Message-ID: <897612684.1161831.1542300818435 at mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.I did trace a call and this is what I see. Only 2 phones
(internal) called. External SIP at callcentric is not being called.
Any idea why it's not being called?
-- Executing [1 at automated_attendant_normal:1]
Verbose("SIP/callcentric15-00000435", "1, Caller
"DEMKOVITCH,IVAN" <13144880983> has entered the sales
queue") in new stack
Caller "aa" <15555555555> has entered the sales queue
-- Executing [1 at automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new
stack
-- Goto (queues,7001,1)
-- Executing [7001 at queues:1]
Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001 at queues:2]
BackGround("SIP/callcentric15-00000435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- <SIP/callcentric15-00000435> Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language
'en')
-- Executing [7001 at queues:3]
Queue("SIP/callcentric15-00000435", "sales,,,,85") in new
stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000437 is ringing
-- SIP/FF9EF375CCFC-SLS-00000436 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000439 is ringing
-- SIP/FF9EF375CCFC-SLS-00000438 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0000043b is ringing
-- SIP/FF9EF375CCFC-SLS-0000043a is ringing
-- Stopped music on hold on SIP/callcentric15-00000435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-00000435'
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------------------------------
Message: 2
Date: Thu, 15 Nov 2018 17:58:20 +0100
From: "Sebastian Nielsen" <sebastian at sebbe.eu>
To: "'Ivan Demkovitch'" <idemkovitch at yahoo.com>,
"'Asterisk Users
Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <000501d47d04$698e9480$3cabbd80$@sebbe.eu>
Content-Type: text/plain; charset="utf-8"
I would suspect that the cell phone does use battery saving causing the SIP
application to lose registration with the server. Would also suggest using TCP
with a fairly short keepalive to prevent the cellular network from tearing down
the connection to the asterisk server.
You need to go into android settings and make sure the SIP client is whitelisted
in battery management.
Från: asterisk-users <asterisk-users-bounces at lists.digium.com> För Ivan
Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users at lists.digium.com
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.
I did trace a call and this is what I see. Only 2 phones (internal) called.
External SIP at callcentric is not being called.
Any idea why it's not being called?
-- Executing [1 at automated_attendant_normal:1]
Verbose("SIP/callcentric15-00000435", "1, Caller
"DEMKOVITCH,IVAN" <13144880983> has entered the sales
queue") in new stack
Caller "aa" <15555555555> has entered the sales queue
-- Executing [1 at automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new
stack
-- Goto (queues,7001,1)
-- Executing [7001 at queues:1]
Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001 at queues:2]
BackGround("SIP/callcentric15-00000435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- <SIP/callcentric15-00000435> Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language
'en')
-- Executing [7001 at queues:3]
Queue("SIP/callcentric15-00000435", "sales,,,,85") in new
stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000437 is ringing
-- SIP/FF9EF375CCFC-SLS-00000436 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000439 is ringing
-- SIP/FF9EF375CCFC-SLS-00000438 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0000043b is ringing
-- SIP/FF9EF375CCFC-SLS-0000043a is ringing
-- Stopped music on hold on SIP/callcentric15-00000435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-00000435'
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------------------------------
Message: 3
Date: Thu, 15 Nov 2018 17:00:48 +0000 (UTC)
From: Ivan Demkovitch <idemkovitch at yahoo.com>
To: Sebastian Nielsen <sebastian at sebbe.eu>, 'Asterisk Users
Mailing
List - Non-Commercial Discussion' <asterisk-users at
lists.digium.com>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <1273692324.1141360.1542301248670 at mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
Sebastian,
I don't think it has to do anything with registration. It is dialing through
the SIP trunk, so it goes out as normal cell phone call.Also, why I don't
see anything in a log? I see only first 2 members being dialed.
From: Sebastian Nielsen <sebastian at sebbe.eu>
To: 'Ivan Demkovitch' <idemkovitch at yahoo.com>; 'Asterisk
Users Mailing List - Non-Commercial Discussion' <asterisk-users at
lists.digium.com>
Sent: Thursday, November 15, 2018 10:58 AM
Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some
reason
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div.yiv7898733751WordSection1 {}#yiv7898733751 I would suspect that the cell
phone does use battery saving causing the SIP application to lose registration
with the server. Would also suggest using TCP with a fairly short keepalive to
prevent the cellular network from tearing down the connection to the asterisk
server.You need to go into android settings and make sure the SIP client is
whitelisted in battery management. Från: asterisk-users
<asterisk-users-bounces at lists.digium.com> För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users at lists.digium.com
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason
Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it
should ring all 4 phones at the same time. And it does work but randomly.I did
trace a call and this is what I see. Only 2 phones (internal) called. External
SIP at callcentric is not being called. Any idea why it's not being called?
-- Executing [1 at automated_attendant_normal:1]
Verbose("SIP/callcentric15-00000435", "1, Caller
"DEMKOVITCH,IVAN" <13144880983> has entered the sales
queue") in new stack
Caller "aa" <15555555555> has entered the sales queue
-- Executing [1 at automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new
stack
-- Goto (queues,7001,1)
-- Executing [7001 at queues:1]
Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001 at queues:2]
BackGround("SIP/callcentric15-00000435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- <SIP/callcentric15-00000435> Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language
'en')
-- Executing [7001 at queues:3]
Queue("SIP/callcentric15-00000435", "sales,,,,85") in new
stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000437 is ringing
-- SIP/FF9EF375CCFC-SLS-00000436 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000439 is ringing
-- SIP/FF9EF375CCFC-SLS-00000438 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0000043b is ringing
-- SIP/FF9EF375CCFC-SLS-0000043a is ringing
-- Stopped music on hold on SIP/callcentric15-00000435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-00000435'
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------------------------------
Message: 4
Date: Thu, 15 Nov 2018 18:20:06 +0100
From: "Sebastian Nielsen" <sebastian at sebbe.eu>
To: "'Ivan Demkovitch'" <idemkovitch at yahoo.com>,
"'Asterisk Users
Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <001301d47d07$73aabf40$5b003dc0$@sebbe.eu>
Content-Type: text/plain; charset="utf-8"
Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred
via data connection to the asterisk server.
Seems theres a problem with the trunk then.
What does ”sip show registry” tell you?
(asterisk -r in console and then sip show registry)
It should show a status of ”Registred” to your trunk operator.
Från: Ivan Demkovitch <idemkovitch at yahoo.com>
Skickat: den 15 november 2018 18:01
Till: Sebastian Nielsen <sebastian at sebbe.eu>; 'Asterisk Users
Mailing List - Non-Commercial Discussion' <asterisk-users at
lists.digium.com>
Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some
reason
Sebastian,
I don't think it has to do anything with registration. It is dialing through
the SIP trunk, so it goes out as normal cell phone call.
Also, why I don't see anything in a log? I see only first 2 members being
dialed.
_____
From: Sebastian Nielsen <sebastian at sebbe.eu <mailto:sebastian at
sebbe.eu> >
To: 'Ivan Demkovitch' <idemkovitch at yahoo.com
<mailto:idemkovitch at yahoo.com> >; 'Asterisk Users Mailing List -
Non-Commercial Discussion' <asterisk-users at lists.digium.com
<mailto:asterisk-users at lists.digium.com> >
Sent: Thursday, November 15, 2018 10:58 AM
Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some
reason
I would suspect that the cell phone does use battery saving causing the SIP
application to lose registration with the server. Would also suggest using TCP
with a fairly short keepalive to prevent the cellular network from tearing down
the connection to the asterisk server.
You need to go into android settings and make sure the SIP client is whitelisted
in battery management.
Från: asterisk-users <asterisk-users-bounces at lists.digium.com
<mailto:asterisk-users-bounces at lists.digium.com> > För Ivan
Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users at lists.digium.com <mailto:asterisk-users at
lists.digium.com>
Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4 phones at the same time. And it does
work but randomly.
I did trace a call and this is what I see. Only 2 phones (internal) called.
External SIP at callcentric is not being called.
Any idea why it's not being called?
-- Executing [1 at automated_attendant_normal:1]
Verbose("SIP/callcentric15-00000435", "1, Caller
"DEMKOVITCH,IVAN" <13144880983> has entered the sales
queue") in new stack
Caller "aa" <15555555555> has entered the sales queue
-- Executing [1 at automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new
stack
-- Goto (queues,7001,1)
-- Executing [7001 at queues:1]
Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001 at queues:2]
BackGround("SIP/callcentric15-00000435",
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
-- <SIP/callcentric15-00000435> Playing
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language
'en')
-- Executing [7001 at queues:3]
Queue("SIP/callcentric15-00000435", "sales,,,,85") in new
stack
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000437 is ringing
-- SIP/FF9EF375CCFC-SLS-00000436 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-00000439 is ringing
-- SIP/FF9EF375CCFC-SLS-00000438 is ringing
-- Nobody picked up in 30000 ms
-- Nobody picked up in 30000 ms
-- Stopped music on hold on SIP/callcentric15-00000435
-- Playing periodic announcement
-- <SIP/callcentric15-00000435> Playing
'queue-periodic-announce.ulaw' (language 'en')
-- Started music on hold, class 'default', on channel
'SIP/callcentric15-00000435'
== Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
== Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0000043b is ringing
-- SIP/FF9EF375CCFC-SLS-0000043a is ringing
-- Stopped music on hold on SIP/callcentric15-00000435
== Spawn extension (queues, 7001, 3) exited non-zero on
'SIP/callcentric15-00000435'
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