thr3ads.net - asterisk users - [asterisk-users] Queue not dialing out to cell phone for some reason [Nov 2018]

If this information is useful, please help other people find it:
Share via:

Ivan Demkovitch

2018-Nov-15 16:53 UTC

[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'
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/aec5fb8f/attachment.html>

Sebastian Nielsen

2018-Nov-15 16:58 UTC

head link

[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> 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'

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 5261 bytes
Desc: S/MIME Cryptographic Signature
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment.bin>

Ivan Demkovitch

2018-Nov-15 17:00 UTC

head link

[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>
 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

#yiv7898733751 #yiv7898733751 -- _filtered #yiv7898733751
{font-family:Helvetica;panose-1:2 11 6 4 2 2 2 2 2 4;} _filtered #yiv7898733751
{panose-1:2 4 5 3 5 4 6 3 2 4;} _filtered #yiv7898733751
{font-family:Calibri;panose-1:2 15 5 2 2 2 4 3 2 4;}#yiv7898733751
#yiv7898733751 p.yiv7898733751MsoNormal, #yiv7898733751
li.yiv7898733751MsoNormal, #yiv7898733751 div.yiv7898733751MsoNormal
{margin:0cm;margin-bottom:.0001pt;font-size:11.0pt;font-family:sans-serif;}#yiv7898733751
a:link, #yiv7898733751 span.yiv7898733751MsoHyperlink
{color:#0563C1;text-decoration:underline;}#yiv7898733751 a:visited,
#yiv7898733751 span.yiv7898733751MsoHyperlinkFollowed
{color:#954F72;text-decoration:underline;}#yiv7898733751
p.yiv7898733751msonormal0, #yiv7898733751 li.yiv7898733751msonormal0,
#yiv7898733751 div.yiv7898733751msonormal0
{margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv7898733751
span.yiv7898733751E-postmall18 {font-family:sans-serif;}#yiv7898733751
.yiv7898733751MsoChpDefault {font-size:10.0pt;} _filtered #yiv7898733751
{margin:70.85pt 70.85pt 70.85pt 70.85pt;}#yiv7898733751
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'

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/a0b9ed53/attachment.html>

John Kiniston

2018-Nov-15 20:20 UTC

head link

[asterisk-users] Queue not dialing out to cell phone for some reason

what does the output of 'queue show sales' show?

Do you have queue logging enabled? Have you looked in the queue log to see
what events are firing?

On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:
> 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'
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/4cb80e3d/attachment.html>

Ivan Demkovitch

2018-Nov-15 21:08 UTC

head link

[asterisk-users] Queue not dialing out to cell phone for some reason

John,
This is output of command below. How do I enable and log queue events?The 1555
at callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was
1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime,
156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
   Members:
      SIP/15555555555 at callcentric (ringinuse disabled) (Not in use) has taken
4 calls (last was 1219067 secs ago)
      SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
      SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
      SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
   No Callers

      From: John Kiniston <johnkiniston at gmail.com>
 To: idemkovitch at yahoo.com; Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users at lists.digium.com>
 Sent: Thursday, November 15, 2018 2:21 PM
 Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
   
what does the output of 'queue show sales' show?

Do you have queue logging enabled? Have you looked in the queue log to see what
events are firing?

On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:

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


   
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/9b4d41cb/attachment.html>

Ivan Demkovitch

2018-Nov-15 21:37 UTC

head link

[asterisk-users] Queue not dialing out to cell phone for some reason

From: John Kiniston <johnkiniston at gmail.com>
 To: idemkovitch at yahoo.com 
 Sent: Thursday, November 15, 2018 3:17 PM
 Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason

OK.

So it looks like asterisk can't ring FF1565AABB2D-SLS because it's
invalid.

is the user at  '15555555555' actually able the answer calls? I
wouldn't expect that agent to work configured that way, I'd use a LOCAL
channel to direct the call to a context that sets the call up before dialing
out.

You configure queue logging in logger.conf , Look at the settings 
queue_log = yes
queue_log_to_file = yes
queue_log_name = queue_log

On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:

John,
This is output of command below. How do I enable and log queue events?The 1555
at callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was
1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime,
156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
   Members:
      SIP/15555555555 at callcentric (ringinuse disabled) (Not in use) has taken
4 calls (last was 1219067 secs ago)
      SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
      SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
      SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
   No Callers

[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

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/b4c386d4/attachment.html>

Ivan Demkovitch

2018-Nov-15 21:37 UTC

head link

[asterisk-users] Queue not dialing out to cell phone for some reason

John,
FF1565AABB2D-SLS is probably invalid because it's not registered/lost
registration. This client is connected via VPN to our network, it usually works
when it's "warm". Not concerned about it too much.
15555555555 at callcentric OTOH is an actual cell phone that should be dialed
out via callcentric trunk. Maybe I'm smoking something thinking it was
working before. I know it works from
extensions.conf -------------------------[globals]
ERIC_CELL=SIP/15555555555 at callcentric...
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
        same => n,VoiceMail(105 at default,u)
-----------------------------------
but in queues.conf I can't use same globals so I just put it in like
that.What do you mean by using LOCAL channel? Can you be more specific? I'm
not very good at this :)


This is logger.conf. Where(which section) should I place logging configuration?
[general]
dateformat=%F %T
[logfiles]
console => notice,warning,error,dtmf
messages => security,notice,warning,error,fax
verbose => verbose



Thank you!

      From: John Kiniston <johnkiniston at gmail.com>
 To: idemkovitch at yahoo.com 
 Sent: Thursday, November 15, 2018 3:17 PM
 Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
   
OK.

So it looks like asterisk can't ring FF1565AABB2D-SLS because it's
invalid.

is the user at  '15555555555' actually able the answer calls? I
wouldn't expect that agent to work configured that way, I'd use a LOCAL
channel to direct the call to a context that sets the call up before dialing
out.

You configure queue logging in logger.conf , Look at the settings 
queue_log = yes
queue_log_to_file = yes
queue_log_name = queue_log



On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:

John,
This is output of command below. How do I enable and log queue events?The 1555
at callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was
1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime,
156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
   Members:
      SIP/15555555555 at callcentric (ringinuse disabled) (Not in use) has taken
4 calls (last was 1219067 secs ago)
      SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
      SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
      SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls
yet
   No Callers

     

[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



   
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/d5d330d1/attachment.html>

John Kiniston

2018-Nov-16 20:42 UTC

head link

[asterisk-users] Queue not dialing out to cell phone for some reason

My settings for the queue.log are in the [general] section of logger.conf

I'm running 13, I didn't see what version you said you were running.


If I wanted to add a LOCAL channel to my queue I'd do it as

member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern

On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com>
wrote:
> John,
>
> FF1565AABB2D-SLS is probably invalid because it's not registered/lost
> registration. This client is connected via VPN to our network, it usually
> works when it's "warm". Not concerned about it too much.
>
> 15555555555 at callcentric OTOH is an actual cell phone that should be
> dialed out via callcentric trunk.
> Maybe I'm smoking something thinking it was working before. I know it
> works from
>
> extensions.conf
> -------------------------
> [globals]
> ERIC_CELL=SIP/15555555555 at callcentric
> ...
>
> exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
>         same => n,VoiceMail(105 at default,u)
> -----------------------------------
>
> but in queues.conf I can't use same globals so I just put it in like
that.
> What do you mean by using LOCAL channel? Can you be more specific? I'm
not
> very good at this :)
>
>
>
> This is logger.conf. Where(which section) should I place logging
> configuration?
>
> [general]
> dateformat=%F %T
>
> [logfiles]
> console => notice,warning,error,dtmf
> messages => security,notice,warning,error,fax
> verbose => verbose
>
>
>
> Thank you!
>
> ------------------------------
> *From:* John Kiniston <johnkiniston at gmail.com>
> *To:* idemkovitch at yahoo.com
> *Sent:* Thursday, November 15, 2018 3:17 PM
> *Subject:* Re: [asterisk-users] Queue not dialing out to cell phone for
> some reason
>
> OK.
>
> So it looks like asterisk can't ring FF1565AABB2D-SLS because it's
invalid.
>
> is the user at  '15555555555' actually able the answer calls? I
wouldn't
> expect that agent to work configured that way, I'd use a LOCAL channel
to
> direct the call to a context that sets the call up before dialing out.
>
> You configure queue logging in logger.conf , Look at the settings
> queue_log = yes
> queue_log_to_file = yes
> queue_log_name = queue_log
>
>
>
> On Thu, Nov 15, 2018 at 2:08 PM Ivan Demkovitch <idemkovitch at
yahoo.com>
> wrote:
>
> John,
>
> This is output of command below. How do I enable and log queue events?
> The 1555 at callcentric is the one I'm curious about. I just tried
calling
> into "sales" again and it didn't change this "last was
1219067" output
>
> Sales has 0 calls (max unlimited) in 'ringall' strategy (9s
holdtime, 156s
> talktime), W:0, C:4, A:6, SL:0.0% within 0s
>    Members:
>       SIP/15555555555 at callcentric (ringinuse disabled) (Not in use) has
> taken 4 calls (last was 1219067 secs ago)
>       SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no
> calls yet
>       SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no
> calls yet
>       SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no
> calls yet
>    No Callers
>
> ------------------------------
>
>
> [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
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20181116/aa639e05/attachment.html>

Apparently Analagous Threads

Search for more maybe matching threads

asterisk users - Nov 2018 - Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

[asterisk-users] Queue not dialing out to cell phone for some reason

Apparently Analagous Threads