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 <johnkinis...@gmail.com>
 To: Ivan Demkovitch <idemkovi...@yahoo.com> 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
 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:<extension>@<context>. 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 <idemkovi...@yahoo.com> 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 <johnkinis...@gmail.com>
 To: Ivan Demkovitch <idemkovi...@yahoo.com>; Asterisk Users Mailing List - 
Non-Commercial Discussion <asterisk-users@lists.digium.com> 
 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 <idemkovi...@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@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@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 <johnkinis...@gmail.com>
 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  '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 <idemkovi...@yahoo.com> 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/15555555555@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@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

   


-- 
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

Reply via email to