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: "[email protected]"
<[email protected]>
To: [email protected]
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 <[email protected]>
To: "[email protected]"
<[email protected]>
Subject: [asterisk-users] Queue not dialing out to cell phone for some
reason
Message-ID: <[email protected]>
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@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@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@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@automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001@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@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" <[email protected]>
To: "'Ivan Demkovitch'" <[email protected]>, "'Asterisk Users
Mailing List - Non-Commercial Discussion'"
<[email protected]>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <[email protected]>
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 <[email protected]> För Ivan
Demkovitch
Skickat: den 15 november 2018 17:55
Till: [email protected]
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@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@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@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@automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001@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@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 <[email protected]>
To: Sebastian Nielsen <[email protected]>, 'Asterisk Users Mailing
List - Non-Commercial Discussion' <[email protected]>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <[email protected]>
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 <[email protected]>
To: 'Ivan Demkovitch' <[email protected]>; 'Asterisk Users Mailing List -
Non-Commercial Discussion' <[email protected]>
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
<[email protected]> För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: [email protected]
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@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@callcentric is not being called. Any idea why it's not being called?
-- Executing [1@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@automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001@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@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" <[email protected]>
To: "'Ivan Demkovitch'" <[email protected]>, "'Asterisk Users
Mailing List - Non-Commercial Discussion'"
<[email protected]>
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for
some reason
Message-ID: <[email protected]>
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 <[email protected]>
Skickat: den 15 november 2018 18:01
Till: Sebastian Nielsen <[email protected]>; 'Asterisk Users Mailing List -
Non-Commercial Discussion' <[email protected]>
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 <[email protected] <mailto:[email protected]> >
To: 'Ivan Demkovitch' <[email protected] <mailto:[email protected]> >;
'Asterisk Users Mailing List - Non-Commercial Discussion'
<[email protected] <mailto:[email protected]> >
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 <[email protected]
<mailto:[email protected]> > För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: [email protected] <mailto:[email protected]>
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@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@callcentric is not being called.
Any idea why it's not being called?
-- Executing [1@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@automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001@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@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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------------------------------
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