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

2018-11-15 Thread Ivan Demkovitch 
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.
155@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/155@callcentric...
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
    same => n,VoiceMail(105@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 
 To: idemkovi...@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  '155' 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  wrote:

John,
This is output of command below. How do I enable and log queue events?The 
1555@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/155@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/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink



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

2018-11-15 Thread Ivan Demkovitch 


  From: John Kiniston 
 To: idemkovi...@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  '155' 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  wrote:

John,
This is output of command below. How do I enable and log queue events?The 
1555@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/155@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/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink



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

2018-11-15 Thread Ivan Demkovitch 
John,
This is output of command below. How do I enable and log queue events?The 
1555@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/155@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 
 To: idemkovi...@yahoo.com; Asterisk Users Mailing List - Non-Commercial 
Discussion  
 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  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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
    -- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
    -- Goto (queues,7001,1)
    -- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
    -- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
    --  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
    -- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0437 is ringing
    -- SIP/FF9EF375CCFC-SLS-0436 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement
    --  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0439 is ringing
    -- SIP/FF9EF375CCFC-SLS-0438 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement


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

2018-11-15 Thread John Kiniston 
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 
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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN"
> <13144880983> has entered the sales queue") in new stack
>   Caller "aa" <155> has entered the sales queue
> -- Executing [1@automated_attendant_normal:2]
> Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
> -- Goto (queues,7001,1)
> -- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435",
> "2,"aa" <155> entering sales queue") in new stack
>   == "aa" <155> entering sales queue
> -- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435",
> "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
> --  Playing
> '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
> -- Executing [7001@queues:3] Queue("SIP/callcentric15-0435",
> "sales85") in new stack
> -- Started music on hold, class 'default', on channel
> 'SIP/callcentric15-0435'
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF9EF375CCFC-SLS
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF4C119EEBF8-SLS
> -- SIP/FF4C119EEBF8-SLS-0437 is ringing
> -- SIP/FF9EF375CCFC-SLS-0436 is ringing
> -- Nobody picked up in 3 ms
> -- Nobody picked up in 3 ms
> -- Stopped music on hold on SIP/callcentric15-0435
> -- Playing periodic announcement
> --  Playing 'queue-periodic-announce.ulaw'
> (language 'en')
> -- Started music on hold, class 'default', on channel
> 'SIP/callcentric15-0435'
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF9EF375CCFC-SLS
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF4C119EEBF8-SLS
> -- SIP/FF4C119EEBF8-SLS-0439 is ringing
> -- SIP/FF9EF375CCFC-SLS-0438 is ringing
> -- Nobody picked up in 3 ms
> -- Nobody picked up in 3 ms
> -- Stopped music on hold on SIP/callcentric15-0435
> -- Playing periodic announcement
> --  Playing 'queue-periodic-announce.ulaw'
> (language 'en')
> -- Started music on hold, class 'default', on channel
> 'SIP/callcentric15-0435'
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF9EF375CCFC-SLS
>   == Using SIP RTP CoS mark 5
> -- Called SIP/FF4C119EEBF8-SLS
> -- SIP/FF4C119EEBF8-SLS-043b is ringing
> -- SIP/FF9EF375CCFC-SLS-043a is ringing
> -- Stopped music on hold on SIP/callcentric15-0435
>   == Spawn extension (queues, 7001, 3) exited non-zero on
> 'SIP/callcentric15-0435'
> --
> _
> -- 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



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

2018-11-15 Thread Sebastian Nielsen 
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  
Skickat: den 15 november 2018 18:01
Till: Sebastian Nielsen ; 'Asterisk Users Mailing List - 
Non-Commercial Discussion' 
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 mailto:sebast...@sebbe.eu> >
To: 'Ivan Demkovitch' mailto:idemkovi...@yahoo.com> >; 
'Asterisk Users Mailing List - Non-Commercial Discussion' 
mailto:asterisk-users@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 mailto:asterisk-users-boun...@lists.digium.com> > För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
--  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
  == Spawn extension (queues, 7001, 3) exited non-zero on 
'SIP/callcentric15-0435'

 



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

2018-11-15 Thread Ivan Demkovitch 
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 
 To: 'Ivan Demkovitch' ; 'Asterisk Users Mailing List - 
Non-Commercial Discussion'  
 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 
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{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 
 För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
    -- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
    -- Goto (queues,7001,1)
    -- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
    -- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
    --  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
    -- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0437 is ringing
    -- SIP/FF9EF375CCFC-SLS-0436 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement
    --  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0439 is ringing
    -- SIP/FF9EF375CCFC-SLS-0438 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement
    --  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-043b is ringing
    -- SIP/

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

2018-11-15 Thread Sebastian Nielsen 
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  För Ivan 
Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users@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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
-- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
-- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
--  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
-- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0437 is ringing
-- SIP/FF9EF375CCFC-SLS-0436 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-0439 is ringing
-- SIP/FF9EF375CCFC-SLS-0438 is ringing
-- Nobody picked up in 3 ms
-- Nobody picked up in 3 ms
-- Stopped music on hold on SIP/callcentric15-0435
-- Playing periodic announcement
--  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
-- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
-- Called SIP/FF4C119EEBF8-SLS
-- SIP/FF4C119EEBF8-SLS-043b is ringing
-- SIP/FF9EF375CCFC-SLS-043a is ringing
-- Stopped music on hold on SIP/callcentric15-0435
  == Spawn extension (queues, 7001, 3) exited non-zero on 
'SIP/callcentric15-0435'



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

2018-11-15 Thread Ivan Demkovitch 
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/1314555@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/callcentric15-0435", "1, Caller "DEMKOVITCH,IVAN" 
<13144880983> has entered the sales queue") in new stack
  Caller "aa" <155> has entered the sales queue
    -- Executing [1@automated_attendant_normal:2] 
Goto("SIP/callcentric15-0435", "queues,7001,1") in new stack
    -- Goto (queues,7001,1)
    -- Executing [7001@queues:1] Verbose("SIP/callcentric15-0435", "2,"aa" 
<155> entering sales queue") in new stack
  == "aa" <155> entering sales queue
    -- Executing [7001@queues:2] BackGround("SIP/callcentric15-0435", 
"/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
    --  Playing 
'/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
    -- Executing [7001@queues:3] Queue("SIP/callcentric15-0435", 
"sales85") in new stack
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0437 is ringing
    -- SIP/FF9EF375CCFC-SLS-0436 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement
    --  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0439 is ringing
    -- SIP/FF9EF375CCFC-SLS-0438 is ringing
    -- Nobody picked up in 3 ms
    -- Nobody picked up in 3 ms
    -- Stopped music on hold on SIP/callcentric15-0435
    -- Playing periodic announcement
    --  Playing 'queue-periodic-announce.ulaw' 
(language 'en')
    -- Started music on hold, class 'default', on channel 
'SIP/callcentric15-0435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-043b is ringing
    -- SIP/FF9EF375CCFC-SLS-043a is ringing
    -- Stopped music on hold on SIP/callcentric15-0435
  == Spawn extension (queues, 7001, 3) exited non-zero on 
'SIP/callcentric15-0435'
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Astricon is coming up October 9-11!  Signup is available at: 
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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