Re: [asterisk-users] Queue Timeout

[Nick Olsen]

Hi, To answer your questions.

1. Not sure on statically configured queues. This is a production system
that I can't break down currently to test.
2. Our member table has the following fields, uniqueid, member name, queue
name, interface, penalty, paused. Which are 425, Nick Test Phone,
nickactest, pjsip/ENDPOINTNAME, 1, null respectively.
3. The relevant log bits
1644245159|sip7-1644245037.26849|nickactest|Nick Test
Phone|RINGNOANSWER|12
1644245159|sip7-1644245037.26849|nickactest|NONE|EXITWITHTIMEOUT|1|1|120

I will mention that despite there not being a great way to query the
timeout variable and see what asterisk really loaded from realtime (That I
can find). If I set the timeout to something low, like 15 seconds. It does
seem to respect it. So it appears it's properly reading it. It just appears
there is an upper limit.

Strangely as well. My test queue I used to reproduce it terminates at 120
seconds. The original queue that brought it to my attention terminates at
60 seconds. Despite nothing being set to 120 or 60 seconds respectively.

On Mon, Feb 7, 2022 at 5:08 PM  wrote:

> Does this happen with statically configured queues/queue members in
> queues.conf?
> What is your queue/member config?
> What is in your /var/log/asterisk/queue_log log file?
>
> On 2/7/2022 4:23 PM, Nick Olsen wrote:
> > Hello, We're running asterisk 16 with Realtime.
> >
> > We have queues configured in realtime.
> >
> > The "Timeout" setting appears to have an upper 2 minute limit. Even
> > when setting the timeout in the queue to 600 seconds, the agent is no
> > longer rung after exactly 120 seconds. The asterisk CLI claims
> > "Exiting due to time-out cycle".
> >
> > We are calling the queue with options "tin". Removing "n" does keep
> > the entire queue from exiting. But the agent is still stopped from
> > ringing and then rung again after the announcements fire.
> >
> > I have also tried dynamically passing a timeout (and not) when calling
> > the queue from the dial plan.
> >
> > IE. Queue(queuename,tin,,,600).
> >
> > This does make the queue completely exit after 600 seconds. But does
> > not cause the agent to just ring for 600 seconds straight.
> >
> > We are answering the call in the dial plan first before entering the
> > queue. So this is not an instance of the incoming call being canceled
> > by some underlying carrier for being in the ring state > 60 seconds.
> >
> > Any thoughts?
>
>
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Re: [asterisk-users] Queue don't call Interface PJSIP

[Roberto]

[SOLVED]!!!

My function that changed the callerid was returning an invalid number. 
Although the asterisk sends the call, the SIP header was wrong and the 
extension did not ring


Thanks.


Em 18/08/2020 09:07, Joshua C. Colp escreveu:
On Tue, Aug 18, 2020 at 9:00 AM Roberto 
> wrote:


Hi Joshua, thanks for answer.
In this particular test my extension is on a simple network. There
is no NAT, just an asterisk running on a virtual machine on a 7-
64bit CentOs. I am simulating an environment to be able to use
PJSIP on my client. And even in this small environment, my
extension does not call.

My problem with NAT was with SIP "one way audio" on a client. All
of this testing is to replace SIP with PJSIP on this client. But
as the queue is unable to call a PJSIP extension, the migration
project on the client is stopped.


I tried to separate the debug file, but it seems to me that in
asterisk 17.16.0, there is a problem or I did not know how to
configure it, because the log did not generate it either.
on console:
"pjsip set logger on"
"pjsip set history on"

on file Logger.conf:
debbuger => debug, trace

asterisk -rx "reload"

Make same calls, and opening the file only the following appears:

[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @
asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\


The PJSIP packet logging are verbose messages, if verbose is enabled 
on console or file they will show up there. The history module also 
uses CLI commands to examine the history log.


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 




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Re: [asterisk-users] Queue don't call Interface PJSIP

[Joshua C. Colp]

On Tue, Aug 18, 2020 at 9:00 AM Roberto <
roberto.med...@gasparimsantos.com.br> wrote:

> Hi Joshua, thanks for answer.
> In this particular test my extension is on a simple network. There is no
> NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I
> am simulating an environment to be able to use PJSIP on my client. And even
> in this small environment, my extension does not call.
>
> My problem with NAT was with SIP "one way audio" on a client. All of this
> testing is to replace SIP with PJSIP on this client. But as the queue is
> unable to call a PJSIP extension, the migration project on the client is
> stopped.
>
>
> I tried to separate the debug file, but it seems to me that in asterisk
> 17.16.0, there is a problem or I did not know how to configure it, because
> the log did not generate it either.
> on console:
> "pjsip set logger on"
> "pjsip set history on"
>
> on file Logger.conf:
> debbuger => debug, trace
>
> asterisk -rx "reload"
>
> Make same calls, and opening the file only the following appears:
>
> [2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog
> on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
>

The PJSIP packet logging are verbose messages, if verbose is enabled on
console or file they will show up there. The history module also uses CLI
commands to examine the history log.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Queue don't call Interface PJSIP

[Roberto]

Hi Joshua, thanks for answer.
In this particular test my extension is on a simple network. There is no 
NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. 
I am simulating an environment to be able to use PJSIP on my client. And 
even in this small environment, my extension does not call.


My problem with NAT was with SIP "one way audio" on a client. All of 
this testing is to replace SIP with PJSIP on this client. But as the 
queue is unable to call a PJSIP extension, the migration project on the 
client is stopped.



I tried to separate the debug file, but it seems to me that in asterisk 
17.16.0, there is a problem or I did not know how to configure it, 
because the log did not generate it either.

on console:
"pjsip set logger on"
"pjsip set history on"

on file Logger.conf:
debbuger => debug, trace

asterisk -rx "reload"

Make same calls, and opening the file only the following appears:

[2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ 
asterisk-homolog on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\


Em 17/08/2020 18:57, Joshua C. Colp escreveu:
On Mon, Aug 17, 2020 at 6:16 PM Roberto 
> wrote:


Hello.


I am having a lot of problems with SIP through NAT. So, I decided
to adopt PJSIP. However, I am not able to make the extensions ring
when receiving a call from the queue. I'm using telnet to include
the extension and on the asterisk console, it even shows Called
PJSIP/6001, but the extension doesn't ring. If I call from
extension to extension, it works normally.


Can you describe the actual network setup further? Is the endpoint 
behind NAT or merely Asterisk? I ask because there is no NAT 
configuration for the endpoint, which if it is behind one can be 
problematic. Failing that you'll need to provide a SIP trace using 
"pjsip set logger on" to show the actual SIP traffic flowing (and 
where to).


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 




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Re: [asterisk-users] Queue don't call Interface PJSIP

[Joshua C. Colp]

On Mon, Aug 17, 2020 at 6:16 PM Roberto <
roberto.med...@gasparimsantos.com.br> wrote:

> Hello.
>
>
> I am having a lot of problems with SIP through NAT. So, I decided to adopt
> PJSIP. However, I am not able to make the extensions ring when receiving a
> call from the queue. I'm using telnet to include the extension and on the
> asterisk console, it even shows Called PJSIP/6001, but the extension
> doesn't ring. If I call from extension to extension, it works normally.
>

Can you describe the actual network setup further? Is the endpoint behind
NAT or merely Asterisk? I ask because there is no NAT configuration for the
endpoint, which if it is behind one can be problematic. Failing that you'll
need to provide a SIP trace using "pjsip set logger on" to show the actual
SIP traffic flowing (and where to).

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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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
> 

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, 

Re: [asterisk-users] Queue member not local - PJSIP - Asterisk 16

[Administrator TOOTAI]

No one on this ?

Le 22/11/2018 à 17:59, Administrator TOOTAI a écrit :

Hi all,

I want to set dynamic queue with non local members. I create an 
extension 115 in [localEP] context which is doing the job, eg calls to 
this extension are forwarded to the non local endpoint (which is an IP 
phone connected to an external Asterisk 13 version). Phones are SNOM.


Queue looks like this (all members defines the same one, test purpose):

deblix9*CLI> queue show q301
q301 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s 
talktime), W:0, C:0, A:18, SL:0.0%, SL2:94.4% within 60s

    Members:
   PJSIP/TOOTAI115@TOOTAiAudio (ringinuse disabled) (dynamic) 
(Invalid) has taken no calls yet
   PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has 
taken no calls yet
   Local/115@localEP/n (ringinuse disabled) (dynamic) (Invalid) has 
taken no calls yet

    No Callers

where Local/115 is the working extension I spoke above. The 
PJSIP/TOOTAI115 being the external member. If I display DEVICE_STATE in 
dialplan, I get the INVALID status as shown above.


I also tried to setup an PPermis115 peer in a phone and modify features 
to forward all calls. This doesn't work either getting below about when 
calling the queue:


PJSIP/PPermis115-00d3 connected line has changed. Saving it until 
answer for PJSIP/PPermis102-00d1

     -- Forwarding PJSIP/PPermis102-00d1 to '125' prevented.
[continuously]

Is there a way to force the state of a member or to tell to a queue to 
call a member anyway even if the state is invalid? Other solution?


Thanks for any hint

Regards



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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



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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, 

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
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Astricon is coming up October 9-11!  Signup is available at: 
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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



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_
-- 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

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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



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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



-- 
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
-- 
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-- 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:
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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'

 



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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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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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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> It does seem like a bug.  However, you have a complicated dialplan with a lot 
> of pieces happening at
> once so it may not actually be an Asterisk bug but a problem with your 
> dialplan.  To unravel this is
> going to take some bookkeeping on your part.

Hi Richard,

Thanks for the detailed response. Need to get my head around it a bit!

I’m going to try to set up a test rig with a less complex dialplan.
I’ll then run some tests and will be able to supply sample files if the problem 
persists.

I’m just a little confused what the different is which transfers of inbound 
queued calls and transfers of inbound Dialled calls.
I wonder if it would make a difference if the queue members were 
Local/DialThisEndpoint_200, instead of PJSIP/endpoint_200.

Since the queue dials the members and the member channels inherit the variables 
correctly, that would mean that Local/DialThisEndpoint_200 would inherit the 
DYNAMIC_FEATURES.
The Local channel would dial the endpoint, and when the endpoint performs a 
transfer and loses its variables, the Local channel, as the parent, would still 
have its variables set and the feature codes would still work.

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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Richard Mudgett]

On Wed, Aug 8, 2018 at 7:43 PM, Daniel Journo 
wrote:

> > Doing some more tests, this reads like a bug to me.
> > Using a hanguphandler with DumpChan in the dialplan context that executes
> > the Queue, I can see that DYNAMIC_FEATURES is set.
> > After the attended transfer when the call is ended, the hanguphandler
> still
> > shows that DYNAMIC_FEATURES is set. It's just not accessible.
> >
> > Any thoughts?
> > It likely depens on how you are doing the attended transfer.  Via DTMF?
> Via SIP
> > or channel technology protocol?
> > Does the Agent B channel have the DYNAMIC_FEATURES channel variable set
> > on it?
> >
>
> Thanks for the reply.
>
> To answer your question, the attended transfers are done via the
> endpoint's feature buttons. So I assume it's via SIP requests.
>
> I've been doing some tests and reviewing the debug logs to try to
> understand the problem and still think it's a bug at this point.
>
> Firstly, most of my inbound calls are answered and then Dial() Local
> channels. These Local channels set __DYNAMIC_FEATURES and various other
> things. They are also needed to ensure functionality like MixMonitor can be
> started on the Local channel and then not affected by any transfers. The
> Local channels then either Dial() some peers via other local channels (as
> some peers are required to press 1 to accept the call) or a Local channel
> that dials a Queue().
>
> For non-Queue calls that are going via the Local Channels that only use
> Dial().
> When endpoint_201 dials *1, it is matched with their own channel.
> > DTMF feature hook 0x7f18d803a978 matched DTMF string '*1' on
> 0x7f18c000b080(PJSIP/endpoint_201-cb55)
>
> Interestingly, after the attended transfer from endpoint_201 to
> endpoint_202, when endpoint_202 dials *1, it can no longer match and passes
> it back to the Local channel that originally dialled endpoint_201.
> At that point, it can match the local channel since that's where
> DYNAMIC_FEATURES was originally set.
> > No DTMF feature hooks on 0x7f189c0660b0(PJSIP/endpoint_202-cb5b)
> match '*1'
> > DTMF feature hook 0x7f1894abd408 matched DTMF string '*1' on
> 0x7f186c04efa0(Local/fromfeature_201@phones-5a17;1)
> So although the transfer caused the variable to be lost, the Local
> channel, as the parent, remained and stepped in to complete the *1 request.
> Probably works by accident.
>
> But calls passing through a Local channel that ends in Queue() don't act
> the same way.
>
> The Queue's initial dial of the queuemembers includes the inheritance as
> expected.
> So when endpoint_201 answers and they dial *1, this is the result.
> > DTMF feature hook 0x7f18c4018278 matched DTMF string '*1' on
> 0x7f18b0002ca0(PJSIP/endpoint_201-cacc)
>
> But following a transfer, using the same SIP messaging as the non-queue
> calls, this is the result...
> > DTMF feature string on 0x7f18bd369720(PJSIP/endpoint_202-cae8) is
> now '*1'
> > No DTMF feature hooks on 0x7f18bd369720(PJSIP/endpoint_202-cae8)
> match '*1'
> > Playing DTMF stream '*1' out to 0x7f18e4112980(Local/queue_
> dialplan_101@queue-59b2;2) < this channel still has
> DYNAMIC_FEATURES set (see below) but it just passes the DTMF through?
> > DTMF begin '*' received on Local/queue_dialplan_101@queue-59b2;1
>

Does Local/queue_dialplan_101@queue-59b2;1 have DYNAMIC_FEATURES set on
it?

> DTMF begin passthrough '*' on Local/queue_dialplan_101@queue-59b2;1
> and it's passed all the way back and played to the caller.
>
> This is in spite of the fact that Local/queue_dialplan_101@queue-59b2;2
> has DYNAMIC_FEATURES set earlier in the dialplan.
> > Set("Local/queue_dialplan_101@queue-59b2;2", "__DYNAMIC_FEATURES=
> NewRecordApp")
> And still set at the end of the call, confirmed using DumpChan within the
> channels hangup handler.
>
> > Dumping Info For Channel: Local/queue_dialplan 101@queue-59b2;2:
> > Variables:
> > DYNAMIC_FEATURES=NewRecordApp
>
> I can't really explain why the channel can still have DYNAMIC_FEATURES,
> but it's not perform matching apart from thinking it's a bug.
>
> I hope that wasn't too long winded!
>

It does seem like a bug.  However, you have a complicated dialplan with a
lot of pieces happening at
once so it may not actually be an Asterisk bug but a problem with your
dialplan.  To unravel this is
going to take some bookkeeping on your part.

Here are some pointers to help:

It will help you to see what is going on by drawing out the channel
chains.  For instance I think your initial
non-queue call chain looks like:
PJSIP/caller --> Bridge1 --> L/caller;1 -- L/caller;2 --> Bridge2 -->
L/feature;1 -- L/feature;2 --> Bridge3 --> PJSIP/201

DTMF feature hooks match DTMF coming from a channel toward a bridge.  They
do not match DTMF
going in the other direction.  (This is why Local/queue_
dialplan_101@queue-59b2;2 does not act on
the DTMF above.)

SIP attended transfers involve two calls.  One call is placed on hold while
the endpoint initiates 

Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> Doing some more tests, this reads like a bug to me.
> Using a hanguphandler with DumpChan in the dialplan context that executes
> the Queue, I can see that DYNAMIC_FEATURES is set.
> After the attended transfer when the call is ended, the hanguphandler still
> shows that DYNAMIC_FEATURES is set. It's just not accessible.
>
> Any thoughts?
> It likely depens on how you are doing the attended transfer.  Via DTMF?  Via 
> SIP
> or channel technology protocol?
> Does the Agent B channel have the DYNAMIC_FEATURES channel variable set
> on it?
>

Thanks for the reply.

To answer your question, the attended transfers are done via the endpoint's 
feature buttons. So I assume it's via SIP requests.

I've been doing some tests and reviewing the debug logs to try to understand 
the problem and still think it's a bug at this point.

Firstly, most of my inbound calls are answered and then Dial() Local channels. 
These Local channels set __DYNAMIC_FEATURES and various other things. They are 
also needed to ensure functionality like MixMonitor can be started on the Local 
channel and then not affected by any transfers. The Local channels then either 
Dial() some peers via other local channels (as some peers are required to press 
1 to accept the call) or a Local channel that dials a Queue().

For non-Queue calls that are going via the Local Channels that only use Dial().
When endpoint_201 dials *1, it is matched with their own channel.
> DTMF feature hook 0x7f18d803a978 matched DTMF string '*1' on 
> 0x7f18c000b080(PJSIP/endpoint_201-cb55)

Interestingly, after the attended transfer from endpoint_201 to endpoint_202, 
when endpoint_202 dials *1, it can no longer match and passes it back to the 
Local channel that originally dialled endpoint_201.
At that point, it can match the local channel since that's where 
DYNAMIC_FEATURES was originally set.
> No DTMF feature hooks on 0x7f189c0660b0(PJSIP/endpoint_202-cb5b) match 
> '*1'
> DTMF feature hook 0x7f1894abd408 matched DTMF string '*1' on 
> 0x7f186c04efa0(Local/fromfeature_201@phones-5a17;1)
So although the transfer caused the variable to be lost, the Local channel, as 
the parent, remained and stepped in to complete the *1 request.
Probably works by accident.

But calls passing through a Local channel that ends in Queue() don't act the 
same way.

The Queue's initial dial of the queuemembers includes the inheritance as 
expected.
So when endpoint_201 answers and they dial *1, this is the result.
> DTMF feature hook 0x7f18c4018278 matched DTMF string '*1' on 
> 0x7f18b0002ca0(PJSIP/endpoint_201-cacc)

But following a transfer, using the same SIP messaging as the non-queue calls, 
this is the result...
> DTMF feature string on 0x7f18bd369720(PJSIP/endpoint_202-cae8) is now '*1'
> No DTMF feature hooks on 0x7f18bd369720(PJSIP/endpoint_202-cae8) match 
> '*1'
> Playing DTMF stream '*1' out to 
> 0x7f18e4112980(Local/queue_dialplan_101@queue-59b2;2) < this channel 
> still has DYNAMIC_FEATURES set (see below) but it just passes the DTMF 
> through?
> DTMF begin '*' received on Local/queue_dialplan_101@queue-59b2;1
> DTMF begin passthrough '*' on Local/queue_dialplan_101@queue-59b2;1
and it's passed all the way back and played to the caller.

This is in spite of the fact that Local/queue_dialplan_101@queue-59b2;2 has 
DYNAMIC_FEATURES set earlier in the dialplan.
> Set("Local/queue_dialplan_101@queue-59b2;2", 
> "__DYNAMIC_FEATURES=NewRecordApp")
And still set at the end of the call, confirmed using DumpChan within the 
channels hangup handler.

> Dumping Info For Channel: Local/queue_dialplan 101@queue-59b2;2:
> Variables:
> DYNAMIC_FEATURES=NewRecordApp

I can't really explain why the channel can still have DYNAMIC_FEATURES, but 
it's not perform matching apart from thinking it's a bug.

I hope that wasn't too long winded!

Thanks for the help and time!
Dan
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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Richard Mudgett]

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo 
wrote:

> > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp.
> > AgentA answers and is able to use that feature code.
> > If AgentA performs an attended transfer of a call from a queue to
> AgentB, the
> > feature code no longer works.
> >
> > It only doesn’t work when using Queue() and an Attended transfer is
> > performed.
> >
> > Is this a bug or is there something that needs to be set to allow the
> > DYNAMIC_FEATURES to be inherited after an attended transfer from a queue?
>
> Doing some more tests, this reads like a bug to me.
> Using a hanguphandler with DumpChan in the dialplan context that executes
> the Queue, I can see that DYNAMIC_FEATURES is set.
> After the attended transfer when the call is ended, the hanguphandler
> still shows that DYNAMIC_FEATURES is set. It's just not accessible.
>
> Any thoughts?
>
It likely depens on how you are doing the attended transfer.  Via DTMF?
Via SIP or channel technology protocol?
Does the Agent B channel have the DYNAMIC_FEATURES channel variable set on
it?

Richard
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Re: [asterisk-users] Queue breaks Dynamic_Features on Attended Transfer

[Daniel Journo]

> Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp.
> AgentA answers and is able to use that feature code.
> If AgentA performs an attended transfer of a call from a queue to AgentB, the
> feature code no longer works.
>
> It only doesn't work when using Queue() and an Attended transfer is
> performed.
>
> Is this a bug or is there something that needs to be set to allow the
> DYNAMIC_FEATURES to be inherited after an attended transfer from a queue?

Doing some more tests, this reads like a bug to me.
Using a hanguphandler with DumpChan in the dialplan context that executes the 
Queue, I can see that DYNAMIC_FEATURES is set.
After the attended transfer when the call is ended, the hanguphandler still 
shows that DYNAMIC_FEATURES is set. It's just not accessible.

Any thoughts?
Thanks
Dan
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Re: [asterisk-users] queue peridiodic-announce-frequency

[Paul Neuwirth]

On Wed, 17 Jan 2018 12:08:40 +0100
Antony Stone  wrote:

> On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote:
> 
> > Hello group,
> > 
> > I tried a lot to enlarge the frequency (i.e. more announces, low
> > wait between). according to config, every 30 seconds the
> > announcement should take place. In fact, the first periodic
> > announce is done after 2 minutes?
> > What is my fault?  
> 
> Config snipped for clarity...
> 
> > [defaultq]
> > timeout = 10
> > retry = 99
> > wrapuptime=15
> > maxlen = 0
> > announce-frequency = 30
> > min-announce-frequency = 30
> > periodic-announce-frequency = 30
> > announce-holdtime = yes
> > announce-position = yes
> > announce-to-first-user = yes
> > periodic-announce = tt-allbusy,hold-or-dial-0
> > reportholdtime = yes  
> 
> I believe that both timeout and retry have to be less than the
> announce intervals.
> 
> In your case, retry is set to 99, with the announce intervals set at
> 30, therefore you would get announcements at 30, 60, 90, 120, 150...
> except that retry=99 stops any announcement before 99, therefore the
> first one you get is at 120 (2 minutes).
> 
> So, try setting retry to a value not less than 30 and see if that
> fixes things.
> 
> 
> 
> Antony.
> 

thank you. working fine now. I misinterpret the retry (thought counts).
Unfortunately the documentation (comments) lack dimensions almost at
all..

Regards

Paul

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Re: [asterisk-users] queue peridiodic-announce-frequency

[Antony Stone]

On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote:

> Hello group,
> 
> I tried a lot to enlarge the frequency (i.e. more announces, low wait
> between). according to config, every 30 seconds the announcement should
> take place. In fact, the first periodic announce is done after 2
> minutes?
> What is my fault?

Config snipped for clarity...

> [defaultq]
> timeout = 10
> retry = 99
> wrapuptime=15
> maxlen = 0
> announce-frequency = 30
> min-announce-frequency = 30
> periodic-announce-frequency = 30
> announce-holdtime = yes
> announce-position = yes
> announce-to-first-user = yes
> periodic-announce = tt-allbusy,hold-or-dial-0
> reportholdtime = yes

I believe that both timeout and retry have to be less than the announce 
intervals.

In your case, retry is set to 99, with the announce intervals set at 30, 
therefore you would get announcements at 30, 60, 90, 120, 150... except that 
retry=99 stops any announcement before 99, therefore the first one you get is 
at 120 (2 minutes).

So, try setting retry to a value not less than 30 and see if that fixes things.



Antony.

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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

[Richard Mudgett]

On Sat, Sep 10, 2016 at 5:18 AM, Jonas Kellens 
wrote:

> On 10-09-16 09:42, Jonas Kellens wrote:
>
>
> On 10-09-16 00:50, Richard Mudgett wrote:
>
>
>
> On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
> wrote:
>
>> Hello
>>
>> when I type on the Asterisk CLi 'queue show', I first get a list of my
>> queues and then the following :
>>
>>
>> failed to extend from 240 to 327
>>
>
> 
>
> failed to extend from 240 to 334
>>
>>
>> I could not find any information on this on the web, except this :
>> https://issues.asterisk.org/jira/browse/ASTERISK-8828
>>
>> which is an old 'bug' that should have been fixed meanwhile.
>>
>> Any more thoughts on why I should be getting this message when asking
>> information about queues (I don't see this message on any other command).
>>
>
> That message is a result of trying to build a string where the buffer is
> too
> small to contain it.  I would expect that there is a truncated string in
> the
> 'queue show' output.
>
> You haven't stated which Asterisk version you are using.  It may already
> be fixed.
>
>
> Hello
>
> I have this with asterisk-certified-13.8-cert1 and also with
> asterisk-certified-13.8-cert2
>
> Could it be that the membername value (and interface value) in my realtime
> MySQL table queue_members is too long ??
>
> It looks like this :
>
> Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue
> Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue
> Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue
>
> I have the idea that this is the "problem".
>
> FYI : it also makes that Asterisk restarts (with core dump) whenever a
> queue is addressed. Very *annoying* !
>
>
> So string size too large and buffer too small.
>
> FYI : I do not have this with any version of Asterisk 1.8. This is a
> "problem" that exists only in Asterisk 13.
>
>
>
> How to fix this ??
>
>
> This is an example output for queue show <> on Asterisk version
> asterisk-certified-13.8-cert2 (same on asterisk-certified-13.8-cert1) :
>
>
> sip*CLI> queue show cvikbubohirndceiaetsq
> cvikbubohirndceiaetsq has 0 calls (max unlimited) in 'ringall' strategy
> (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
>Members:
>   cvikbubohirndceiaets012 
> (Local/cvikbubohirndceiaets012@ExternalCallFromQueue
> from Local/cvikbubohirndceiaets012@ExternalCallFromQueue) (ringinuse
> disabled) (realtime) (Not in use) has taken no
>   cvikbubohirndceiaets248 
> (Local/cvikbubohirndceiaets248@ExternalCallFromQueue
> from Local/cvikbubohirndceiaets248@ExternalCallFromQueue) (ringinuse
> disabled) (realtime) (Not in use) has taken no
>   cvikbubohirndceiaets428 
> (Local/cvikbubohirndceiaets428@ExternalCallFromQueue
> from Local/cvikbubohirndceiaets428@ExternalCallFromQueue) (ringinuse
> disabled) (realtime) (Not in use) has taken no
>   cvikbubohirndceiaets461 
> (Local/cvikbubohirndceiaets461@ExternalCallFromQueue
> from Local/cvikbubohirndceiaets461@ExternalCallFromQueue) (ringinuse
> disabled) (realtime) (Not in use) has taken no
>   cvikbubohirndceiaets629 
> (Local/cvikbubohirndceiaets629@ExternalCallFromQueue
> from Local/cvikbubohirndceiaets629@ExternalCallFromQueue) (ringinuse
> disabled) (realtime) (Not in use) has taken no
>No Callers
>
> failed to extend from 240 to 327
> failed to extend from 240 to 327
> failed to extend from 240 to 327
> failed to extend from 240 to 327
> failed to extend from 240 to 327
>

I have created https://issues.asterisk.org/jira/browse/ASTERISK-26360 to
keep track of the
issue.  I have also put a patch up on gerrit to increase the size of the
buffer.  Since it is
unlikely that you have a Digium SLA [1] concerning certified Asterisk, the
fix will not make its
way into certified asterisk.

Richard

[1] https://www.digium.com/products/asterisk/certified-asterisk
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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

[Jonas Kellens]

On 10-09-16 09:42, Jonas Kellens wrote:


On 10-09-16 00:50, Richard Mudgett wrote:



On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
> wrote:


Hello

when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and then the following :


failed to extend from 240 to 327




failed to extend from 240 to 334


I could not find any information on this on the web, except this
: https://issues.asterisk.org/jira/browse/ASTERISK-8828


which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when
asking information about queues (I don't see this message on any
other command).


That message is a result of trying to build a string where the buffer 
is too
small to contain it.  I would expect that there is a truncated string 
in the

'queue show' output.

You haven't stated which Asterisk version you are using.  It may 
already be fixed.


Hello

I have this with asterisk-certified-13.8-cert1 and also with 
asterisk-certified-13.8-cert2


Could it be that the membername value (and interface value) in my 
realtime MySQL table queue_members is too long ??


It looks like this :

Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue
Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue
Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue

I have the idea that this is the "problem".

FYI : it also makes that Asterisk restarts (with core dump) whenever a 
queue is addressed. Very /annoying/ !



So string size too large and buffer too small.

FYI : I do not have this with any version of Asterisk 1.8. This is a 
"problem" that exists only in Asterisk 13.




How to fix this ??



This is an example output for queue show <> on Asterisk version 
asterisk-certified-13.8-cert2 (same on asterisk-certified-13.8-cert1) :



sip*CLI> queue show cvikbubohirndceiaetsq
cvikbubohirndceiaetsq has 0 calls (max unlimited) in 'ringall' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s

   Members:
  cvikbubohirndceiaets012 
(Local/cvikbubohirndceiaets012@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets012@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets248 
(Local/cvikbubohirndceiaets248@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets248@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets428 
(Local/cvikbubohirndceiaets428@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets428@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets461 
(Local/cvikbubohirndceiaets461@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets461@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets629 
(Local/cvikbubohirndceiaets629@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets629@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no

   No Callers

failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327



Any idea on how to fix this ??


Kind regards.

J.

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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

[Jonas Kellens]

On 10-09-16 00:50, Richard Mudgett wrote:



On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
> wrote:


Hello

when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and then the following :


failed to extend from 240 to 327




failed to extend from 240 to 334


I could not find any information on this on the web, except this :
https://issues.asterisk.org/jira/browse/ASTERISK-8828


which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when
asking information about queues (I don't see this message on any
other command).


That message is a result of trying to build a string where the buffer 
is too
small to contain it.  I would expect that there is a truncated string 
in the

'queue show' output.

You haven't stated which Asterisk version you are using.  It may 
already be fixed.


Hello

I have this with asterisk-certified-13.8-cert1 and also with 
asterisk-certified-13.8-cert2


Could it be that the membername value (and interface value) in my 
realtime MySQL table queue_members is too long ??


It looks like this :

Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue
Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue
Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue

I have the idea that this is the "problem".

FYI : it also makes that Asterisk restarts (with core dump) whenever a 
queue is addressed. Very /annoying/ !



So string size too large and buffer too small.

FYI : I do not have this with any version of Asterisk 1.8. This is a 
"problem" that exists only in Asterisk 13.




How to fix this ??


Kind regards.

Jonas.

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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

[Richard Mudgett]

On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
wrote:

> Hello
>
> when I type on the Asterisk CLi 'queue show', I first get a list of my
> queues and then the following :
>
>
> failed to extend from 240 to 327
>



failed to extend from 240 to 334
>
>
> I could not find any information on this on the web, except this :
> https://issues.asterisk.org/jira/browse/ASTERISK-8828
>
> which is an old 'bug' that should have been fixed meanwhile.
>
> Any more thoughts on why I should be getting this message when asking
> information about queues (I don't see this message on any other command).
>

That message is a result of trying to build a string where the buffer is too
small to contain it.  I would expect that there is a truncated string in the
'queue show' output.

You haven't stated which Asterisk version you are using.  It may already be
fixed.

Richard
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Re: [asterisk-users] Queue grouping - how can it be implemented?

[John Kiniston]

Use Local Channels and hints to combine SIP/MOM and SIP/MOMMobile into a
single extension you add to the queue.


extensions.conf:

[queue-phones]
exten => MOM,1,Dial(SIP/MOM/MOMMOBILE,60,tkw)
exten => MOM,hint,SIP/MOM/MOMMOBILE

exten => DAD,1,Dial(SIP/DAD/DADMOBILE,60,tkw)
exten => DAD,hint,SIP/DAD/DADMOBILE
queues.conf:
[myqueuequeue]
member => LOCAL/MOM@queue-phones,0,MOM,hint:MOM@queue-phones
member => LOCAL/DAD@queue-phones,0,Dad,hint:DAD@queue-phones



On Wed, Jun 15, 2016 at 1:27 AM, Sebastian Nielsen 
wrote:

> I have a Asterisk set up. In this, I want to use queues.
>
>
>
> Now I want to group ”agents” into groups, such as so if one phone in a
> group is busy, the whole group is considered busy.
>
>
>
> Eg:
>
> Group1:
>
> SIP/Dad
>
> SIP/DadsMobile
>
>
>
> Group2:
>
> SIP/Mom
>
> SIP/MomsMobile
>
>
>
>
>
> If there is three persons in queue, then, then, first, all 4 phones should
> ring. Now lets say Mom takes the call via the Mobile.
>
> Now, for the next call in queue, only Dad and DadsMobile should ring. He
> picks up the call via the home phone.
>
>
>
> Now, even if SIP/Mom and SIP/DadsMobile is vacant, both groups should be
> considered busy, and the third person in queue, has to wait in queue until
> either SIP/MomsMobile or SIP/Dad is complete with the call.
>
>
>
> How can this be implemented? Can it be implemented with the standard Queue
> application through advanced dialplan programming or does it need something
> completely custom?
>
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Re: [asterisk-users] Queue logfile txt format in mySQL needed

[Kevin Larsen]

> From: Thomas 
> To: asterisk-users@lists.digium.com, 
> Date: 01/21/2016 04:17 AM
> Subject: [asterisk-users] Queue logfile txt format in mySQL needed
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hello,
> 
> Iam using queues and agents, thats OK.
> 
> I have interesting information form Asterisk in txt file format
> var/log/asterisk/queue_log
> 
> Today Iam reading these txt files and wrote them in an mySQL databases.
> 
> I would need this information more realtime. Some information I do 
writing in 
> the dialplan direct in an mySQL database.
> 
> Is there any way that Asterisk write this information direct in an mySQL 

> database instead of using var/log/asterisk/queue_log?

I haven't done this myself, but it looks like you just need to set up the 
appropriate database connections. See here for a semi-recent example:
http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

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Re: [asterisk-users] Queue logfile txt format in mySQL needed

[Thomas]

Am Donnerstag, 21. Januar 2016, 09:52:53 schrieben Sie:
> > From: Thomas 
> > To: asterisk-users@lists.digium.com,
> > Date: 01/21/2016 04:17 AM
> > Subject: [asterisk-users] Queue logfile txt format in mySQL needed
> > Sent by: asterisk-users-boun...@lists.digium.com
> > 

> > database instead of using var/log/asterisk/queue_log?
> 
> I haven't done this myself, but it looks like you just need to set up the
> appropriate database connections. See here for a semi-recent example:
> http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

Hi,
thanks...


in /etc/asterisk/res_config_mysql.c
for example define in [mydatabase] acsess to your database

in /etc/asterisk/extconfig.conf

queue_log => mysql,mydatabase,queue_log


CREATE TABLE IF NOT EXISTS `queue_log` (
`id` int(10) unsigned NOT NULL auto_increment,
`time` varchar(40) default NULL,
`callid` varchar(80) NOT NULL default '',
`queuename` varchar(80) NOT NULL default '',
`agent` varchar(80) NOT NULL default '',
`event` varchar(32) NOT NULL default '',
`data` varchar(255) NOT NULL default '',
`data1` varchar(255) NOT NULL default '',
`data2` varchar(255) NOT NULL default '',
`data3` varchar(255) NOT NULL default '',
`data4` varchar(255) NOT NULL default '',
`data5` varchar(255) NOT NULL default '',
PRIMARY KEY (`id`)
);


after make an reload of asterisk info will be written in this table and not 
any more in the txt file



I found some explantion for the data written in the fields:
https://wiki.asterisk.org/wiki/display/AST/Queue+Logs




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Re: [asterisk-users] queue periodic-announce without stopping ringing

[Matthew Jordan]

On Mon, Jun 15, 2015 at 9:22 AM, Marek Cervenka cerv...@fpf.slu.cz wrote:
 hello,

 is it possible to play queue periodic-announce without stopping agents
 ringing? actual situation is sequential - ring agents, play announce (for 15
 sec), ring agents , ... (i need to connect agent with caller asap when agent
 is free)

 is it possible with ARI?


ARI does not change or otherwise allow for the manipulation of the
mechanics in app_queue. If you want to use app_queue, ARI is not the
API for you.

If you are looking to write your own call queue application, than ARI
has the ability to manipulate media on a channel, as well as whether
or not ringing is being indicated to the channel. Since you want to
ring an agent, play media to the agent, then ring the agent again, you
will most likely need to indicate ringing to the agent using inband
ringing (via a 'tone' media URI [1]).

For more information on ARI and its intended use, see [2].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-play
[2] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573

Matt

-- 
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Queue PJSIP, not all contacts rings

[Joshua Colp]

Nick Awesome wrote:

Hay guys, have question.

When I do regular dial I use
$this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);

 to get all contacts of current endpoint and so I dial to all phones
at once,

but if I exec QUEUE, I have just one phone rings, seems like it take
first one as Dial app by default, is there way to fix this?


There is no way to directly do this. The best option is to use a Local 
channel into the dialplan which dials instead. Once answered everything 
should fall into place.


--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] Queue PJSIP, not all contacts rings

[Nick Awesome]

Works, thank you!

 On Feb 23, 2015, at 7:11 PM, Joshua Colp jc...@digium.com wrote:
 
 Nick Awesome wrote:
 Hay guys, have question.
 
 When I do regular dial I use
 $this-AGI-get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
 
 to get all contacts of current endpoint and so I dial to all phones
 at once,
 
 but if I exec QUEUE, I have just one phone rings, seems like it take
 first one as Dial app by default, is there way to fix this?
 
 There is no way to directly do this. The best option is to use a Local 
 channel into the dialplan which dials instead. Once answered everything 
 should fall into place.
 
 -- 
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

[Paul Belanger]

On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
 On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi

 We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
 queues.

 For a particular customer, when I run queue show queue_name I get the
 following numbers:

 queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s
 holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s

 So from that data we look at
 17s holdtime
 And assume that is the average hold time before calls get answered by a
 queue members.

 However, if I calculate the average hold time from out queue log table using
 the following SQL

 select sum(data1)/ count(*) as ave_hold_time from queue_log where time 
 DATE(NOW()) and queuename='queue_name' and event='CONNECT';

 I get the vastly different figure of 92.4.

 So, is the queue show figure wrong due to a bug or am I making an incorrect
 assumption as to what it means?

 Thanks in advance

 Welcome to business logic embedded into app_queue.  The issue with the
 queue show command rendering stats, is what timeframe are the stats
 aggregated over?  IIRC, the calculations are using a moving
 average[1].

Opps, sent instead of pasting.

Either way, your likely better off rendering the data using the raw
sql info vs depending on CLI output.  That's what we've done.

[1] http://en.wikipedia.org/wiki/Moving_average
-- 
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

[Paul Belanger]

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi

 We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
 queues.

 For a particular customer, when I run queue show queue_name I get the
 following numbers:

 queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s
 holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s

 So from that data we look at
 17s holdtime
 And assume that is the average hold time before calls get answered by a
 queue members.

 However, if I calculate the average hold time from out queue log table using
 the following SQL

 select sum(data1)/ count(*) as ave_hold_time from queue_log where time 
 DATE(NOW()) and queuename='queue_name' and event='CONNECT';

 I get the vastly different figure of 92.4.

 So, is the queue show figure wrong due to a bug or am I making an incorrect
 assumption as to what it means?

 Thanks in advance

Welcome to business logic embedded into app_queue.  The issue with the
queue show command rendering stats, is what timeframe are the stats
aggregated over?  IIRC, the calculations are using a moving
average[1].




-- 
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Re: [asterisk-users] queue reload command

[Ishfaq Malik]

That's what I would have guessed but it's not working:

[ish@??? ~]$ asterisk -rx 'queue show axon-all'
axon-all has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:2, SL:0.0% within 20s
   Members:
  AXON200 (realtime) (Not in use) has taken no calls yet
  AXON201 (realtime) (Not in use) has taken no calls yet
  AXON202 (realtime) (Not in use) has taken no calls yet
  AXON203 (realtime) (Not in use) has taken no calls yet
  AXON204 (realtime) (In use) has taken no calls yet
  AXON205 (realtime) (Not in use) has taken no calls yet
  AXON206 (realtime) (Not in use) has taken no calls yet
  AXON207 (realtime) (Not in use) has taken no calls yet
  AXON208 (realtime) (Unavailable) has taken no calls yet
  AXON209 (realtime) (Not in use) has taken no calls yet
  AXON210 (realtime) (Unavailable) has taken no calls yet
  AXON211 (realtime) (Unavailable) has taken no calls yet
  AXON214 (realtime) (Not in use) has taken no calls yet
  AXON221 (realtime) (Not in use) has taken no calls yet
  AXON222 (realtime) (Not in use) has taken no calls yet
  AXON223 (realtime) (Unavailable) has taken no calls yet
  AXON225 (realtime) (Not in use) has taken no calls yet
  AXON231 (realtime) (Unavailable) has taken no calls yet
  AXON232 (realtime) (Not in use) has taken no calls yet
  AXON233 (realtime) (Not in use) has taken no calls yet
   No Callers

[ish@??? ~]$ asterisk -rx 'queue reload axon-all'
No such command 'queue reload axon-all' (type 'core show help queue reload
axon-all' for other possible commands)


On 8 January 2015 at 14:23, Andrew Colin and...@convergedgroup.net wrote:

 Hi



 queue reload(queue name) or queue reload all



 for example



 queue reload reception



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ishfaq Malik
 *Sent:* Thursday, January 8, 2015 2:10 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] queue reload command



 Hi



 I'm using asterisk 1.8



 Does anyone know how to use the queue reload command. The built in help
 doesn't really help.



 queue reload {parameters|membe Reload queues, members, queue rules, or
 parameters



 Regards



 Ish



 --

 Ishfaq Malik

 Department: VOIP Support

 Company: Packnet Limited

 t: +44 (0)845 004 4994

 f: +44 (0)161 660 9825

 e: i...@pack-net.co.uk

 w: http://www.pack-net.co.uk



 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House

 37 Ducie Street

 Manchester, M1 2JW

 COMPANY REG NO. 04920552


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-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] queue reload command

[Andrew Colin]

Hi



queue reload(queue name) or queue reload all



for example



queue reload reception



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, January 8, 2015 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] queue reload command



Hi



I'm using asterisk 1.8



Does anyone know how to use the queue reload command. The built in help 
doesn't really help.



queue reload {parameters|membe Reload queues, members, queue rules, or 
parameters



Regards



Ish




-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] queue log realtime mysql

[Jonas Kellens]

On 04-11-14 11:52, Jonas Kellens wrote:

On 04-11-14 11:50, Ishfaq Malik wrote:


On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be 
mailto:jonas.kell...@telenet.be wrote:


Hello,

I have 5 Asterisk servers all using mysql realtime to store queue
log information.

There is 1 out of 5 servers which stores the data in 4 columns :
'data1' -- 'data 5'.

All other servers store data in 1 column 'data' with the data
seperated by pipe.

I see no difference in my configuration of extconfig.conf and
logger.conf. Maybe a hidden default value ?

Can someone tell me which setting makes the mysql realtime driver
store data in 1 column or in seperate columns ?

Using Asterisk 1.8.12.2



Kind regards,

Jonas.



Are you using mysql_realtime or odbc with a mysql back end?



Using mysql_realtime, not using odbc.



Hello,

is there any more feedback on this ?

I still haven't found the difference in realtime configuration between 
this 1 server and my 4 other servers.



Kind regards,

Jonas.


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Re: [asterisk-users] queue log realtime mysql

[Ishfaq Malik]

On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be wrote:

  Hello,

 I have 5 Asterisk servers all using mysql realtime to store queue log
 information.

 There is 1 out of 5 servers which stores the data in 4 columns : 'data1'
 -- 'data 5'.

 All other servers store data in 1 column 'data' with the data seperated by
 pipe.

 I see no difference in my configuration of extconfig.conf and logger.conf.
 Maybe a hidden default value ?

 Can someone tell me which setting makes the mysql realtime driver store
 data in 1 column or in seperate columns ?

 Using Asterisk 1.8.12.2



 Kind regards,

 Jonas.



Are you using mysql_realtime or odbc with a mysql back end?


-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] queue log realtime mysql

[Jonas Kellens]

On 04-11-14 11:50, Ishfaq Malik wrote:


On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be 
mailto:jonas.kell...@telenet.be wrote:


Hello,

I have 5 Asterisk servers all using mysql realtime to store queue
log information.

There is 1 out of 5 servers which stores the data in 4 columns :
'data1' -- 'data 5'.

All other servers store data in 1 column 'data' with the data
seperated by pipe.

I see no difference in my configuration of extconfig.conf and
logger.conf. Maybe a hidden default value ?

Can someone tell me which setting makes the mysql realtime driver
store data in 1 column or in seperate columns ?

Using Asterisk 1.8.12.2



Kind regards,

Jonas.



Are you using mysql_realtime or odbc with a mysql back end?



Using mysql_realtime, not using odbc.


Kind regards,

Jonas
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Re: [asterisk-users] Queue is not working

[Ishfaq Malik]

On 22 May 2014 12:42, omakhileshchand omakhileshch...@gmail.com wrote:

 Dear All,
 I have make a queue in my dailplan and queue is not working
 properly,prbolem is that all call goes to same extenstion at a
 time.Because,I use eyeBeam(softphone) and eyeBeam have six line and
 whenever a call comes into eyeBeam that call reserved by Line 1 suppose to
 2nd call will come that call goes to Line 2(same extension used by Line 1)
 and 3rd call goes to 3rd line and so on.

 But i want to whenever 2nd call will come that call goes into different
 extentsion that call never hit into reserved extention.

 extenstion.conf

 [Queue_Test]
 exten = s,1,Answer ; Important, see notes
 exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really
 needed
 exten = s,3,Hangup()


 queues.conf

 [Queue_Test]
 music = default
 strategy = fewestcalls
 context = queue-out ; Here we go when the caller presses a single digit,
 while in the queue
 timeout = 15
 wrapuptime=10
 announce-frequency = 30
 announce-holdtime = yes
 joinempty = yes
 member = Sip/4001
 member = Sip/4003
 member = Sip/4004
 member = Sip/4005
 member = Sip/4006
 member = Sip/4007

 Regards
 Akhilesh


In your sip.conf have you got callcounter = yes set?
What stats is queue show Queue_Test showing at various times? (this will
give you an indication of how many calls each member has taken)
What happens when you choose rrmemory as the stratergy?
Have you read and fully understood the joinempty parameter?

Regards

Ish

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Queue is not working

[Mikael Fredin]

I would research the ringinuse option as well.


On 22 May 2014 13:42, omakhileshchand omakhileshch...@gmail.com wrote:

 Dear All,
 I have make a queue in my dailplan and queue is not working
 properly,prbolem is that all call goes to same extenstion at a
 time.Because,I use eyeBeam(softphone) and eyeBeam have six line and
 whenever a call comes into eyeBeam that call reserved by Line 1 suppose to
 2nd call will come that call goes to Line 2(same extension used by Line 1)
 and 3rd call goes to 3rd line and so on.

 But i want to whenever 2nd call will come that call goes into different
 extentsion that call never hit into reserved extention.

 extenstion.conf

 [Queue_Test]
 exten = s,1,Answer ; Important, see notes
 exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really
 needed
 exten = s,3,Hangup()


 queues.conf

 [Queue_Test]
 music = default
 strategy = fewestcalls
 context = queue-out ; Here we go when the caller presses a single digit,
 while in the queue
 timeout = 15
 wrapuptime=10
 announce-frequency = 30
 announce-holdtime = yes
 joinempty = yes
 member = Sip/4001
 member = Sip/4003
 member = Sip/4004
 member = Sip/4005
 member = Sip/4006
 member = Sip/4007

 Regards
 Akhilesh

 Sent from Samsung Mobile

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Re: [asterisk-users] Queue with linear strategy does not work

[Rusty Newton]

On Tue, Dec 10, 2013 at 10:14 PM, Thorben Jensen i...@thorben.dk wrote:
 I have a queue with linear strategy. When I add dynamic members it does NOT
 ring the members in the order they are added.

 I use the command AddQueueMember to add members but it seems to be random
 how it rings the members.

Does it ring them in the same order for every call(even if it is not
the expected order)?

Does the order change up for each call even when no new members have been added?

Can you provide a pastebin of a verbose log (see logger.conf)
demonstrating the problem?

-- 
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direct: +1 256 428 6200

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Re: [asterisk-users] Queue linear unordered feature when using realtime

[Steven Wheeler]

From: Leandro Dardini [mailto:ldard...@gmail.com]
Sent: Thursday, November 14, 2013 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Queue linear unordered feature when using realtime

Hello,
I was trying to use a queue in linear order and to provide the exact order of 
members to dial by adjusting the uniqueid value. Obviously it doesn't work and 
it seems an old problem:

https://issues.asterisk.org/jira/browse/ASTERISK-18480

Realtime configuration can't identify orders in the list of results, so the 
members for the queue are returned in random order.

Anyone experiencing the same problem? How do you solve it?

Leandro

I opened the ticket you linked to.  We ended up prefixing the interface value 
with an integer which indicated the agent's position in the queue.  In our 
dialplan this ended up looking like 'Local/001-agent@queue/n' our 'queue' 
context then strips off the prefix and continues as normal.

Steven Wheeler


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Re: [asterisk-users] Queue Management

[Lenz Emilitri]

This should happen automatically - not sure what you want to do.
l.


2013/9/26 akhilesh chand omakhileshch...@gmail.com:
 Dear All,


 I have six different campaign and  5 different agent have login on that
 campaign.Same thing i have done using agi and database,i never use queue
 management on this scenario. Agent can also shuffling  one campaign to
 anther campaign.
 Now i want to do some work with queue.I want to use single queue to managing
 this.

 Eg:
 campaign   Agent Login

 A   a_1,a_3
 (In campaign A 2 agents are login)
 B   a_2,a_1
 (In campaign B 2 agents are login)
 C   a_3,a_1,a_4
 (In campaign C 3 agents are login)
 D   a_4,a_5,a_3
 (In campaign D 3 agents are login)
 E   a_1,a_3,1_2
 (In campaign E 3 agents are login)
 Fa_5,a_4
 (In campaign F 2 agents are login)

 When a call come to campaign A that call goes to agent a_1 or a_3 not goes
 to other campaigns agents.

 Regards
 Akhilesh



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-- 
Loway - home of QueueMetrics - http://queuemetrics.com
Try the WombatDialer auto-dialer @ http://wombatdialer.com

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Re: [asterisk-users] queue member ackcall - cpuspikes

[zendel fernandez]

And further,

No matter what contains in the GOSUB (In this case relatively simple
stuff), when the A party hangup, the queue should signal the B
Channel(Member) to hangup. [ Which should tear down member LEG immediately ]

The problem here is Queue is not able to hangup the member leg even though
the original caller has disappeared [Just because B channels is in a GOSUB
].

.




On Thu, Aug 8, 2013 at 4:41 PM, zendel fernandez zendel.fernan...@gmail.com
 wrote:

 hi!

 GOSUB X
 * Presents Background message to the called party
 * check if there's any inputs from the user ( Press 1 etc )
 * exit if called party provide input *or not*
 

 See the example URL for for similar implementations.



 Regds




 On Thu, Aug 8, 2013 at 2:03 PM, Paul Belanger 
 paul.belan...@polybeacon.com wrote:

 On 13-08-07 08:42 PM, zendel fernandez wrote:

 hi!,

 Asterisk Version:1.6.1.20
 OS: CentOS release 5.3 (Final)
 uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686
 i386
 GNU/Linux
 Application: Queue
 Specific Details: Obtain Acknowledgement from queue member before
 bridging
 the caller.
 Language: AEL
 Similar Example:http://www.voip-info.**org/wiki/view/Asterisk+tips+**
 Queue+Member+ackcallhttp://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall

 Scenario:
 1. User calls in a General Number

 2. Call is queued in Queue Application

 3. Queue calls a Local/@members channel

 4. At members context:
 Dial The real member(called party) channel with a U(GOSUB X) routine
 4.1 The called party answers,  is led to the GOSUB routine X:
 Here the prompt is given to the called party to acknowledge the incoming
 call
 [ depending on the out put, this will return appropriate GOSUB result ]
 4.2 Based on the GOSUB result, the Dial proceeds

 5. The Queue proceeds based on the result taken at 4.2 above.
 i.e.
 Take it as a success  build the bridge between the caller  member
 Whether to DIAL the next member

 The Question: All goes well  the dial-plan works. If between step 4.1 
 4.2, the caller hangs up asterisk gives CPU spikes.
 Symptom: ASTERISK CLI gets stuck until step 4.2 returns.

 Console Error: app_dial.c: Could not stop autoservice on calling channel
 [ Somehow get the feeling that this is not the real error]

 What could be the reason for CPU SPIKES. How to avoid this ?

  What are you doing in your GOSUB X routine, you are likely blocking the
 thread in Asterisk, which is causing your autoservice errors (and yes, they
 are real errors) which increases the CPU on asterisk.

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

 --
 __**__**_
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Re: [asterisk-users] queue member ackcall - cpuspikes

[zendel fernandez]

hi!

GOSUB X
* Presents Background message to the called party
* check if there's any inputs from the user ( Press 1 etc )
* exit if called party provide input *or not*


See the example URL for for similar implementations.



Regds




On Thu, Aug 8, 2013 at 2:03 PM, Paul Belanger
paul.belan...@polybeacon.comwrote:

 On 13-08-07 08:42 PM, zendel fernandez wrote:

 hi!,

 Asterisk Version:1.6.1.20
 OS: CentOS release 5.3 (Final)
 uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686
 i386
 GNU/Linux
 Application: Queue
 Specific Details: Obtain Acknowledgement from queue member before bridging
 the caller.
 Language: AEL
 Similar Example:http://www.voip-info.**org/wiki/view/Asterisk+tips+**
 Queue+Member+ackcallhttp://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall

 Scenario:
 1. User calls in a General Number

 2. Call is queued in Queue Application

 3. Queue calls a Local/@members channel

 4. At members context:
 Dial The real member(called party) channel with a U(GOSUB X) routine
 4.1 The called party answers,  is led to the GOSUB routine X:
 Here the prompt is given to the called party to acknowledge the incoming
 call
 [ depending on the out put, this will return appropriate GOSUB result ]
 4.2 Based on the GOSUB result, the Dial proceeds

 5. The Queue proceeds based on the result taken at 4.2 above.
 i.e.
 Take it as a success  build the bridge between the caller  member
 Whether to DIAL the next member

 The Question: All goes well  the dial-plan works. If between step 4.1 
 4.2, the caller hangs up asterisk gives CPU spikes.
 Symptom: ASTERISK CLI gets stuck until step 4.2 returns.

 Console Error: app_dial.c: Could not stop autoservice on calling channel
 [ Somehow get the feeling that this is not the real error]

 What could be the reason for CPU SPIKES. How to avoid this ?

  What are you doing in your GOSUB X routine, you are likely blocking the
 thread in Asterisk, which is causing your autoservice errors (and yes, they
 are real errors) which increases the CPU on asterisk.

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger

 --
 __**__**_
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 asterisk-users mailing list
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Re: [asterisk-users] queue member ackcall - cpuspikes

[Paul Belanger]

On 13-08-07 08:42 PM, zendel fernandez wrote:

hi!,

Asterisk Version:1.6.1.20
OS: CentOS release 5.3 (Final)
uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386
GNU/Linux
Application: Queue
Specific Details: Obtain Acknowledgement from queue member before bridging
the caller.
Language: AEL
Similar 
Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall

Scenario:
1. User calls in a General Number

2. Call is queued in Queue Application

3. Queue calls a Local/@members channel

4. At members context:
Dial The real member(called party) channel with a U(GOSUB X) routine
4.1 The called party answers,  is led to the GOSUB routine X:
Here the prompt is given to the called party to acknowledge the incoming
call
[ depending on the out put, this will return appropriate GOSUB result ]
4.2 Based on the GOSUB result, the Dial proceeds

5. The Queue proceeds based on the result taken at 4.2 above.
i.e.
Take it as a success  build the bridge between the caller  member
Whether to DIAL the next member

The Question: All goes well  the dial-plan works. If between step 4.1 
4.2, the caller hangs up asterisk gives CPU spikes.
Symptom: ASTERISK CLI gets stuck until step 4.2 returns.

Console Error: app_dial.c: Could not stop autoservice on calling channel
[ Somehow get the feeling that this is not the real error]

What could be the reason for CPU SPIKES. How to avoid this ?

What are you doing in your GOSUB X routine, you are likely blocking the 
thread in Asterisk, which is causing your autoservice errors (and yes, 
they are real errors) which increases the CPU on asterisk.


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?

[Ishfaq Malik]

On 23 July 2013 23:18, Shishir Pokharel shishir.pokha...@on24.com wrote:

  Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT. 

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jarek Jarzebowski
 *Sent:* Tuesday, July 23, 2013 3:04 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Queue - how to jump to next member after NO
 ANSWER?

 ** **

 Hi all,

 I have a Queue with 3 members:

 SIP/100

 SIP/200

 SIP/300

 When call arrives SIP/100 is ringing.. After given timeout ringing stops
 but call is not routed to next member but SIP/100 starts ringing again.***
 *

 I know that this is because SIP/100 is still available in the Queue but is
 it any way to make a Queue witch strategy:

 call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after
 given number of time - hump to the next member?

 Thanks in advance.

 Jarek

 --
 _


This is in the queues.conf in 1.8

 If you want the queue to avoid sending calls to members whose devices are
 known to be 'in use' (via the channel driver supporting that device state)
 uncomment this option. (Note: only the SIP channel driver currently is able
 to report 'in use'.)

 ringinuse = no

-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552
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Re: [asterisk-users] Queue - how to jump to next member after NO ANSWER?

[Shishir Pokharel]

Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT.
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jarek Jarzebowski
Sent: Tuesday, July 23, 2013 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Queue - how to jump to next member after NO ANSWER?

Hi all,
I have a Queue with 3 members:
SIP/100
SIP/200
SIP/300
When call arrives SIP/100 is ringing.. After given timeout ringing stops but 
call is not routed to next member but SIP/100 starts ringing again.
I know that this is because SIP/100 is still available in the Queue but is it 
any way to make a Queue witch strategy:
call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given 
number of time - hump to the next member?
Thanks in advance.
Jarek
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Re: [asterisk-users] queue moh

[Ishfaq Malik]

Have you looked at mohsuggest in the sip configuration?

Regards

Ish


On 10 July 2013 17:55, Andrew Thomas a...@datavox.co.uk wrote:

 Hi All,

 Sorry if this has been covered already, but I don't tend to follow this
 list as close as I should these days.

 Problem is that if a call comes in to a queue without option 'r'
 specified - moh plays as expected.  Now, when that call is answered, all
 is fine. Trouble comes when that person then puts the caller on-hold.
 No moh is heard by the caller (in fact, they get silence).

 If I use 'r' - then ringing is heard - but the queue's
 musiconhold/musicclass is ignored completely.  When the caller is put on
 hold, they do hear moh but the default moh context is used - not the moh
 of the queue.

 What I need is for the queue's moh to be used when the caller is put on
 hold (and without using the 'r' feature).  Is this possible?

 * 1.8.16.0 (tried on various flavours of 1.8).
 Queue static and realtime (same outcome).

 Cheers
 Andy











 --

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 and are intended for the confidential use of the named recipient only.
 If you are not the intended recipient, employee or agent responsible
 for delivery of this message to the intended recipient, take note that
 any dissemination, distribution or copying of this communication and
 its attachments is strictly prohibited, and may be subject to civil or
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 Every effort has been made to ensure that this e-mail or any attachments
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-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552
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Re: [asterisk-users] queue moh

[Ioan Indreias]

Hello Andy,

Have you tried using SetMusicOnHold command before Queue command?

BR,
Ioan


On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote:

 Hi All,

 Sorry if this has been covered already, but I don't tend to follow this
 list as close as I should these days.

 Problem is that if a call comes in to a queue without option 'r'
 specified - moh plays as expected.  Now, when that call is answered, all
 is fine. Trouble comes when that person then puts the caller on-hold.
 No moh is heard by the caller (in fact, they get silence).

 If I use 'r' - then ringing is heard - but the queue's
 musiconhold/musicclass is ignored completely.  When the caller is put on
 hold, they do hear moh but the default moh context is used - not the moh
 of the queue.

 What I need is for the queue's moh to be used when the caller is put on
 hold (and without using the 'r' feature).  Is this possible?

 * 1.8.16.0 (tried on various flavours of 1.8).
 Queue static and realtime (same outcome).

 Cheers
 Andy











 --

  If you have received this communication in error we would appreciate
 you advising us either by telephone or return of e-mail. The contents
 of this message, and any attachments, are the property of DataVox,
 and are intended for the confidential use of the named recipient only.
 If you are not the intended recipient, employee or agent responsible
 for delivery of this message to the intended recipient, take note that
 any dissemination, distribution or copying of this communication and
 its attachments is strictly prohibited, and may be subject to civil or
 criminal action for which you may be liable.
 Every effort has been made to ensure that this e-mail or any attachments
 are free from viruses. While the company has taken every reasonable
 precaution to minimise this risk, neither company, nor the sender can
 accept liability for any damage which you sustain as a result of viruses.
 It is recommended that you should carry out your own virus checks
 before opening any attachments.

 Registered in England. No. 27459085.



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Re: [asterisk-users] Queue questions - Asterisk 11

[Administrator TOOTAI]

Le 04/07/2013 07:29, Satish Barot a écrit :

[...]

Already tested, I tried again as the option passed to queue was
changed (n option)

Logs:

-- Started music on hold, class 'default', on SIP/gw-005e
-- Executing [909@memberconnector:1] Dial(Local/909@
memberconnector-0002;2, SIP/s-ntfe_909,60,) in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/s-ntfe_909
-- SIP/s-ntfe_909-0060 is ringing
-- Local/909@memberconnector- 0002;1 is ringing
-- SIP/s-ntfe_909-0060 is ringing
-- Stopped music on hold on SIP/gw-005e
  == Spawn extension (macro-toQueue, s, 11) exited non-zero on
'SIP/gw-005e' in macro 'toQueue'
  == Spawn extension (incoming-Swiss-itech, 1, 204) exited
non-zero on 'SIP/gw-005e'
-- Executing [h@incoming-Swiss-itech:1]
NoOp(SIP/gw-005e, Call ended with QUEUESTATUS= and
DIALSTATUS= and HANGUPCAUSE=0) in new stack

From extension:

[memberconnector]
;
exten = _XXX,1,Dial(SIP/${peerPrefix}$ {EXTEN},${TIMERINGQUEUE},)
  same = n,NoOp(DIALSTATUS=${ DIALSTATUS})

As you can see, all status are empty,


-- 
Daniel




QUEUESTATUS will contain different values in different scenarios. i.e. 
If a call gets answered then the value is CONTINUE, If a call doesn't 
get answered and Queue timeout happens then TIMEOUT. If a caller hangs 
up when call is in Queue then QUEUESTATUS will be blank.


Have something like this,
... ...
same = n,queue(support,c,,,20)
same = n,Noop(QSTATUS=${QUEUESTATUS})
... ...
exten = h,1,Noop(QSTATUS=${QUEUESTATUS})

[memberconnector]
exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
same = n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same = n,Noop(DIALSTATUS=${DIALSTATUS})

exten = h,1,Noop(DIALSTATUS=${DIALSTATUS})



Hi Satish,

the c option has a result of DIALSTATUS shown on h extension in 
[memberconnector] and _only_ here. I then have to put the result in a 
global variable as DIALSTATUS is resetted.


Many thanks for your help

Regards

--
Daniel

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Re: [asterisk-users] Queue questions - Asterisk 11

[Satish Barot]

On Thu, Jul 4, 2013 at 5:36 PM, Administrator TOOTAI ad...@tootai.netwrote:

 Le 04/07/2013 07:29, Satish Barot a écrit :

 [...]

 Already tested, I tried again as the option passed to queue was
 changed (n option)

 Logs:

 -- Started music on hold, class 'default', on SIP/gw-005e
 -- Executing [909@memberconnector:1] Dial(Local/909@
 memberconnector-0002;2, SIP/s-ntfe_909,60,) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/s-ntfe_909
 -- SIP/s-ntfe_909-0060 is ringing
 -- Local/909@memberconnector- 0002;1 is ringing

 -- SIP/s-ntfe_909-0060 is ringing
 -- Stopped music on hold on SIP/gw-005e
   == Spawn extension (macro-toQueue, s, 11) exited non-zero on
 'SIP/gw-005e' in macro 'toQueue'
   == Spawn extension (incoming-Swiss-itech, 1, 204) exited
 non-zero on 'SIP/gw-005e'
 -- Executing [h@incoming-Swiss-itech:1]
 NoOp(SIP/gw-005e, Call ended with QUEUESTATUS= and
 DIALSTATUS= and HANGUPCAUSE=0) in new stack

 From extension:

 [memberconnector]
 ;
 exten = _XXX,1,Dial(SIP/${peerPrefix}$ {EXTEN},${TIMERINGQUEUE},)
   same = n,NoOp(DIALSTATUS=${ DIALSTATUS})

 As you can see, all status are empty,


 -- Daniel



 QUEUESTATUS will contain different values in different scenarios. i.e. If
 a call gets answered then the value is CONTINUE, If a call doesn't get
 answered and Queue timeout happens then TIMEOUT. If a caller hangs up when
 call is in Queue then QUEUESTATUS will be blank.

 Have something like this,
 ... ...
 same = n,queue(support,c,,,20)
 same = n,Noop(QSTATUS=${QUEUESTATUS})
 ... ...
 exten = h,1,Noop(QSTATUS=${**QUEUESTATUS})

 [memberconnector]
 exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
 same = n,Dial(SIP/${EXTEN})
 ;Check the Dialstatus for Member
 same = n,Noop(DIALSTATUS=${**DIALSTATUS})

 exten = h,1,Noop(DIALSTATUS=${**DIALSTATUS})


 Hi Satish,

 the c option has a result of DIALSTATUS shown on h extension in
 [memberconnector] and _only_ here. I then have to put the result in a
 global variable as DIALSTATUS is resetted.

 Many thanks for your help

 Regards
 --
 Daniel


That is because DIALSTATUS is set on a local channel and not on a caller
channel. Use SHARED Function to get the value back in caller channel.

same = n,Set(__ORIGCHANNEL=${CHANNEL})
same = n,queue(support,c,,,20)
same = n,Noop(QSTATUS=${QUEUESTATUS})
... ...
exten =
h,1,Noop(QSTATUS=${QUEUESTATUS},DIALSTATUS=${SHARED(DIALSTATUS,${ORIGCHANNEL})})

[memberconnector]
exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
same = n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same = n,Noop(DIALSTATUS=${DIALSTATUS})
same = n,Set(SHARED(DIALSTATUS,${ORIGCHANNEL})=${DIALSTATUS})

exten = h,1,Noop(DIALSTATUS=${DIALSTATUS})
same = n,Set(SHARED(DIALSTATUS,${ORIGCHANNEL})=${DIALSTATUS})


--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Queue questions - Asterisk 11

[Satish Barot]

On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI ad...@tootai.netwrote:

 Hi all,

 I have to questions about queues. Member is a phone like SIP/myphone and
 only one member in the queue.

 At first, DIALSTATUS doesn't return any status. How to now if a call in
 queue has been answered or if caller just hangup?

Queue application uses QUEUESTATUS and not DIALSTATUS


 Second, how to deal with timeout, I have strange behaviors. If I put
 timeout=60 in queue.conf and I call the queue passing also 60 as timeout
 value, asterisk is returning after 5000ms the 4000 then 2000 then 2000 aso.
 I can replace the 60sec value on both place, or 60 in queue conf and 10
 when calling queue, I never have a stable behavior and more, not what I
 want.

 Exemple: let say asterisk should try all 20 seconds to call the member for
 8 seconds: how to configure this? What I found, is to put timeout=0 in
 queue conf and passing 20 to queue, so caller stays in queue 20 seconds
 before timeout. But asterisk rings 20 seconds :-(

A snippet from queues.conf...
 A Queue has two different timeout values associated with it. One is
the  timeout parameter configured in queues.conf. This timeout specifies
the  amount of time to try ringing a member's phone before considering the
member to be unavailable. The other timeout value is the timeout argument
to the Queue() application. This timeout represents the absolute amount of
time to allow a caller to stay in the queue before the caller is removed
from the queue. 

Correct me if I am wrong but if you want your caller to be in Queue for 20
seconds and try calling member for 8 seconds you should have 20 as a
timeout argument in Queue application in your dialplan and in timeout in
queues.conf should be 8. Also check the 'retry' and 'timeoutpriority'
parameters for queues.conf


 I read documentation on voip-info.org, not very clear. timeout in
 queue.conf is only for calling agent, not members? If I put a timeout=8 in
 queue conf, how to tell asterisk to retry each 20 seconds playing MOH to
 the caller?


 Thanks for any hint

 --
 Daniel


--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Queue questions - Asterisk 11

[Administrator TOOTAI]

Hi Satish

Le 03/07/2013 09:15, Satish Barot a écrit :


On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI 
ad...@tootai.net mailto:ad...@tootai.net wrote:


Hi all,

I have to questions about queues. Member is a phone like
SIP/myphone and only one member in the queue.

At first, DIALSTATUS doesn't return any status. How to now if a
call in queue has been answered or if caller just hangup?

Queue application uses QUEUESTATUS and not DIALSTATUS


QUEUESTATUS returns the status of the queue not the one from the dial 
command used in the queue. And there is no such information I'm looking 
for in QUEUESTATUS.


Other clue?



Second, how to deal with timeout, I have strange behaviors. If I
put timeout=60 in queue.conf and I call the queue passing also 60
as timeout value, asterisk is returning after 5000ms the 4000 then
2000 then 2000 aso. I can replace the 60sec value on both place,
or 60 in queue conf and 10 when calling queue, I never have a
stable behavior and more, not what I want.

Exemple: let say asterisk should try all 20 seconds to call the
member for 8 seconds: how to configure this? What I found, is to
put timeout=0 in queue conf and passing 20 to queue, so caller
stays in queue 20 seconds before timeout. But asterisk rings 20
seconds :-(

A snippet from queues.conf...
 A Queue has two different timeout values associated with it. One 
is the  timeout parameter configured in queues.conf. This timeout 
specifies the  amount of time to try ringing a member's phone before 
considering the member to be unavailable. The other timeout value is 
the timeout argument to the Queue() application. This timeout 
represents the absolute amount of time to allow a caller to stay in 
the queue before the caller is removed from the queue. 


Correct me if I am wrong but if you want your caller to be in Queue 
for 20 seconds and try calling member for 8 seconds you should have 20 
as a timeout argument in Queue application in your dialplan and in 
timeout in queues.conf should be 8. Also check the 'retry' and 
'timeoutpriority' parameters for queues.conf


OK, I got it: parameter n was send to queue. Now everything is working 
like it should.


Thanks for your help.

--
Daniel

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Re: [asterisk-users] Queue questions - Asterisk 11

[Satish Barot]

On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI ad...@tootai.netwrote:

 Hi Satish

 Le 03/07/2013 09:15, Satish Barot a écrit :


 On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI 
 ad...@tootai.netmailto:
 ad...@tootai.net wrote:

 Hi all,

 I have to questions about queues. Member is a phone like
 SIP/myphone and only one member in the queue.

 At first, DIALSTATUS doesn't return any status. How to now if a
 call in queue has been answered or if caller just hangup?

 Queue application uses QUEUESTATUS and not DIALSTATUS


 QUEUESTATUS returns the status of the queue not the one from the dial
 command used in the queue. And there is no such information I'm looking for
 in QUEUESTATUS.

 Other clue?

Then you should add Local channel as a queue member and dial your SIP
member from Local channel context.  A little hint here. Suppose you have a
support queue configured in queues.conf

;queues.conf
[support]
... ...
member = Local/1000@memberconnector,0,John Smith,SIP/1000
... ...
Now In your dialplan add a context for local channel,
[memberconnector]
exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
same = n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same = n,Noop(DIALSTATUS=${DIALSTATUS})




 Second, how to deal with timeout, I have strange behaviors. If I
 put timeout=60 in queue.conf and I call the queue passing also 60
 as timeout value, asterisk is returning after 5000ms the 4000 then
 2000 then 2000 aso. I can replace the 60sec value on both place,
 or 60 in queue conf and 10 when calling queue, I never have a
 stable behavior and more, not what I want.

 Exemple: let say asterisk should try all 20 seconds to call the
 member for 8 seconds: how to configure this? What I found, is to
 put timeout=0 in queue conf and passing 20 to queue, so caller
 stays in queue 20 seconds before timeout. But asterisk rings 20
 seconds :-(

 A snippet from queues.conf...
  A Queue has two different timeout values associated with it. One is
 the  timeout parameter configured in queues.conf. This timeout specifies
 the  amount of time to try ringing a member's phone before considering the
 member to be unavailable. The other timeout value is the timeout argument
 to the Queue() application. This timeout represents the absolute amount of
 time to allow a caller to stay in the queue before the caller is removed
 from the queue. 

 Correct me if I am wrong but if you want your caller to be in Queue for
 20 seconds and try calling member for 8 seconds you should have 20 as a
 timeout argument in Queue application in your dialplan and in timeout in
 queues.conf should be 8. Also check the 'retry' and 'timeoutpriority'
 parameters for queues.conf


 OK, I got it: parameter n was send to queue. Now everything is working
 like it should.

 Thanks for your help.


 --
 Daniel


--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Queue questions - Asterisk 11

[Administrator TOOTAI]

Le 03/07/2013 15:07, Satish Barot a écrit :

[...]
Then you should add Local channel as a queue member and dial your SIP 
member from Local channel context.  A little hint here. Suppose you 
have a support queue configured in queues.conf


;queues.conf
[support]
... ...
member = Local/1000@memberconnector,0,John Smith,SIP/1000
... ...
Now In your dialplan add a context for local channel,
[memberconnector]
exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
same = n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same = n,Noop(DIALSTATUS=${DIALSTATUS})


Already tested, I tried again as the option passed to queue was changed 
(n option)


Logs:

-- Started music on hold, class 'default', on SIP/gw-005e
-- Executing [909@memberconnector:1] 
Dial(Local/909@memberconnector-0002;2, SIP/s-ntfe_909,60,) in 
new stack

  == Using SIP RTP CoS mark 5
-- Called SIP/s-ntfe_909
-- SIP/s-ntfe_909-0060 is ringing
-- Local/909@memberconnector-0002;1 is ringing
-- SIP/s-ntfe_909-0060 is ringing
-- Stopped music on hold on SIP/gw-005e
  == Spawn extension (macro-toQueue, s, 11) exited non-zero on 
'SIP/gw-005e' in macro 'toQueue'
  == Spawn extension (incoming-Swiss-itech, 1, 204) exited non-zero 
on 'SIP/gw-005e'
-- Executing [h@incoming-Swiss-itech:1] NoOp(SIP/gw-005e, 
Call ended with QUEUESTATUS= and DIALSTATUS= and HANGUPCAUSE=0) in new 
stack


From extension:

[memberconnector]
;
exten = _XXX,1,Dial(SIP/${peerPrefix}${EXTEN},${TIMERINGQUEUE},)
  same = n,NoOp(DIALSTATUS=${DIALSTATUS})

As you can see, all status are empty,

--
Daniel

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Re: [asterisk-users] Queue questions - Asterisk 11

[Satish Barot]

On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI ad...@tootai.netwrote:

 Le 03/07/2013 15:07, Satish Barot a écrit :

 [...]

 Then you should add Local channel as a queue member and dial your SIP
 member from Local channel context.  A little hint here. Suppose you have a
 support queue configured in queues.conf

 ;queues.conf
 [support]
 ... ...
 member = Local/1000@memberconnector,0,**John Smith,SIP/1000
 ... ...
 Now In your dialplan add a context for local channel,
 [memberconnector]
 exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
 same = n,Dial(SIP/${EXTEN})
 ;Check the Dialstatus for Member
 same = n,Noop(DIALSTATUS=${**DIALSTATUS})


 Already tested, I tried again as the option passed to queue was changed (n
 option)

 Logs:

 -- Started music on hold, class 'default', on SIP/gw-005e
 -- Executing [909@memberconnector:1] 
 Dial(Local/909@**memberconnector-0002;2,
 SIP/s-ntfe_909,60,) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/s-ntfe_909
 -- SIP/s-ntfe_909-0060 is ringing
 -- Local/909@memberconnector-**0002;1 is ringing
 -- SIP/s-ntfe_909-0060 is ringing
 -- Stopped music on hold on SIP/gw-005e
   == Spawn extension (macro-toQueue, s, 11) exited non-zero on
 'SIP/gw-005e' in macro 'toQueue'
   == Spawn extension (incoming-Swiss-itech, 1, 204) exited non-zero on
 'SIP/gw-005e'
 -- Executing [h@incoming-Swiss-itech:1] NoOp(SIP/gw-005e, Call
 ended with QUEUESTATUS= and DIALSTATUS= and HANGUPCAUSE=0) in new stack

 From extension:

 [memberconnector]
 ;
 exten = _XXX,1,Dial(SIP/${peerPrefix}$**{EXTEN},${TIMERINGQUEUE},)
   same = n,NoOp(DIALSTATUS=${**DIALSTATUS})

 As you can see, all status are empty,


 --
 Daniel



QUEUESTATUS will contain different values in different scenarios. i.e. If a
call gets answered then the value is CONTINUE, If a call doesn't get
answered and Queue timeout happens then TIMEOUT. If a caller hangs up when
call is in Queue then QUEUESTATUS will be blank.

Have something like this,
... ...
same = n,queue(support,c,,,20)
same = n,Noop(QSTATUS=${QUEUESTATUS})
... ...
exten = h,1,Noop(QSTATUS=${QUEUESTATUS})

[memberconnector]
exten = _X.,1,Noop(Connecting to Member at ${EXTEN})
same = n,Dial(SIP/${EXTEN})
;Check the Dialstatus for Member
same = n,Noop(DIALSTATUS=${DIALSTATUS})

exten = h,1,Noop(DIALSTATUS=${DIALSTATUS})

--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] Queue Ring inuse is shared ?

[Satish Barot]

I have 1.8.7.0, Realtime queue table with ringinuse set to 0, callcounter
set to yes in sip .conf for my SIP members.
Above allows me Queue not sending a call to a member when (s)he is on
call(Be it from same Queue or any other call). Member can also
transfer(through features.conf) a call without any issue.

call-limit I think is deprecated in 1.8.

--Satish Barot
Ahmedabad, India




On Sat, Jun 22, 2013 at 2:41 PM, Shanavaz E A shanava...@yahoo.com wrote:

 Hi,

 I use asterisk 1.8.

 My issue is : I have the same SIP members added to two queues. I use
 realtime configuration and has set the field ringinuse=0 for both the
 queues. But if an extension is answering the call in one queue, and some
 new call comes in the second queue, still that extension is ringed. In the
 queue_log table I am getting RINGNOANSWER events each second for the
 extension until the call gets answered.

 Is this a normal behaviour ? Can we prevent it? Can we set not to ring
 any queue member if he is answering a call either in the same queue or a
 different queue? Pls guide me.

 Regards
 Shanavaz.


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Re: [asterisk-users] Queue Ring inuse is shared ?

[Barry Flanagan]

On 22 June 2013 10:11, Shanavaz E A shanava...@yahoo.com wrote:

 Hi,

 I use asterisk 1.8.

 My issue is : I have the same SIP members added to two queues. I use
 realtime configuration and has set the field ringinuse=0 for both the
 queues.


Should that not be ringinuse = no?

-Barry
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Re: [asterisk-users] Queue Ring inuse is shared ?

[Shanavaz E A]

Hi,

I found in another mail that setting call-limit=1 in the sip configuration 
works. I tried that. It works but in that case the agents are not able to 
transfer the call to another extension, because only one call is allowed at a 
time.

Any other methods ?

Thanks  Regards
Shanavaz.




 From: Shanavaz E A shanava...@yahoo.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com 
Sent: Saturday, June 22, 2013 1:11 PM
Subject: [asterisk-users] Queue Ring inuse is shared ?
 


Hi,

I use asterisk 1.8.

My issue is : I have the same SIP members added to two queues. I use realtime 
configuration and has set the field ringinuse=0 for both the queues. But if an 
extension is answering the call in one queue, and some new call comes in the 
second queue, still that extension is ringed. In the queue_log table I am 
getting RINGNOANSWER events each second for the extension until the call gets 
answered.

Is this a normal behaviour ? Can we prevent it? Can we set not to ring any 
queue member if he is answering a call either in the same queue or a different 
queue? Pls guide me.

Regards
Shanavaz.

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Re: [asterisk-users] Queue Limit Callers

[Shanavaz E A]

Thanks all for the inputs... Let me work on it and come back again with some 
results...






 From: Ioan Indreias indre...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Tuesday, June 18, 2013 1:43 PM
Subject: Re: [asterisk-users] Queue Limit Callers
 


Hello Shanavaz.,



Please find some quick thoughts:

* 2 main queues
* agents logged on one or on both main queues
* before sending a new call to one of the main queues check the number of 
waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for 30 
sec) the call on a empty members queue/parking slot/music-on-hold if the queue 
threshold is reached.

The threshold could be read from a database, internal astdb or could be set as 
a global variable updated when agents login/logout/pause/unpause or could be 
dynamically computed based on QUEUE_MEMBER_COUNT / QUEUE_MEMBER_LIST

After the divert period is ended the call will return and the threshold is 
checked again, etc.

This method have some negative impacts (the entry position number for calls 
over the threshold //origposition// will have no meaning, a newer call could be 
served before an older one, etc.) but you could manipulate the call flow 
exactly how you want.

HTH,
Ioan
http://www.modulo.ro


On Tue, Jun 18, 2013 at 12:05 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:

You should have different sets of agents logged in to different queues and you 
should have a monitor to move them from one queue to the other based on 
incoming traffic.

l.



2013/6/17 Shanavaz E A shanava...@yahoo.com

Hi,


I have a requirement, which I am not sure whether it can be implemented. I 
had done some searches but didnt find an answer to this. Kindly let me know 
if some one has an idea to implement this:


I have two Queues - Sales  Booking
I have 12 Agents who are added to both the queues



Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales 
Queue.


Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls 
should remain in hold.
4 calls in the Sales Queue should hit the other 4 agents and the other 2 call 
should be in hold.


Means at a time a maximum of 8 Booking calls only should hit the agents and 4 
Sales Calls only should hit the agents.


If number of logged in agents are less, proportionally the number of call 
limit should be reduced. For example, if there are only 10 agents, 7 Booking 
Calls should hit and 3 Sales calls should hit. The idea is that all agents 
should be able to answer calls in both queues in rotation. Otherwise its 
possible to add some agents to booking queue and other agents to sales queue. 
But thats not what is required.



Kindly help if there is some idea to implement this.


RegardsShanavaz.

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Re: [asterisk-users] Queue Limit Callers

[Barry Flanagan]

On 17 June 2013 11:02, Shanavaz E A shanava...@yahoo.com wrote:

 Hi,

 I have a requirement, which I am not sure whether it can be implemented. I
 had done some searches but didnt find an answer to this. Kindly let me know
 if some one has an idea to implement this:



I am not aware of an existing way to do this. By default, Asterisk does not
appear to be able to dynamically change the priority of a queue, which
seems to be what you are after.

On one of my systems I implemented the patch at
https://issues.asterisk.org/jira/browse/ASTERISK-17570 which, although it
does not do exactly what you are after, it does help a great deal.
Basically it dynamically increases the priority of callers based on their
hold time across all queues, so that for a given agent they will be
presented with the call having the longest wait time across all the queues
they are a member of. This has made a big difference to our avg hold time,
as queues are no longer competing against one anther for available agents.

Hope this helps.

-Barry Flanagan

 I have two Queues - Sales  Booking
 I have 12 Agents who are added to both the queues

 Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales
 Queue.

 Only 8 calls in the Booking Queue should hit the Agents and the other 4
 calls should remain in hold.
 4 calls in the Sales Queue should hit the other 4 agents and the other 2
 call should be in hold.

 Means at a time a maximum of 8 Booking calls only should hit the agents
 and 4 Sales Calls only should hit the agents.

 If number of logged in agents are less, proportionally the number of call
 limit should be reduced. For example, if there are only 10 agents, 7
 Booking Calls should hit and 3 Sales calls should hit. The idea is that all
 agents should be able to answer calls in both queues in rotation. Otherwise
 its possible to add some agents to booking queue and other agents to sales
 queue. But thats not what is required.

 Kindly help if there is some idea to implement this.

 Regards
 Shanavaz.

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Re: [asterisk-users] Queue Limit Callers

[Lenz Emilitri]

You should have different sets of agents logged in to different queues and
you should have a monitor to move them from one queue to the other based on
incoming traffic.
l.


2013/6/17 Shanavaz E A shanava...@yahoo.com

 Hi,

 I have a requirement, which I am not sure whether it can be implemented. I
 had done some searches but didnt find an answer to this. Kindly let me know
 if some one has an idea to implement this:

 I have two Queues - Sales  Booking
 I have 12 Agents who are added to both the queues

 Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales
 Queue.

 Only 8 calls in the Booking Queue should hit the Agents and the other 4
 calls should remain in hold.
 4 calls in the Sales Queue should hit the other 4 agents and the other 2
 call should be in hold.

 Means at a time a maximum of 8 Booking calls only should hit the agents
 and 4 Sales Calls only should hit the agents.

 If number of logged in agents are less, proportionally the number of call
 limit should be reduced. For example, if there are only 10 agents, 7
 Booking Calls should hit and 3 Sales calls should hit. The idea is that all
 agents should be able to answer calls in both queues in rotation. Otherwise
 its possible to add some agents to booking queue and other agents to sales
 queue. But thats not what is required.

 Kindly help if there is some idea to implement this.

 Regards
 Shanavaz.

 --
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Re: [asterisk-users] Queue Limit Callers

[Ioan Indreias]

Hello Shanavaz.,

Please find some quick thoughts:

* 2 main queues
* agents logged on one or on both main queues
* before sending a new call to one of the main queues check the number of
waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for
30 sec) the call on a empty members queue/parking slot/music-on-hold if the
queue threshold is reached.

The threshold could be read from a database, internal astdb or could be set
as a global variable updated when agents login/logout/pause/unpause or
could be dynamically computed based on QUEUE_MEMBER_COUNT /
QUEUE_MEMBER_LIST

After the divert period is ended the call will return and the threshold is
checked again, etc.

This method have some negative impacts (the entry position number for calls
over the threshold //origposition// will have no meaning, a newer call
could be served before an older one, etc.) but you could manipulate the
call flow exactly how you want.

HTH,
Ioan
http://www.modulo.ro


On Tue, Jun 18, 2013 at 12:05 PM, Lenz Emilitri lenz.lo...@gmail.comwrote:

 You should have different sets of agents logged in to different queues and
 you should have a monitor to move them from one queue to the other based on
 incoming traffic.
 l.


 2013/6/17 Shanavaz E A shanava...@yahoo.com

 Hi,

 I have a requirement, which I am not sure whether it can be implemented.
 I had done some searches but didnt find an answer to this. Kindly let me
 know if some one has an idea to implement this:

 I have two Queues - Sales  Booking
 I have 12 Agents who are added to both the queues

 Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales
 Queue.

 Only 8 calls in the Booking Queue should hit the Agents and the other 4
 calls should remain in hold.
 4 calls in the Sales Queue should hit the other 4 agents and the other 2
 call should be in hold.

 Means at a time a maximum of 8 Booking calls only should hit the agents
 and 4 Sales Calls only should hit the agents.

 If number of logged in agents are less, proportionally the number of call
 limit should be reduced. For example, if there are only 10 agents, 7
 Booking Calls should hit and 3 Sales calls should hit. The idea is that all
 agents should be able to answer calls in both queues in rotation. Otherwise
 its possible to add some agents to booking queue and other agents to sales
 queue. But thats not what is required.

 Kindly help if there is some idea to implement this.

 Regards
 Shanavaz.

 --
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Re: [asterisk-users] Queue Periodic Announce not working...

[Doug Lytle]

 periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav 

Try it without the .wav 

Doug 

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Re: [asterisk-users] Queue Periodic Announce not working...

[Gopalakrishnan N]

It works.

Thanks
 On 30 May 2013 19:39, Doug Lytle supp...@drdos.info wrote:

  periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav

 Try it without the .wav

 Doug

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonathan Rose]

I was poking around with the Add/Remove QueueMember code a while back.  From 
the sound of what you are saying I might have just missed something critical. 
for your case.

It'd be good to know what version you are using so that I can verify whether or 
not my changes could have affected you.

- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Saturday, December 8, 2012 5:55:39 AM
Subject: [asterisk-users] Queue joinempty, even after AddQueueMember


Hello, 

I add a member to a queue with AddQueueMember, but the Queue still indicates 
joinempty : 

Add member to queue : 

-- Executing [queueadd@sub-GetParams:2] AddQueueMember(SIP/sip17-5c1e, 
myqueue11,member3) in new stack 
-- Executing [queueadd@sub-GetParams:3] NoOp(SIP/sip17-5c1e, AQMSTATUS = 
ADDED) in new stack 

... but JOINEMPTY when entering the Call Queue : 

-- Executing [queue@pbx-routing:4] Queue(SIP/SipIncoming-5da9, 
myqueue1160) in new stack 
-- Executing [queue@pbx-routing:5] NoOp(SIP/SipIncoming-5da9, 
queuestatus == JOINEMPTY) in new stack 


How is this possible ? 



Kind regards, 
Jonas. 

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonas Kellens]

Hello,

using Asterisk 1.8.12.2


Jonas.

On 09-12-12 09:15, Jonathan Rose wrote:

I was poking around with the Add/Remove QueueMember code a while back.  From 
the sound of what you are saying I might have just missed something critical. 
for your case.

It'd be good to know what version you are using so that I can verify whether or 
not my changes could have affected you.

- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Saturday, December 8, 2012 5:55:39 AM
Subject: [asterisk-users] Queue joinempty, even after AddQueueMember


Hello,

I add a member to a queue with AddQueueMember, but the Queue still indicates 
joinempty :

Add member to queue :

-- Executing [queueadd@sub-GetParams:2] AddQueueMember(SIP/sip17-5c1e, 
myqueue11,member3) in new stack
-- Executing [queueadd@sub-GetParams:3] NoOp(SIP/sip17-5c1e, AQMSTATUS = 
ADDED) in new stack

... but JOINEMPTY when entering the Call Queue :

-- Executing [queue@pbx-routing:4] Queue(SIP/SipIncoming-5da9, 
myqueue1160) in new stack
-- Executing [queue@pbx-routing:5] NoOp(SIP/SipIncoming-5da9, queuestatus == 
JOINEMPTY) in new stack


How is this possible ?



Kind regards,
Jonas.

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonathan Rose]

Jonas Kellens wrote:
 Hello,
 
 using Asterisk 1.8.12.2

I think that was tagged before any of my recent app_queue patches. In that case,
it might work if you just update to the latest 1.8 release. If it doesn't, go 
ahead and
file an issue on JIRA.

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonas Kellens]

it might work...

How come app_queue is suddenly so unstable ?

Which version has a stable app_queue ?
I thought unstable versions are released with rc- added ?



Kind regards,
Jonas.


On 09-12-12 19:19, Jonathan Rose wrote:

Jonas Kellens wrote:

Hello,

using Asterisk 1.8.12.2

I think that was tagged before any of my recent app_queue patches. In that case,
it might work if you just update to the latest 1.8 release. If it doesn't, go 
ahead and
file an issue on JIRA.

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direct +1 256 428 6139

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Joshua Colp]

Jonas Kellens wrote:

it might work...


Without labbing things up with your exact scenario Jonathan can't 
confirm it. I did a quick search of the issue tracker for anything open 
similar to the issue you specified and nothing came up. The 
functionality you are using is commonly used so either it's something 
specific to how you are using it or was an issue in the version you are 
using and is not in recent versions.


As well - if the log you provided has not been altered then you are 
attempting to add an interface member3 to the queue. While this will 
succeed it is ultimately not a valid interface and would not be 
considered as available. This would explain why it does not work.



How come app_queue is suddenly so unstable ?


I don't quite know what you are referring to here. There are issues but 
it is in use by many companies and works for them, within the confines 
of what it can do.



Which version has a stable app_queue ?


You'd have to be specific in what you mean by stable.


I thought unstable versions are released with rc- added ?


rc stands for release candidate. It is not inherently unstable, it's 
just a candidate for release put out there for testing. It helps to 
uncover issues.


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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonas Kellens]

On 09-12-12 19:49, Joshua Colp wrote:

Jonas Kellens wrote:

it might work...


Without labbing things up with your exact scenario Jonathan can't 
confirm it. I did a quick search of the issue tracker for anything 
open similar to the issue you specified and nothing came up. The 
functionality you are using is commonly used so either it's something 
specific to how you are using it or was an issue in the version you 
are using and is not in recent versions.


As well - if the log you provided has not been altered then you are 
attempting to add an interface member3 to the queue. While this will 
succeed it is ultimately not a valid interface and would not be 
considered as available. This would explain why it does not work.


Hello,

what is then a correct interface ? SIP/member3 maybe is more correct ?


Thanks for your help.



Jonas.

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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Joshua Colp]

Jonas Kellens wrote:

On 09-12-12 19:49, Joshua Colp wrote:

As well - if the log you provided has not been altered then you are
attempting to add an interface member3 to the queue. While this will
succeed it is ultimately not a valid interface and would not be
considered as available. This would explain why it does not work.


Hello,


Hola,


what is then a correct interface ? SIP/member3 maybe is more correct ?


That is correct. That type of string is what interface refers to in the 
AddQueueMember documentation. SIP/member3, IAX2/joe, etc.


Cheers,

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Queue joinempty, even after AddQueueMember

[Jonas Kellens]

On 09-12-12 20:10, Joshua Colp wrote:

Jonas Kellens wrote:

On 09-12-12 19:49, Joshua Colp wrote:

As well - if the log you provided has not been altered then you are
attempting to add an interface member3 to the queue. While this will
succeed it is ultimately not a valid interface and would not be
considered as available. This would explain why it does not work.


Hello,


Hola,


what is then a correct interface ? SIP/member3 maybe is more correct ?


That is correct. That type of string is what interface refers to in 
the AddQueueMember documentation. SIP/member3, IAX2/joe, etc.


Cheers,



Hello,

I will try that. It might be the solution...


Jonas.



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Re: [asterisk-users] Queue logging

[Lenz Emilitri]

How large is your systems? because the information created by of a call on
a queue is just like a hundred bytes, so it is usually safe to keep them
all in any case on modern systems.


2012/11/27 Jonas Kellens jonas.kell...@telenet.be

  Hello,

 at the moment I am logging queues into a MySQL DB, but this can quickly
 become a lot of information.

 Is there a way to exclude certain queues from being logged into the queue
 log ?



 Thanks,
 Jonas.

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Re: [asterisk-users] Queue logging

[mthayeb]

Sent from My Blackberry® @ Tata Docomo

-Original Message-
From: Lenz Emilitri lenz.lo...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 28 Nov 2012 13:18:11 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Queue logging

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Re: [asterisk-users] Queue logging

[Danny Nicholas]

Are you using triggering?  If so, perhaps you could modify the trigger
values.  PS asterisk version?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, November 27, 2012 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Queue logging

 

Hello,

at the moment I am logging queues into a MySQL DB, but this can quickly
become a lot of information.

Is there a way to exclude certain queues from being logged into the queue
log ?



Thanks,
Jonas.

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Re: [asterisk-users] Queue logging

[Jonas Kellens]

Hello,

I am not using triggering (what is this ?).

Just using extconfig.conf

Asterisk 1.8.12.2


Kind regards,
Jonas.


On 27-11-12 17:28, Danny Nicholas wrote:


Are you using triggering?  If so, perhaps you could modify the trigger 
values.  PS asterisk version?


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, November 27, 2012 10:21 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Queue logging

Hello,

at the moment I am logging queues into a MySQL DB, but this can 
quickly become a lot of information.


Is there a way to exclude certain queues from being logged into the 
queue log ?




Thanks,
Jonas.



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Re: [asterisk-users] Queue logging

[Danny Nicholas]

Triggering is a MYSQL mechanism that forces database action on specified
conditions.  My best guess is that you would have to tweak
addons/res_config_mysql.c to be able to filter logs.  It would probably be
easier to write a daemon to clear the unwanted data on a periodic basis.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, November 27, 2012 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue logging

 

Hello,

I am not using triggering (what is this ?).

Just using extconfig.conf

Asterisk 1.8.12.2


Kind regards,
Jonas.



On 27-11-12 17:28, Danny Nicholas wrote:

Are you using triggering?  If so, perhaps you could modify the trigger
values.  PS asterisk version?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, November 27, 2012 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Queue logging

 

Hello,

at the moment I am logging queues into a MySQL DB, but this can quickly
become a lot of information.

Is there a way to exclude certain queues from being logged into the queue
log ?



Thanks,
Jonas.






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Re: [asterisk-users] Queue logging

[Jonas Kellens]

Ah OK, that triggering I know.

I though maybe there was some kind of setting on a per queue base that 
could control the logging, like there is amaflags on a peer.



Jonas.


On 27-11-12 20:53, Danny Nicholas wrote:


Triggering is a MYSQL mechanism that forces database action on 
specified conditions.  My best guess is that you would have to tweak 
addons/res_config_mysql.c to be able to filter logs.  It would 
probably be easier to write a daemon to clear the unwanted data on a 
periodic basis.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, November 27, 2012 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Queue logging

Hello,

I am not using triggering (what is this ?).

Just using extconfig.conf

Asterisk 1.8.12.2


Kind regards,
Jonas.

On 27-11-12 17:28, Danny Nicholas wrote:

Are you using triggering?  If so, perhaps you could modify the
trigger values.  PS asterisk version?

*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Tuesday, November 27, 2012 10:21 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Queue logging

Hello,

at the moment I am logging queues into a MySQL DB, but this can
quickly become a lot of information.

Is there a way to exclude certain queues from being logged into
the queue log ?



Thanks,
Jonas.




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Re: [asterisk-users] Queue Member login from IAX trunk

[SamyGo]

Hi,

exten = 105,n,Read(AGENT_SIP,agent-**newlocation)
exten = 105,n,Set(AGENT_SIP=${DB(**agent_sip/${agent-newlocation}**)})

Above two lines are very suspicious: AGENT_SIP is a variable which is
getting some DTMF from caller.  agent-**newlocation  is the message you
want to be played while getting the AGENT_SIP input.
What is the next line doing in SET() !!? Please explain that.

Regards,
Sammy

On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger ja...@j-mb.dewrote:

 Hello

 i got two Asterisk servers connected with IAX2.

 At server 1 i hosted a queue named support. Now i want the Agent to login
 from SIP Phones Connected to server B.
 Therefore i wrote a extension like

 exten = 105,1,Authenticate(1234)
 exten = 105,n,AddQueueMember(support,,**)
 exten = 105,n,Read(AGENT_SIP,agent-**newlocation)
 exten = 105,n,Set(AGENT_SIP=${DB(**agent_sip/${agent-newlocation}**)})
 exten = 105,n,Playback(agent-loginok)
 exten = 105,n,Playback(vm-goodbye)
 exten = 105,n,Hangup

 Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the
 entered number in agent-newlocation. Do you have any ideas how to solve
 that?

 Best regards Jakob


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Re: [asterisk-users] Queue Member login from IAX trunk

[Jakob-Matthias Böttger]

mhh that makes sense indeed. My intention is to add Agents wich are 
connected to serverb to the queue at servera. But i don't know how to 
tell the queue that the Agent ist reachable at IAX/serverb/agent_sip


so the extension should be something like

exten = 105,n,Read(AGENT_SIP,agent-newlocation)
exten = 105,n,Set(AGENT_SIP=${DB(IAX2/intranet/agent_ip)})

so the agent enters the number from the phone he is connected. Then 
Asterisk adds IAX2/serverb to the number and saves it as agend phone 
number...


Regards Jakob


Am 04.07.2012 11:45, schrieb SamyGo:

Hi,

exten = 105,n,Read(AGENT_SIP,agent-newlocation)
exten = 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})

Above two lines are very suspicious: AGENT_SIP is a variable which is 
getting some DTMF from caller. agent-newlocation is the message you 
want to be played while getting the AGENT_SIP input.

What is the next line doing in SET() !!? Please explain that.

Regards,
Sammy

On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger ja...@j-mb.de 
mailto:ja...@j-mb.de wrote:


Hello

i got two Asterisk servers connected with IAX2.

At server 1 i hosted a queue named support. Now i want the Agent
to login from SIP Phones Connected to server B.
Therefore i wrote a extension like

exten = 105,1,Authenticate(1234)
exten = 105,n,AddQueueMember(support,,)
exten = 105,n,Read(AGENT_SIP,agent-newlocation)
exten = 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
exten = 105,n,Playback(agent-loginok)
exten = 105,n,Playback(vm-goodbye)
exten = 105,n,Hangup

Now server A takes IAX/serverB as argument for AGENT_SIP but
deletes the entered number in agent-newlocation. Do you have any
ideas how to solve that?

Best regards Jakob


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Re: [asterisk-users] Queue Member login from IAX trunk

[SamyGo]

hmm before I think of any solution: All I can tell you is that if you're
trying to use ;
exten = 105,n,Set(AGENT_SIP=${*DB(IAX2/intranet/agent_ip)*})

Both your Asterisk's server will need to have this DB in common/accessible.

Got it.. Use realtime Queues on Server-A ;
when Agent Logs in to Server B put values of the new Agent in MySQL
queue-memebers table:

Only Server-A needs to be in realtime. Server-B will only be inserting and
deleting agents from the table.

I hope it works.
Sammy


On Wed, Jul 4, 2012 at 2:57 PM, Jakob-Matthias Böttger ja...@j-mb.dewrote:

  mhh that makes sense indeed. My intention is to add Agents wich are
 connected to serverb to the queue at servera. But i don't know how to tell
 the queue that the Agent ist reachable at IAX/serverb/agent_sip

 so the extension should be something like

 exten = 105,n,Read(AGENT_SIP,agent-newlocation)
 exten = 105,n,Set(AGENT_SIP=${DB(IAX2/intranet/agent_ip)})

 so the agent enters the number from the phone he is connected. Then
 Asterisk adds IAX2/serverb to the number and saves it as agend phone
 number...

 Regards Jakob


 Am 04.07.2012 11:45, schrieb SamyGo:

 Hi,

 exten = 105,n,Read(AGENT_SIP,agent-newlocation)
 exten = 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})

 Above two lines are very suspicious: AGENT_SIP is a variable which is
 getting some DTMF from caller.  agent-newlocation  is the message you
 want to be played while getting the AGENT_SIP input.
 What is the next line doing in SET() !!? Please explain that.

 Regards,
 Sammy

  On Wed, Jul 4, 2012 at 2:32 PM, Jakob-Matthias Böttger ja...@j-mb.dewrote:

 Hello

 i got two Asterisk servers connected with IAX2.

 At server 1 i hosted a queue named support. Now i want the Agent to login
 from SIP Phones Connected to server B.
 Therefore i wrote a extension like

 exten = 105,1,Authenticate(1234)
 exten = 105,n,AddQueueMember(support,,)
 exten = 105,n,Read(AGENT_SIP,agent-newlocation)
 exten = 105,n,Set(AGENT_SIP=${DB(agent_sip/${agent-newlocation})})
 exten = 105,n,Playback(agent-loginok)
 exten = 105,n,Playback(vm-goodbye)
 exten = 105,n,Hangup

 Now server A takes IAX/serverB as argument for AGENT_SIP but deletes the
 entered number in agent-newlocation. Do you have any ideas how to solve
 that?

 Best regards Jakob


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Re: [asterisk-users] Queue callers with Callback option, without lose their place

[Christina Casey]

Hi,

In response to the questions that have been asked about this - it's very 
easy to achieve with Asterisk. Take a look at www.orderlyq.com for 
further details.


Christina Casey BA (Hons)
Senior Account Manager
Orderly Software



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Re: [asterisk-users] Queue callers with Callback option without lose their place

[Satish Barot]

I believe you want your caller to request for a callback while he/she waits
in a queue and when your agents are free, you want to call him back and
place in a same position in a Queue where he/she has left the Queue.

There exists an ugly(!) way of doing this.

(1)Set parameter 'context' in queues.conf to some real context available in
your dialplan
(2)Set 'setqueueentryvar' and 'setqueuevar' to yes in queues.conf
(3)Set paramet 'periodic-announce' to a custom audio file name announcing
to caller somethink like ..'To get a callback press any key any'.(This
sends the caller into context set by 'context' parameter when s/he presses
any key while waiting in a queue)
(4)A variable 'QUEUEPOSITION' would give you a last position of caller in a
queue. (You can get this variable in a context set by 'context' parameter.
Store the value somewhere in Database)
(5)When you think your Agents are free, Generate a callfile OR use AMI to
call the caller who has requested a callback.
(6)Once call is answered, send him to Queue application with 'position'
parameter set to the value of 'QUEUEPOSITION' of caller from database.

--Satish Barot

On Thu, May 31, 2012 at 9:18 PM, equis software equissoftw...@gmail.comwrote:

 Is there any option in Asterisk distribution of this?

 Thanks.

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Re: [asterisk-users] Queue callers with Callback option without lose their place

[Israel Gottlieb]

http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback



On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot satish4aster...@gmail.comwrote:

 I believe you want your caller to request for a callback while he/she
 waits in a queue and when your agents are free, you want to call him back
 and place in a same position in a Queue where he/she has left the Queue.

 There exists an ugly(!) way of doing this.

 (1)Set parameter 'context' in queues.conf to some real context available
 in your dialplan
 (2)Set 'setqueueentryvar' and 'setqueuevar' to yes in queues.conf
 (3)Set paramet 'periodic-announce' to a custom audio file name announcing
 to caller somethink like ..'To get a callback press any key any'.(This
 sends the caller into context set by 'context' parameter when s/he presses
 any key while waiting in a queue)
 (4)A variable 'QUEUEPOSITION' would give you a last position of caller in
 a queue. (You can get this variable in a context set by 'context'
 parameter. Store the value somewhere in Database)
 (5)When you think your Agents are free, Generate a callfile OR use AMI to
 call the caller who has requested a callback.
 (6)Once call is answered, send him to Queue application with 'position'
 parameter set to the value of 'QUEUEPOSITION' of caller from database.

 --Satish Barot

 On Thu, May 31, 2012 at 9:18 PM, equis software 
 equissoftw...@gmail.comwrote:

 Is there any option in Asterisk distribution of this?

 Thanks.

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Re: [asterisk-users] Queue callers with Callback option without lose their place

[Miguel Molina]

Known as Virtual Hold, you'll have to program inside asterisk to achieve 
that.


El 31/05/12 10:48, equis software escribió:

Is there any option in Asterisk distribution of this?

Thanks.


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Re: [asterisk-users] Queue callers with Callback option without lose their place

[equis software]

was what I was afraid ...
Thanks

2012/5/31 Miguel Molina mfmolina-lis...@millenium.com.co

  Known as Virtual Hold, you'll have to program inside asterisk to achieve
 that.

 El 31/05/12 10:48, equis software escribió:

 Is there any option in Asterisk distribution of this?

 Thanks.


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Re: [asterisk-users] Queue option 'R'

[bakko]

Hello,

I think this option work only with Asterisk 1.8.X

On the Asterisk 1.8.X CHANGES files:

* Added 'R' option to app_queue.  This option stops moh and indicates 
ringing
  to the caller when an Agent's phone is ringing.  This can be used to 
indicate
  to the caller that their call is about to be picked up, which is nice 
when

  one has been on hold for an extened period of time.

Regards 



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Re: [asterisk-users] Queue option 'R'

[georg]

Hi,

 I think this option work only with Asterisk 1.8.X

Ah yes, I see. Now Asterisk Trunk in the description makes sense...

Thanks,
Georg


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Re: [asterisk-users] queue not skipping ringing phone

[Sebastian Denz]

Am Mittwoch 21 Dezember 2011, 07:04:03 schrieb Matt Hamilton:
 I have a queue that distributes calls among 3 phones. When a phone is in
 use (including on hold), queue skips that device and sends the call to the
 next available one as expected. On the other hand, if a call comes in
 while one of the phones is ringing, the queue doesn't seem to recognize
 that phone as in use and sends the second call to the ringing phone. If
 the first call is answered, the second call is sent to the next available
 phone right away.
 
 I'm new to asterisk and wondering if this is normal; I thought the ringing
 phone would be skipped as in use as well. Is there a setting on the
 asterisk side that I can use to force the queue to skip the ringing
 phone, or should this somehow be done on the phone itself?
 
 Thanks,
 Matt

I think it is up to your phones to allow only one concurrent session,
you could check call-waiting is deactivated on your phones?!

If your phones allow more than one active dialog you probably wont have that 
much fun with queues...

And make sure you have read the Queue Empty Options section of the 
queues.conf example as some parameters changed to be more flexible
(joinempty = ringing etc... ). That could be interesting too...

hth,
Sebastian Denz
-- 
Sebastian Denz d...@gonicus.de (Senior Technical Consultant)
* GONICUS GmbH * Moehnestrasse 11-17 * D-59755 Arnsberg
* Tel.: +49 (0) 29 32 / 9 16 - 0 * Fax: +49 (0) 29 32 / 9 16  - 270
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*Amtsgericht Arnsberg * HRB 1968 

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Re: [asterisk-users] queue not skipping ringing phone

[Matt Hamilton]

Thanks Sebastian. It was a phone related issue. Factory resetting the phones 
and reconfiguring them fixed it. It probably was a CW issue as you suggested.


 I think it is up to your phones to allow only one concurrent session,
 you could check call-waiting is deactivated on your phones?!
 
 If your phones allow more than one active dialog you probably wont have that 
 much fun with queues...
 
 And make sure you have read the Queue Empty Options section of the 
 queues.conf example as some parameters changed to be more flexible
 (joinempty = ringing etc... ). That could be interesting too...
 
 hth,
 Sebastian Denz

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Re: [asterisk-users] Queue-Tip/Adhearsion installation tip

[Paul Belanger]

On 11-11-28 03:47 AM, Olivier wrote:

Hi,

I'm giving Queue-Tip a try, following installation instructions in
http://queue-tip.rubyforge.org/install.html.

My setup is :

ruby 1.8.7
rubygems 1.3.7
rails 3.1.3
Adhearsion 1.2.3

I'm struck in step 7 in the above installation procedure :

# rake --trace db:create
(in /usr/local/src/queue-tip)
rake aborted!
no such file to load -- initializer
/usr/lib/ruby/1.8/rubygems/custom_require.rb:31:in `gem_original_require'
/usr/lib/ruby/1.8/rubygems/custom_require.rb:31:in `require'
/usr/local/src/queue-tip/config/boot.rb:54:in `load_initializer'
/usr/local/src/queue-tip/config/boot.rb:38:in `run'
/usr/local/src/queue-tip/config/boot.rb:11:in `boot!'
/usr/local/src/queue-tip/config/boot.rb:109
/usr/local/src/queue-tip/Rakefile:4:in `require'
/usr/local/src/queue-tip/Rakefile:4
/usr/lib/ruby/1.8/rake.rb:2383:in `load'
/usr/lib/ruby/1.8/rake.rb:2383:in `raw_load_rakefile'
/usr/lib/ruby/1.8/rake.rb:2017:in `load_rakefile'
/usr/lib/ruby/1.8/rake.rb:2068:in `standard_exception_handling'
/usr/lib/ruby/1.8/rake.rb:2016:in `load_rakefile'
/usr/lib/ruby/1.8/rake.rb:2000:in `run'
/usr/lib/ruby/1.8/rake.rb:2068:in `standard_exception_handling'
/usr/lib/ruby/1.8/rake.rb:1998:in `run'
/usr/bin/rake:28


I'm completely new to ruby, rails and so on.
Suggestions ?


Contact the maintainer for support?  Since this is not an asterisk issue.

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Re: [asterisk-users] Queue-Tip/Adhearsion installation tip

[Olivier]

2011/11/28 Paul Belanger pabelan...@digium.com

 On 11-11-28 03:47 AM, Olivier wrote:

 Hi,

 I'm giving Queue-Tip a try, following installation instructions in
 http://queue-tip.rubyforge.**org/install.htmlhttp://queue-tip.rubyforge.org/install.html
 .

 My setup is :

 ruby 1.8.7
 rubygems 1.3.7
 rails 3.1.3
 Adhearsion 1.2.3

 I'm struck in step 7 in the above installation procedure :

 # rake --trace db:create
 (in /usr/local/src/queue-tip)
 rake aborted!
 no such file to load -- initializer
 /usr/lib/ruby/1.8/rubygems/**custom_require.rb:31:in
 `gem_original_require'
 /usr/lib/ruby/1.8/rubygems/**custom_require.rb:31:in `require'
 /usr/local/src/queue-tip/**config/boot.rb:54:in `load_initializer'
 /usr/local/src/queue-tip/**config/boot.rb:38:in `run'
 /usr/local/src/queue-tip/**config/boot.rb:11:in `boot!'
 /usr/local/src/queue-tip/**config/boot.rb:109
 /usr/local/src/queue-tip/**Rakefile:4:in `require'
 /usr/local/src/queue-tip/**Rakefile:4
 /usr/lib/ruby/1.8/rake.rb:**2383:in `load'
 /usr/lib/ruby/1.8/rake.rb:**2383:in `raw_load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2017:in `load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2068:in `standard_exception_handling'
 /usr/lib/ruby/1.8/rake.rb:**2016:in `load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2000:in `run'
 /usr/lib/ruby/1.8/rake.rb:**2068:in `standard_exception_handling'
 /usr/lib/ruby/1.8/rake.rb:**1998:in `run'
 /usr/bin/rake:28


 I'm completely new to ruby, rails and so on.
 Suggestions ?

  Contact the maintainer for support?  Since this is not an asterisk issue.


Yes, you're right (though I hoped I could get interesting answers here).


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Re: [asterisk-users] Queue-Tip/Adhearsion installation tip

[gokulnath]

Some links which might be of interest for you:
beginrescueend.com -- do go through this before installing ruby.
github.com/adhearsion/adhearsion/wiki/Getting-Started


On Mon, Nov 28, 2011 at 6:54 PM, Olivier oza_4...@yahoo.fr wrote:



 2011/11/28 Paul Belanger pabelan...@digium.com

 On 11-11-28 03:47 AM, Olivier wrote:

 Hi,

 I'm giving Queue-Tip a try, following installation instructions in
 http://queue-tip.rubyforge.**org/install.htmlhttp://queue-tip.rubyforge.org/install.html
 .

 My setup is :

 ruby 1.8.7
 rubygems 1.3.7
 rails 3.1.3
 Adhearsion 1.2.3

 I'm struck in step 7 in the above installation procedure :

 # rake --trace db:create
 (in /usr/local/src/queue-tip)
 rake aborted!
 no such file to load -- initializer
 /usr/lib/ruby/1.8/rubygems/**custom_require.rb:31:in
 `gem_original_require'
 /usr/lib/ruby/1.8/rubygems/**custom_require.rb:31:in `require'
 /usr/local/src/queue-tip/**config/boot.rb:54:in `load_initializer'
 /usr/local/src/queue-tip/**config/boot.rb:38:in `run'
 /usr/local/src/queue-tip/**config/boot.rb:11:in `boot!'
 /usr/local/src/queue-tip/**config/boot.rb:109
 /usr/local/src/queue-tip/**Rakefile:4:in `require'
 /usr/local/src/queue-tip/**Rakefile:4
 /usr/lib/ruby/1.8/rake.rb:**2383:in `load'
 /usr/lib/ruby/1.8/rake.rb:**2383:in `raw_load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2017:in `load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2068:in `standard_exception_handling'
 /usr/lib/ruby/1.8/rake.rb:**2016:in `load_rakefile'
 /usr/lib/ruby/1.8/rake.rb:**2000:in `run'
 /usr/lib/ruby/1.8/rake.rb:**2068:in `standard_exception_handling'
 /usr/lib/ruby/1.8/rake.rb:**1998:in `run'
 /usr/bin/rake:28


 I'm completely new to ruby, rails and so on.
 Suggestions ?

  Contact the maintainer for support?  Since this is not an asterisk
 issue.


 Yes, you're right (though I hoped I could get interesting answers here).


 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org

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-- 
Thanks  Regards
Gokulnath
@8129845320
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Re: [asterisk-users] queue ring delay

[Douglas Mortensen]

So I found a good description of the timeoutrestart setting here  
https://issues.asterisk.org/view.php?id=12690#87263. It definitely isn't what 
I'm looking for. So I think I may be left with two options:

1. Set Skip Busy Agents to No. (not sure how this will work with my KIRK 
phones. Currently I have call-waiting disabled on these phones, as they are not 
intuitive for handling a 2nd call while already on the phone with 1 call. So 
I'm not sure whether asterisk would continue to try to send the queue calls 
these phones (during this split second while the phones are still reporting a 
status of Ringing/BUSY), or whether it would actually send the call to the 
extension's VM (which would be even worse)..
2. Manually adjust the diaplan to introduce some delay after a ringing queue 
call is answered by an agent, but before the subsequent call ring the queue 
agents. If this becomes the solution, I may need some assistance (although I'm 
sure I'd eventually figure it out).

Again, your help is appreciated.

-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.

From: Douglas Mortensen [mailto:d...@impalanetworks.com]
Sent: Monday, November 21, 2011 9:56 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] queue ring delay

Hi,

Does a parameter exist for a queue to delay ringing/sending a caller to all 
agent phones after the previous call is answered by an agent? My queue ring 
strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. 
And it looks like the KIRK wireless server may need a split send to realize all 
wireless phones are no longer ringing (busy) after 1 call rings  is 
unanswered, prior to sending a 2nd call.

In other words, I think that what we are currently experiencing is this: 
Incoming call gets routed to our queue. It rings all phones. In the meantime, a 
2nd caller gets routed to our queue (in line behing the first caller that is 
currently ringing our phones). One queue agent answers the first phone call in 
the queue. Asterisk immediately starts ringing all queue agent extensions again 
with the 2nd caller. However, most of the agents extensions are reporting busy, 
and so their phones don't ring. The 2nd caller may wind up getting routed to 
our queue fallback destination.

So it seems to me that asterisk is sending the 2nd call to the queue agents 
before their phones are ready (i.e. before the KIRK Wireless Server is able to 
realize that they are no longer ringing [busy] from the first caller).

So I'm thinking that if I can introduce some type of delay of 500ms-1 second 
AFTER a queue call rings all phones, but before a subsequent call is permitted 
to ring all phones, my problem will be solved.

I am using FreePBX. I know this is not the place to get FreePBX support, but I 
believe that the FreePBX gui is just providing a front-end for standard 
asterisk features  parameters behind the scenes. I am on Digium AsteriskNOW 
with asterisk 1.6. I also believe that this mailing list may be the best source 
of community support for asterisk, so I am posting here. :-)

From within FreePBX, in the queue configuration, I have a parameter for 
Wrap-Up-Time and Member Delay. My questions would be:
1. Does Wrap-Up-Time apply to all queue agents/extensions that just rang, or 
only the one who actually answered the call (I assume the latter)?
2. Does the Member Delay delay the ringing of new calls to agents, or only 
come into play AFTER the agent answers the ringing call?

Any other suggestions for how I can resolve this issue? I am wondering whether 
Agent Timeout or Agent Timeout Restart (or a combination of both) may be 
able to help me here. It sounds like the 2nd option may help me. But I'm not 
familiar with exactly how it would work in this situation.

Anyway, that's it. As for some background, we initially were using ring groups, 
but realized that these phones do NOT have the ability to handle a 2nd ringing 
call. So in the event that 2 inbound calls rang within a few seconds, asterisk 
would send the first to all phones, and then when tyring to send the 2nd, would 
receive a BUSY message from the phones (because they were busy processing a 
ring for the first caller), and the 2nd caller would wind up going straight to 
the unavilable destination for the ring group, instead of eventually ringing 
through to the phones after someone answered the first call.

I greatly appreciate your help  insight with this issue!
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.comhttp://www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545
.

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Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2

[Alex Kauffmann]

On 30/10/2011 05:53 a.m., Raj Mathur (राज माथुर) wrote:

On Sunday 30 Oct 2011, Sammy Govind wrote:

hmmm so  IAX channel is playing with you guys.

1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of the command core show file versions in your
both asterisk servers. Mainly lookout for IAX channel version.

Also try enabling IAX debug and paste the output on console.

1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial
server.

I doubt if we'll be able to change the architecture of an infrastructure
handling up to 450 simultaneous calls for the past 6 months at this
stage, so SIP is out.  IAX2 has been working beautifully for our needs
up to this point, and we need to find a solution that we can integrate
into this architecture itself!

Incidentally, if anyone's interested, the installation itself is
detailed at:

http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html

Regards,

-- Raj
Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are 
running out of bandwidth in your IAX2 trunk. The setting 'trunkmaxsize' 
defaults to 128000 bytes.


From the readme file:

...Once this limit is
; reached, calls may be dropped or begin to lose audio.  Depending on the codec 
in use and
; number of channels to be supported this value may need to be raised, but in 
most cases the
; default value is large enough.


--
Alex

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