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

[John Kiniston]

So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead
of PJSIP, SIP, IAX, it's sending a call to a dialplan target.

Your entry in queues.conf with LOCAL/105@internal would send the call to
the context 'internal' extension '105' and execute whatever that dialplan
does.

The parameters I gave are actually part of the Queue member definition,

>From the example queues.conf:

 Each member of this call queue is listed on a separate line in
; the form technology/dialstring.  "member" means a normal member of a
; queue.  An optional penalty may be specified after a comma, such that
; entries with higher penalties are considered last.  An optional member
; name may also be specified after a second comma, which is used in log
; messages as a "friendly name".  Multiple interfaces may share a single
; member name. An optional state interface may be specified after a third
; comma. This interface will be the one for which app_queue receives device
; state notifications, even though the first interface specified is the one
; that is actually called.
;
; A hint can also be used in place of the state interface using the format
; hint:@. If no context is specified then 'default' will
; be used.


So 0 is the Penalty for the user
Then 'eric' is the Member name
and the state interface is using the hint defined for the user.

On Fri, Nov 16, 2018 at 1:58 PM Ivan Demkovitch 
wrote:

> John,
>
> Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and
> configured as per book..
>
> So, from what I understand - LOCAL means I want local extension to be a
> member of a queue.
>
> For example, I have this:
>
> [internal]
>
> ;Eric on extension 105
> exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
> same => n,VoiceMail(105@default,u)
>
> 
>
> Do I understand correctly that I should just put this in queues? That
> would replace 2 members I had (office and cell)
>
> member => LOCAL/105@internal,0,Eric,hint:105@internal
>
>
> Can you direct me to specification of parameters under LOCAL (tried to
> search but don't see any)
> what is 0? What is "Eric"? hint? Wonder what all of them do.
>
> Also, my queues.conf setup like this:
>
> timeout=30
> retry=1
>
> Which means if I send it to "Eric" - it will go to his voicemail after 30
> seconds. Should I change timings?
>
> Thank you!
>
> --
> *From:* John Kiniston 
> *To:* Ivan Demkovitch ; Asterisk Users Mailing
> List - Non-Commercial Discussion 
> *Sent:* Friday, November 16, 2018 2:43 PM
> *Subject:* Re: [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@kiniston-intern,0,John,hint:7124@kiniston-intern
>
> On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch 
> 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.
>
> 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 

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

[Ivan Demkovitch]

Got it working! Thanks a lot again. As a bonus, is there a background on why 
SIP/ did not work with a sip trunk provider? :)



  From: John Kiniston 
 To: Ivan Demkovitch  
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 

 Sent: Friday, November 16, 2018 3:08 PM
 Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some 
reason
   
So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead of 
PJSIP, SIP, IAX, it's sending a call to a dialplan target.

Your entry in queues.conf with LOCAL/105@internal would send the call to the 
context 'internal' extension '105' and execute whatever that dialplan does.

The parameters I gave are actually part of the Queue member definition, 

>From the example queues.conf:

 Each member of this call queue is listed on a separate line in
; the form technology/dialstring.  "member" means a normal member of a
; queue.  An optional penalty may be specified after a comma, such that
; entries with higher penalties are considered last.  An optional member
; name may also be specified after a second comma, which is used in log
; messages as a "friendly name".  Multiple interfaces may share a single
; member name. An optional state interface may be specified after a third
; comma. This interface will be the one for which app_queue receives device
; state notifications, even though the first interface specified is the one
; that is actually called.
;
; A hint can also be used in place of the state interface using the format
; hint:@. If no context is specified then 'default' will
; be used.


So 0 is the Penalty for the user
Then 'eric' is the Member name 
and the state interface is using the hint defined for the user.

On Fri, Nov 16, 2018 at 1:58 PM Ivan Demkovitch  wrote:

John,
Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and 
configured as per book..
So, from what I understand - LOCAL means I want local extension to be a member 
of a queue.
For example, I have this:
[internal]
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
    same => n,VoiceMail(105@default,u)

Do I understand correctly that I should just put this in queues? That would 
replace 2 members I had (office and cell)
member => LOCAL/105@internal,0,Eric,hint:105@internal

Can you direct me to specification of parameters under LOCAL (tried to search 
but don't see any)what is 0? What is "Eric"? hint? Wonder what all of them do.
Also, my queues.conf setup like this:
timeout=30
retry=1
Which means if I send it to "Eric" - it will go to his voicemail after 30 
seconds. Should I change timings?
Thank you!

  From: John Kiniston 
 To: Ivan Demkovitch ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
 Sent: Friday, November 16, 2018 2:43 PM
 Subject: Re: [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@kiniston-intern,0,John,hint:7124@kiniston-intern

On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch  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.
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 

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

[Ivan Demkovitch]

John,
Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and 
configured as per book..
So, from what I understand - LOCAL means I want local extension to be a member 
of a queue.
For example, I have this:
[internal]
;Eric on extension 105
exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
    same => n,VoiceMail(105@default,u)

Do I understand correctly that I should just put this in queues? That would 
replace 2 members I had (office and cell)
member => LOCAL/105@internal,0,Eric,hint:105@internal

Can you direct me to specification of parameters under LOCAL (tried to search 
but don't see any)what is 0? What is "Eric"? hint? Wonder what all of them do.
Also, my queues.conf setup like this:
timeout=30
retry=1
Which means if I send it to "Eric" - it will go to his voicemail after 30 
seconds. Should I change timings?
Thank you!

  From: John Kiniston 
 To: Ivan Demkovitch ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
 Sent: Friday, November 16, 2018 2:43 PM
 Subject: Re: [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@kiniston-intern,0,John,hint:7124@kiniston-intern

On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch  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.
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



   -- 
_
-- 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 

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

[John Kiniston]

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@kiniston-intern,0,John,hint:7124@kiniston-intern

On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch 
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.
>
> 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
>
>
>
>
> --
> _
> -- 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
-- 
_
-- 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

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

[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



   -- 
_
-- 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

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

[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



   -- 
_
-- 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

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

[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

[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

[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'

 



smime.p7s
Description: S/MIME Cryptographic Signature
-- 
_
-- Bandwidth and Colocation Provided by 

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

[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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.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 
 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
    -- 

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

[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

[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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