[asterisk-users] Queues - how to add back a agent without all other calls to agents stoping and re-starting

[Paddy Grice]

Hi All

 

I have a problem with queues that I have been trying to solve for many
months - the customer has now picked back up onto this and wanting a
solution - any guidance, ideas or solutions welcome.

 

This is the situation :-

 

We have a number of agents in a ringall group, a call joins the queue and
all handsets ring - great. 

An agent answers and handles the call - great.

 

The call is cleared but the agents handset is not re-rung (for the next
call) until timeout seconds elapse at that time all other agents handsets
stop ringing for a moment and then start again.

 

As I say any help or guidance appreciated

 

Paddy

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[asterisk-users] Queues, Penalties, RINGALL, and Use Cases

[Ben Ford]

Hey everyone,

I posted a topic on the community forum which can be found here:
https://community.asterisk.org/t/request-for-feedback-queues-penalties-ringall-and-use-cases/80960

It covers a scenario encountered while debugging an issue, and I would love
some feedback on it to help make a decision on how to fix it moving forward.

Thanks!

-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US

Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] Queues and penalties

[Steve Davies]

I'm fairly sure the patch to App Queue that was added to Asterisk 13+
should do the job... It causes agent priorities to "float up" over time so
that new agents are included without excluding old agents.

I can't find it right now but there can't be that many app_queue patches to
ast 13 in the last 18 months

Steve


On Fri, 30 Nov 2018 at 09:18, Paddy Grice  wrote:

> Thanks Leon
>
> I will implement and test but I knew there would be a fix for what I
> believe is a short coming in app_queue. How do I suggest this as a option
> to the base code?
>
> Paddy
>
> --
> *From:* Leon Wright [mailto:lwri...@corpcloud.com.au]
> *Sent:* 30 November 2018 02:17
> *To:* pa...@wizaner.com; asterisk-users@lists.digium.com
> *Cc:* johnkinis...@gmail.com
> *Subject:* Re: [asterisk-users] Queues and penalties
>
> Paddy,
>
> This appears to be how the queue app works. I ended up patching the queue
> app:
>
> diff --git a/apps/app_queue.c b/apps/app_queue.c
> index e3a4e22..72072d0 100644
> --- a/apps/app_queue.c
> +++ b/apps/app_queue.c
> @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct
> callattempt *outgoing, int *bus
> struct callattempt *cur;
> /* Ring everyone who shares this best metric (for
> ringall) */
> for (cur = outgoing; cur; cur = cur->q_next) {
> -   if (cur->stillgoing && !cur->chan &&
> cur->metric <= best->metric) {
> +   if (cur->stillgoing && !cur->chan &&
> cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty
> * 100) {
> ast_debug(1, "(Parallel) Trying
> '%s' with metric %d\n", cur->interface, cur->metric);
> ret |= ring_entry(qe, cur, busies);
> }
>
> So the penalties get calculated during the 'ringall' strategy and allowing
> the queue app to exit, looping and raising the max penalty and calling the
> queue app again.
>
> Leon
>
> On Thu, 29 Nov 2018 at 18:24, Paddy Grice  wrote:
>
>> Hi John
>>
>> This works fine providing extensions 1001,1002 and 1003 are "Incall" or
>> "Paused" - the problem appears to be that is a handset say 1002 is
>> "ringing" then the 2xxx then the penalty is not honoured.
>>
>> This is well described in the History section of the following link
>> https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
>>
>> As I say this seems to be a real shortcoming in app_queue.
>>
>> Any ideas, suggestions, anyone want to work with me to sort this ?
>>
>> Paddy
>>
>>
>> --
>> *From:* John Kiniston [mailto:johnkinis...@gmail.com]
>> *Sent:* 28 November 2018 21:17
>> *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> *Subject:* Re: [asterisk-users] Queues and penalties
>>
>> This should work, How are you defining your timeouts in the queues.conf ?
>>
>> And to verify, in your extensions.conf you are calling Queue with the
>> queue name and the ruleset to apply from queuerules.conf?
>>
>> On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:
>>
>>> Hi All
>>>
>>> I have been looking at this problem for a few days/weeks now and after
>>> some advice please.
>>>
>>> I currently have a customer on 11.25.3 and I am in the process of
>>> upgrading versions and OS (Debian) and all things that involves mysql ->
>>> PDO etc
>>>
>>> The problem I have is the customer want a simple call distribution like
>>> this
>>>
>>> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
>>> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
>>> extensions and if no one answers after another 20 seconds the add in 3001,
>>> 3002, 3003.
>>>
>>> Seems a simple queue application to me
>>>
>>> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
>>> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
>>> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
>>>
>>> and rules
>>>
>>> increasing the maxpenalty 1->2 after 20 seconds
>>> and increasing maxpenalty 2->3 after another 20 seconds.
>>>
>>> 

Re: [asterisk-users] Queues and penalties

[Paddy Grice]

Thanks Leon 
 
I will implement and test but I knew there would be a fix for what I believe
is a short coming in app_queue. How do I suggest this as a option to the
base code?
 
Paddy

  _  

From: Leon Wright [mailto:lwri...@corpcloud.com.au] 
Sent: 30 November 2018 02:17
To: pa...@wizaner.com; asterisk-users@lists.digium.com
Cc: johnkinis...@gmail.com
Subject: Re: [asterisk-users] Queues and penalties


Paddy, 

This appears to be how the queue app works. I ended up patching the queue
app:

diff --git a/apps/app_queue.c b/apps/app_queue.c
index e3a4e22..72072d0 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct
callattempt *outgoing, int *bus
struct callattempt *cur;
/* Ring everyone who shares this best metric (for
ringall) */
for (cur = outgoing; cur; cur = cur->q_next) {
-   if (cur->stillgoing && !cur->chan &&
cur->metric <= best->metric) {
+   if (cur->stillgoing && !cur->chan &&
cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty *
100) {
ast_debug(1, "(Parallel) Trying '%s'
with metric %d\n", cur->interface, cur->metric);
ret |= ring_entry(qe, cur, busies);
}

So the penalties get calculated during the 'ringall' strategy and allowing
the queue app to exit, looping and raising the max penalty and calling the
queue app again.

Leon

On Thu, 29 Nov 2018 at 18:24, Paddy Grice  wrote:



Hi John
 
This works fine providing extensions 1001,1002 and 1003 are "Incall" or
"Paused" - the problem appears to be that is a handset say 1002 is "ringing"
then the 2xxx then the penalty is not honoured.
 
This is well described in the History section of the following link
https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
 
As I say this seems to be a real shortcoming in app_queue. 
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy
 
 
  _  

From: John Kiniston [mailto:johnkinis...@gmail.com] 
Sent: 28 November 2018 21:17
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Queues and penalties


This should work, How are you defining your timeouts in the queues.conf ? 


And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf? 


On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:



Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall

2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall

3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 

 
 
 
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Re: [asterisk-users] Queues and penalties

[Leon Wright]

Paddy,

This appears to be how the queue app works. I ended up patching the queue
app:

diff --git a/apps/app_queue.c b/apps/app_queue.c
index e3a4e22..72072d0 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct
callattempt *outgoing, int *bus
struct callattempt *cur;
/* Ring everyone who shares this best metric (for
ringall) */
for (cur = outgoing; cur; cur = cur->q_next) {
-   if (cur->stillgoing && !cur->chan &&
cur->metric <= best->metric) {
+   if (cur->stillgoing && !cur->chan &&
cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty
* 100) {
ast_debug(1, "(Parallel) Trying
'%s' with metric %d\n", cur->interface, cur->metric);
ret |= ring_entry(qe, cur, busies);
}

So the penalties get calculated during the 'ringall' strategy and allowing
the queue app to exit, looping and raising the max penalty and calling the
queue app again.

Leon

On Thu, 29 Nov 2018 at 18:24, Paddy Grice  wrote:

> Hi John
>
> This works fine providing extensions 1001,1002 and 1003 are "Incall" or
> "Paused" - the problem appears to be that is a handset say 1002 is
> "ringing" then the 2xxx then the penalty is not honoured.
>
> This is well described in the History section of the following link
> https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
>
> As I say this seems to be a real shortcoming in app_queue.
>
> Any ideas, suggestions, anyone want to work with me to sort this ?
>
> Paddy
>
>
> --
> *From:* John Kiniston [mailto:johnkinis...@gmail.com]
> *Sent:* 28 November 2018 21:17
> *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
> Discussion
> *Subject:* Re: [asterisk-users] Queues and penalties
>
> This should work, How are you defining your timeouts in the queues.conf ?
>
> And to verify, in your extensions.conf you are calling Queue with the
> queue name and the ruleset to apply from queuerules.conf?
>
> On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:
>
>> Hi All
>>
>> I have been looking at this problem for a few days/weeks now and after
>> some advice please.
>>
>> I currently have a customer on 11.25.3 and I am in the process of
>> upgrading versions and OS (Debian) and all things that involves mysql ->
>> PDO etc
>>
>> The problem I have is the customer want a simple call distribution like
>> this
>>
>> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
>> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
>> extensions and if no one answers after another 20 seconds the add in 3001,
>> 3002, 3003.
>>
>> Seems a simple queue application to me
>>
>> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
>> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
>> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
>>
>> and rules
>>
>> increasing the maxpenalty 1->2 after 20 seconds
>> and increasing maxpenalty 2->3 after another 20 seconds.
>>
>> But this doesn't work if users don't answer!!
>>
>> if user 1002 or (2001 etc)  just lets his phone ring - he forgot to
>> logoff or DND then the penalty is ignored.
>>
>> There seems to have been a patch for FreePBX on V13 - LazyMembers - but
>> that is all I can find and later versions have no mention of this
>>
>> I guess I can use autopause and some AMI / Script but this stops phones
>> ringing because of the timeout so the user has a ringing phone and then it
>> stops and then it starts again whereas the penalty just adds handsets into
>> the ringing group.
>>
>> This seems to be a real shortcoming in app_queue.
>>
>> Any ideas, suggestions, anyone want to work with me to sort this ?
>>
>> Paddy Grice
>>
>>
>>
>>
>>
>>
>>
>>
>> --
>> _
>> -- 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.or

Re: [asterisk-users] Queues and penalties

[Paddy Grice]

Hi John
 
This works fine providing extensions 1001,1002 and 1003 are "Incall" or
"Paused" - the problem appears to be that is a handset say 1002 is "ringing"
then the 2xxx then the penalty is not honoured.
 
This is well described in the History section of the following link
https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
 
As I say this seems to be a real shortcoming in app_queue. 
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy
 
 
  _  

From: John Kiniston [mailto:johnkinis...@gmail.com] 
Sent: 28 November 2018 21:17
To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Queues and penalties


This should work, How are you defining your timeouts in the queues.conf ? 


And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf? 


On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:



Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall

2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall

3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 

 
 
 
-- 
_
-- 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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-- 

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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Re: [asterisk-users] Queues and penalties

[John Kiniston]

This should work, How are you defining your timeouts in the queues.conf ?

And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf?

On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:

> Hi All
>
> I have been looking at this problem for a few days/weeks now and after
> some advice please.
>
> I currently have a customer on 11.25.3 and I am in the process of
> upgrading versions and OS (Debian) and all things that involves mysql ->
> PDO etc
>
> The problem I have is the customer want a simple call distribution like
> this
>
> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
> extensions and if no one answers after another 20 seconds the add in 3001,
> 3002, 3003.
>
> Seems a simple queue application to me
>
> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
>
> and rules
>
> increasing the maxpenalty 1->2 after 20 seconds
> and increasing maxpenalty 2->3 after another 20 seconds.
>
> But this doesn't work if users don't answer!!
>
> if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
> or DND then the penalty is ignored.
>
> There seems to have been a patch for FreePBX on V13 - LazyMembers - but
> that is all I can find and later versions have no mention of this
>
> I guess I can use autopause and some AMI / Script but this stops phones
> ringing because of the timeout so the user has a ringing phone and then it
> stops and then it starts again whereas the penalty just adds handsets into
> the ringing group.
>
> This seems to be a real shortcoming in app_queue.
>
> Any ideas, suggestions, anyone want to work with me to sort this ?
>
> Paddy Grice
>
>
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Queues and penalties

[Paddy Grice]

Hi All 
 
I have been looking at this problem for a few days/weeks now and after some
advice please.
 
I currently have a customer on 11.25.3 and I am in the process of upgrading
versions and OS (Debian) and all things that involves mysql -> PDO etc
 
The problem I have is the customer want a simple call distribution like this
 
Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
extensions and if no one answers after another 20 seconds the add in 3001,
3002, 3003.
 
Seems a simple queue application to me
 
1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
 
and rules 
 
increasing the maxpenalty 1->2 after 20 seconds
and increasing maxpenalty 2->3 after another 20 seconds.
 
But this doesn't work if users don't answer!!
 
if user 1002 or (2001 etc)  just lets his phone ring - he forgot to logoff
or DND then the penalty is ignored.
 
There seems to have been a patch for FreePBX on V13 - LazyMembers - but that
is all I can find and later versions have no mention of this 
 
I guess I can use autopause and some AMI / Script but this stops phones
ringing because of the timeout so the user has a ringing phone and then it
stops and then it starts again whereas the penalty just adds handsets into
the ringing group.
 
This seems to be a real shortcoming in app_queue.
 
Any ideas, suggestions, anyone want to work with me to sort this ?
 
Paddy Grice
 
 
 
 
 
 
 
 
-- 
_
-- 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
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[asterisk-users] Queues - different moh for queue waiting and subsequent onhold

[Mike]

Hello,

Quick one here: is it possible to setup my dialplan on such a way that the
MoH while waiting to be answered by an agent on a Queue be different than
the one that I will hear when that agent puts me manually on hold for a
few minutes AFTER I finally got a hold of someone?

I seem to recall that setting up a musicclass in the dialplan was what was
used for onhold MoH, while the "music" field of the Queue was the "queue
waiting" MoH.  But that as back on 1.8, I am on Asterisk 13 right now, and
the "music" field of the Queue seems to overwrite the dialplan MoH
definition.

Mike





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Re: [asterisk-users] Queues - periodic announce while ringing members

[Tiago Geada]

Hi, what I did, I mixed the music on hold to have the announce in at a
specific time without leaving queue

On 25 February 2016 at 16:53, Daniel Chavez  wrote:

> Ish,
> I use the same version of Asterisk on CentOS 6.7. I wonder the same thing.
> Hopefully we will find this out.
>
>
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Re: [asterisk-users] Queues - periodic announce while ringing members

[Daniel Chavez]

Ish,
I use the same version of Asterisk on CentOS 6.7. I wonder the same thing. 
Hopefully we will find this out.


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[asterisk-users] Queues - periodic announce while ringing members

[Ishfaq Malik]

Hi

I'm using asterisk 1.8.32.3 on CentOS 6

I've noticed when using queues that the members of the queue stop ringing
for the duration of any set periodic announce. Is this the only behaviour
possible or is there a way to set the members to continue ringing while the
periodic announce plays?

Thanks in Advance

Ish

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[John Kiniston]

Wow, Looks like they have really increased the options since I last looked.

I just pulled down the Asterisk 13 queues.conf.sample and it's got this in
it:

; paused: a member is not considered available if he is paused
; penalty: a member is not considered available if his penalty is less than
QUEUE_MAX_PENALTY
; inuse: a member is not considered available if he is currently on a call
; ringing: a member is not considered available if his phone is currently
ringing
; unavailable: This applies mainly to Agent channels. If the agent is a
member of the queue
;  but has not logged in, then do not consider the member to be
available
; invalid: Do not consider a member to be available if he has an invalid
device state.
;  This generally is caused by an error condition in the member's
channel driver.
; unknown: Do not consider a member to be available if we are unable to
determine the member's
;  current device state.
; wrapup: A member is not considered available if he is currently in his
wrapuptime after
; taking a call.

An unknown state would be a device that has a valid configuration but isn't
registered.

On Tue, Jul 28, 2015 at 8:51 PM, Andrew Martin amar...@xes-inc.com wrote:

 - Original Message -
  From: John Kiniston johnkinis...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
  Sent: Tuesday, July 28, 2015 12:12:05 PM
  Subject: Re: [asterisk-users] Queues don't follow dialplan if no members
 are  registered
 
  In your queues.conf do you have a leavewhenempty and joinempty set?
 
  in queues.conf
  [myqueue]
  leavewhenempty = strict
  joinempty = strict
  strategy = ringall
  ringinuse = no
 
 

 John,

 Thanks for the fast reply! I had joinempty=yes in queues.conf,
 which explains why I was seeing this behavior. It looks like the
 strict setting is partially-deprecated, so instead I'm using
 the following combination:

 [myqueue]
 musiconhold=default
 music=default
 strategy=ringall
 joinempty=unavailable,invalid,unknown
 leavewhenempty=unavailable,invalid,unknown
 timeout=18

 member = SIP/100
 member = SIP/101

 Is there any reason that using any of these options would be a
 problem, in particular unknown? It is not very well defined
 what an unknown state is exactly.

 Thanks,

 Andrew

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

- Original Message -
 From: John Kiniston johnkinis...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Wednesday, July 29, 2015 11:53:13 AM
 Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are  
 registered
 
 Wow, Looks like they have really increased the options since I last looked.
 
 I just pulled down the Asterisk 13 queues.conf.sample and it's got this in
 it:
 
 ; paused: a member is not considered available if he is paused
 ; penalty: a member is not considered available if his penalty is less than
 QUEUE_MAX_PENALTY
 ; inuse: a member is not considered available if he is currently on a call
 ; ringing: a member is not considered available if his phone is currently
 ringing
 ; unavailable: This applies mainly to Agent channels. If the agent is a
 member of the queue
 ; but has not logged in, then do not consider the member to be available
 ; invalid: Do not consider a member to be available if he has an invalid
 device state.
 ; This generally is caused by an error condition in the member's channel
 driver.
 ; unknown: Do not consider a member to be available if we are unable to
 determine the member's
 ; current device state.
 ; wrapup: A member is not considered available if he is currently in his
 wrapuptime after
 ; taking a call.
 
 An unknown state would be a device that has a valid configuration but isn't
 registered.
 
John,

Thanks for the clarification and your help resolving this issue!

Andrew

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

- Original Message -
 From: John Kiniston johnkinis...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Sent: Tuesday, July 28, 2015 12:12:05 PM
 Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are  
 registered
 
 In your queues.conf do you have a leavewhenempty and joinempty set?
 
 in queues.conf
 [myqueue]
 leavewhenempty = strict
 joinempty = strict
 strategy = ringall
 ringinuse = no
 
 

John,

Thanks for the fast reply! I had joinempty=yes in queues.conf,
which explains why I was seeing this behavior. It looks like the
strict setting is partially-deprecated, so instead I'm using
the following combination:

[myqueue]
musiconhold=default
music=default
strategy=ringall
joinempty=unavailable,invalid,unknown
leavewhenempty=unavailable,invalid,unknown
timeout=18

member = SIP/100
member = SIP/101

Is there any reason that using any of these options would be a
problem, in particular unknown? It is not very well defined
what an unknown state is exactly.

Thanks,

Andrew

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[asterisk-users] Queues don't follow dialplan if no members are registered

[Andrew Martin]

Hello,

I am running Asterisk 11 on CentOS 6.x. I have configured several queues as 
follows in extensions.conf:
exten = s,1,Queue(myqueue,rtnC,18)
same = n,Background(user_unavail)
same = n,WaitExten(10)
exten = 1,1,Voicemail(@my-vm,s)

This rings the phones in the queue for 18 seconds. If no queue members answer,
the caller is then prompted to press 1 and leave a voicemail. This works well
when at least 1 member is registered in the queue, however if no members are
registered in the queue, the Queue() call never seems to return, and thus the
remaining steps in the dialplan never execute. How can I correct this behavior
so that even if the queue has no registered members, the dialplan is still
followed?

Thanks,

Andrew

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Re: [asterisk-users] Queues don't follow dialplan if no members are registered

[John Kiniston]

In your queues.conf do you have a leavewhenempty and joinempty set?

in queues.conf
[myqueue]
leavewhenempty = strict
joinempty = strict
strategy = ringall
ringinuse = no


On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin amar...@xes-inc.com wrote:

 Hello,

 I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
 follows in extensions.conf:
 exten = s,1,Queue(myqueue,rtnC,18)
 same = n,Background(user_unavail)
 same = n,WaitExten(10)
 exten = 1,1,Voicemail(@my-vm,s)

 This rings the phones in the queue for 18 seconds. If no queue members
 answer,
 the caller is then prompted to press 1 and leave a voicemail. This works
 well
 when at least 1 member is registered in the queue, however if no members
 are
 registered in the queue, the Queue() call never seems to return, and thus
 the
 remaining steps in the dialplan never execute. How can I correct this
 behavior
 so that even if the queue has no registered members, the dialplan is still
 followed?

 Thanks,

 Andrew

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cooperate, act alone, solve equations, analyze a new problem, pitch manure,
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[asterisk-users] Queues and RingInUse

[Magnus Löfqvist]

Hi,

I have some problems with a Queue and state interface.

In the queue we have 7 deskphones and 7 cordless.
Every user have one deskphone and one cordless phone.

I have set up every user with hints, eg:

exten = 802,hint,SIP/9144-802SIP/9144-902
exten = 902,hint,SIP/9144-902SIP/9144-802

Queue_members:
Queue_name Interface
state_interface
9144-vmi   SIP/9144-802 
hint:802@hints
9144-vmi   SIP/9144-902 
hint:902@hints

802 = is a cordless phone
902 = is a deskphone
Both are in use by same user.

To the problem, when we have RingInUse = 0, we only get calls on one of the 
devices (because the other one gets in RINGING state).
If the user is not on phone we want both phones to call, but if the user is in 
phone the other phone should not call.

Any idées...


Med vänlig hälsning / Best regards
Magnus Löfqvist

VMI Internet Services AB
Hantverksvägen 15
760 40 VÄDDÖ, SWEDEN
Tel +46 176 20 89 00 (02)
Fax +46 176 20 89 19
E-mail: m...@vmi.se
Homepage: www.vmi.sehttp://www.vmi.se/
Facebook: www.vmi.se/facebookhttp://www.vmi.se/facebook

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It is intended solely for the addressee. Access to this email by anyone else is 
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distribution or any action taken or omitted to be taken in reliance on it, is 
prohibited and may be unlawful. Any opinions or advice contained in this e-mail 
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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Lenz Emilitri]

You need to do this when the call connects. If you can do this within a
couple of seconds, this is usually good enough to be usable (that's what
we do on the QueueMetrics agents pages).
Thanks
l.



2013/8/3 Timothy Smith timotsm...@gmail.com

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

Our queue members are Local channels, thus when dialing the agent, the
dialplan will do several stuff including:

Set(CALLERID(name)=${CALLERID(name)}:Sales)
UserEvent(something,data: ${bunch-of-data-in-some-format})
Dial(SIP/final-agent-phone,timeout,A(Sales))

The UserEvent will be picked up by our client-register-ticket-stuff software

The announcement A() will be heard by the agent upon answering the call
like sales call


On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:

 We do something very similar.

 Use the gosub parameter of the Queue application to call a subroutine in
 the dial plan when the agent answers the call.

 same =n,Queue(sales,tc,,sub-**QueueConnected)

 [sub-QueueConnected]
 ; this runs on the agent/member's channel
 exten =s,1,NoOp()
   ; whatever you need to do here
   same =n,Return()

 See https://wiki.asterisk.org/**wiki/display/AST/Asterisk+11+**
 Application_Queuehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue


 Mitch


 On 08/03/2013 12:45 PM, Timothy Smith wrote:

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Timothy Smith]

Dear Mitch,

Thank you so much. This partly solves my problem by a great deal, as
we'll send a message to the agent immediately on picking the call. As
the agents are local SIP channels, I will attempt looking up the
caller's name (if it exists in our database) and set it prior to
entering the queue.

Is there any way of informing the agent (just) before they pick up?
e.g when their phone starts ringing, so that they prepare accordingly?

Regards,
Wilson

On Sun, Aug 4, 2013 at 4:59 AM, Mitch Claborn mitch...@claborn.net wrote:
 We do something very similar.

 Use the gosub parameter of the Queue application to call a subroutine in the
 dial plan when the agent answers the call.

 same =n,Queue(sales,tc,,sub-QueueConnected)

 [sub-QueueConnected]
 ; this runs on the agent/member's channel
 exten =s,1,NoOp()
   ; whatever you need to do here
   same =n,Return()

 See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue


 Mitch


 On 08/03/2013 12:45 PM, Timothy Smith wrote:

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Timothy Smith]

Dear Tiago,

Thanks for your answer, but I have a few questions.

Do you use queues? We are operating a call centre with several queues,
so I don't see how we would use the Dial command. When a call comes
in, we enter the caller (depending on what options he has selected)
into a queue. Do you have any alternative method, which would involve
dialling the agent directly as you described below?

regards,
T

On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote:
 Hi,

 Our queue members are Local channels, thus when dialing the agent, the
 dialplan will do several stuff including:

 Set(CALLERID(name)=${CALLERID(name)}:Sales)
 UserEvent(something,data: ${bunch-of-data-in-some-format})
 Dial(SIP/final-agent-phone,timeout,A(Sales))

 The UserEvent will be picked up by our client-register-ticket-stuff software

 The announcement A() will be heard by the agent upon answering the call like
 sales call


 On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:

 We do something very similar.

 Use the gosub parameter of the Queue application to call a subroutine in
 the dial plan when the agent answers the call.

 same =n,Queue(sales,tc,,sub-QueueConnected)

 [sub-QueueConnected]
 ; this runs on the agent/member's channel
 exten =s,1,NoOp()
   ; whatever you need to do here
   same =n,Return()

 See
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue


 Mitch


 On 08/03/2013 12:45 PM, Timothy Smith wrote:

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

You just said you use Local channels. Local channel is a dialplan that has
a Dial() to a sip device?

We use queues, and have a queue-macro that sends the UserEvent upon
bridging the call...


On 4 August 2013 16:41, Timothy Smith timotsm...@gmail.com wrote:

 Dear Tiago,

 Thanks for your answer, but I have a few questions.

 Do you use queues? We are operating a call centre with several queues,
 so I don't see how we would use the Dial command. When a call comes
 in, we enter the caller (depending on what options he has selected)
 into a queue. Do you have any alternative method, which would involve
 dialling the agent directly as you described below?

 regards,
 T

 On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote:
  Hi,
 
  Our queue members are Local channels, thus when dialing the agent, the
  dialplan will do several stuff including:
 
  Set(CALLERID(name)=${CALLERID(name)}:Sales)
  UserEvent(something,data: ${bunch-of-data-in-some-format})
  Dial(SIP/final-agent-phone,timeout,A(Sales))
 
  The UserEvent will be picked up by our client-register-ticket-stuff
 software
 
  The announcement A() will be heard by the agent upon answering the call
 like
  sales call
 
 
  On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:
 
  We do something very similar.
 
  Use the gosub parameter of the Queue application to call a subroutine in
  the dial plan when the agent answers the call.
 
  same =n,Queue(sales,tc,,sub-QueueConnected)
 
  [sub-QueueConnected]
  ; this runs on the agent/member's channel
  exten =s,1,NoOp()
; whatever you need to do here
same =n,Return()
 
  See
 
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
 
 
  Mitch
 
 
  On 08/03/2013 12:45 PM, Timothy Smith wrote:
 
  Hello Folks,
 
  I am setting up a call center but we have few agents so one agent is
  able to handle calls of different languages and different queues. For
  the agent to identify the caller, I want a popup to appear as the
  phone starts to ring with the caller's number, language (selected in
  the IVR), Queue (sales, support etc) and any other information (e.g a
  URL with parameters)
 
  I can send this information either via netcat (to a client such as
  yac) to a Windows PC but the problem is I do not know when the caller
  is about to be connected to the agent, so that I run the command. If I
  wasn't using queues, it would be easy because  I would run the netcat
  command and then dial the user's extension.
 
  My Question is: Is there a way I can know when the caller is just
  about to be connected to an agent (when the agent's SIP extension
  starts ringing)?
 
  There are these settings setinterfacevar, setqueueentryvar,
  setqueuevar in queues.conf but when can I use them?
 
  Have you guys been in this situation before? Any alternative solutions
  (sending caller info to an agent)?
 
  I am using Asterisk 11 and Windows 7 PCs for agents.
 
  Thank you!
 
  Kind Regards,
  Wilson
 
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[asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Timothy Smith]

Hello Folks,

I am setting up a call center but we have few agents so one agent is
able to handle calls of different languages and different queues. For
the agent to identify the caller, I want a popup to appear as the
phone starts to ring with the caller's number, language (selected in
the IVR), Queue (sales, support etc) and any other information (e.g a
URL with parameters)

I can send this information either via netcat (to a client such as
yac) to a Windows PC but the problem is I do not know when the caller
is about to be connected to the agent, so that I run the command. If I
wasn't using queues, it would be easy because  I would run the netcat
command and then dial the user's extension.

My Question is: Is there a way I can know when the caller is just
about to be connected to an agent (when the agent's SIP extension
starts ringing)?

There are these settings setinterfacevar, setqueueentryvar,
setqueuevar in queues.conf but when can I use them?

Have you guys been in this situation before? Any alternative solutions
(sending caller info to an agent)?

I am using Asterisk 11 and Windows 7 PCs for agents.

Thank you!

Kind Regards,
Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Mitch Claborn]

We do something very similar.

Use the gosub parameter of the Queue application to call a subroutine in 
the dial plan when the agent answers the call.


same =n,Queue(sales,tc,,sub-QueueConnected)

[sub-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
  ; whatever you need to do here
  same =n,Return()

See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue


Mitch

On 08/03/2013 12:45 PM, Timothy Smith wrote:

Hello Folks,

I am setting up a call center but we have few agents so one agent is
able to handle calls of different languages and different queues. For
the agent to identify the caller, I want a popup to appear as the
phone starts to ring with the caller's number, language (selected in
the IVR), Queue (sales, support etc) and any other information (e.g a
URL with parameters)

I can send this information either via netcat (to a client such as
yac) to a Windows PC but the problem is I do not know when the caller
is about to be connected to the agent, so that I run the command. If I
wasn't using queues, it would be easy because  I would run the netcat
command and then dial the user's extension.

My Question is: Is there a way I can know when the caller is just
about to be connected to an agent (when the agent's SIP extension
starts ringing)?

There are these settings setinterfacevar, setqueueentryvar,
setqueuevar in queues.conf but when can I use them?

Have you guys been in this situation before? Any alternative solutions
(sending caller info to an agent)?

I am using Asterisk 11 and Windows 7 PCs for agents.

Thank you!

Kind Regards,
Wilson

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[asterisk-users] Queues with different technologies for members, like Khomp Driver

[Daniel Varella]

Guys,

I saw in the Asterisk documentation (queues.conf) that members can
register with technologies such as SIP, Dahdi and Location.
But I have a specific need for members to be registered as Khompchannel.

Ex: member = Khomp/b0L1/9200

But reloading module app_queue.so when I run the command queue show, the
member registered as Khomp appears as invalid:

   Khomp/b0L1/9200 (Invalid)

I also read that I can try to load the driver used by members (in this
case the Khomp driver) before app_queue.so to the Queue application can
recognize that as a valid driver for a member.
But I tried to do this with the preload = chan_khomp.so in
modules.conf, but even then the Queue application continues to recognize the
member as invalid.

Has anyone tried to do something like this, using other
technologies to members
of queues?

   PS.: Khomp is a kind of Channel Bank for E1 connections here in Brazil.

Thanks in advance.

--

Daniel Varella de Oliveira
Consultor de T.I.
Cel.: +55(21)8615-6050

Digium Certified Asterisk Administrator - (dCAA)

Novell Certified Linux Administrator (Novell CLA) 
Novell Data Center Technical Specialist (Novell DCTS)
SUSE Linux Enterprise 11

Linux Professional Certified - LPI

Information Technology Infrastructure Library - ITIL Certified

Cisco Certified Network Associate - CCNA
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Re: [asterisk-users] Queues and distributed device state over WAN

[Alec Davis]

   Oliver wrote:
snip
   Before diving into this, I've got the following question :
   
   - let say we have two Asterisk servers A and B,
   - both are interconnected through PSTN (no SIP trunk)
   - agent Alice's phone is connected (ie registered) to server A
   - Alice's phone can be reached from server A phones dialing
Local/6789
   - Alice's phone can also be reached from server B phones dialing
Local/00123456789
   
   
   1. How do you configure both servers so that Alice's phone becomes a
Queue Member from a server B given queue ?
   Simply calling AddQueueMember on server B, passing Local/00123456789
as interface value (ie AddQueueMember second argument) ?
   
   2. Then, how should server A publish Alice's phone status ?
   How should server B consume Alice's phone status and associate it
with the Queue member activity ? Using AddQueueMember stateinterface
argument ?


Before trying to get distributed device state going:
Device names across all Asterisk proxy's participating in
'distributed device state' need to be unique.
IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and
SIP/cisco1 on server B for BOB.

You need to get XMPP distributed device state working.
I followed
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP
P+PubSub

You need a reasonbly reliable WAN links to the jabber server support the
XMPP updates between servers.
Asterisk segfaults if it can't contact the jabber server!!! See
https://issues.asterisk.org/jira/browse/ASTERISK-18078

Then:
With Alice reqistered on Server A as SIP/cisco1
With Server B hosting the queue named 'queue1'.

;(on Server B)
queues.conf:
[queue1]
;what makes this work with distributed states is the 'State Interface'
parameter 
...
member = Local/00123456789,0,ALICE,SIP/cisco1

Alice will need a number to ring to login/logout of queue1 hosted on Server
B;
Dialplan Example: on server B:
...
exten = s,n,Set(queuename=queue1)
exten = s,n,Set(interface=Local/00123456789)
exten = s,n,Set(penalty=0)
exten = s,n,Set(stateinterface=SIP/cisco1)
exten =
s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,,
${stateinterface})

And to remove the member;
...
exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface})

Alec Davis





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Re: [asterisk-users] Queues and distributed device state over WAN

[Olivier]

2013/1/25 Alec Davis siva...@paradise.net.nz

Oliver wrote:
 snip
Before diving into this, I've got the following question :
 
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
- Alice's phone can be reached from server A phones dialing
 Local/6789
- Alice's phone can also be reached from server B phones dialing
 Local/00123456789
 
 
1. How do you configure both servers so that Alice's phone becomes
 a
 Queue Member from a server B given queue ?
Simply calling AddQueueMember on server B, passing
 Local/00123456789
 as interface value (ie AddQueueMember second argument) ?
 
2. Then, how should server A publish Alice's phone status ?
How should server B consume Alice's phone status and associate it
 with the Queue member activity ? Using AddQueueMember stateinterface
 argument ?


 Before trying to get distributed device state going:
 Device names across all Asterisk proxy's participating in
 'distributed device state' need to be unique.
 IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and
 SIP/cisco1 on server B for BOB.

 You need to get XMPP distributed device state working.
 I followed

 https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP
 P+PubSub

 You need a reasonbly reliable WAN links to the jabber server support the
 XMPP updates between servers.
 Asterisk segfaults if it can't contact the jabber server!!! See
 https://issues.asterisk.org/jira/browse/ASTERISK-18078

 Then:
 With Alice reqistered on Server A as SIP/cisco1
 With Server B hosting the queue named 'queue1'.

 ;(on Server B)
 queues.conf:
 [queue1]
 ;what makes this work with distributed states is the 'State Interface'
 parameter
 ...
 member = Local/00123456789,0,ALICE,SIP/cisco1

 Alice will need a number to ring to login/logout of queue1 hosted on Server
 B;
 Dialplan Example: on server B:
 ...
 exten = s,n,Set(queuename=queue1)
 exten = s,n,Set(interface=Local/00123456789)
 exten = s,n,Set(penalty=0)
 exten = s,n,Set(stateinterface=SIP/cisco1)
 exten =

 s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,,
 ${stateinterface})

 And to remove the member;
 ...
 exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface})

 Alec Davis



I've not tried to publish device state with XMPP yet but I've discovered
this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as one asterisk
server so hopefully, I won't be hit by the issue above but have you met
this issue ?
Could you get around ?
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Re: [asterisk-users] Queues and distributed device state over WAN

[Alec Davis]

 I've not tried to publish device state with XMPP yet but I've 
 discovered this issue 
 https://issues.asterisk.org/jira/browse/ASTERISK-18078
 I'm planning to install my XMPP server on the same machine as 
 one asterisk server so hopefully, I won't be hit by the issue 
 above but have you met this issue ?
 Could you get around ? 
 
 

I installed Tigase on the asterisk server hosting the queues, our main
office.

Yes I have experienced
https://issues.asterisk.org/jira/browse/ASTERISK-18078
But for us, we have a fibre link between 3 offices, so hardly ever see the
problem, only when I reboot the server with XMPP do the other 2 asterisk's
segfault.

The work around for us is:
  Don't reboot XMPP/Asterisk server during critical periods, however they
have a cron script checking to see whether asterisk is alive, and if not
restart asterisk.

Prior to having the luxury of private 10Mb fibre links, we had to rely on
internet ADSL VPN links between our offices, no good for voice, but reliably
enough for device state updates.

Alec


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Re: [asterisk-users] Queues and distributed device state over WAN

[Matthew Jordan]

On 01/25/2013 01:59 PM, Alec Davis wrote:
 
 I've not tried to publish device state with XMPP yet but I've 
 discovered this issue 
 https://issues.asterisk.org/jira/browse/ASTERISK-18078
 I'm planning to install my XMPP server on the same machine as 
 one asterisk server so hopefully, I won't be hit by the issue 
 above but have you met this issue ?
 Could you get around ? 


 
 I installed Tigase on the asterisk server hosting the queues, our main
 office.
 
 Yes I have experienced
 https://issues.asterisk.org/jira/browse/ASTERISK-18078
 But for us, we have a fibre link between 3 offices, so hardly ever see the
 problem, only when I reboot the server with XMPP do the other 2 asterisk's
 segfault.
 
 The work around for us is:
   Don't reboot XMPP/Asterisk server during critical periods, however they
 have a cron script checking to see whether asterisk is alive, and if not
 restart asterisk.
 
 Prior to having the luxury of private 10Mb fibre links, we had to rely on
 internet ADSL VPN links between our offices, no good for voice, but reliably
 enough for device state updates.
 

Not that this is an excuse or a valid workaround for everyone, but I
believe that issue won't apply if you're using Asterisk 11 and res_xmpp.

res_jabber: yup, totally still a problem.

-- 
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Digium, Inc. | Engineering Manager
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] Queues and distributed device state over WAN

[Alec Davis]

 Not that this is an excuse or a valid workaround for 
 everyone, but I believe that issue won't apply if you're 
 using Asterisk 11 and res_xmpp.
 
 res_jabber: yup, totally still a problem.

Hmm. We're using Asterisk 11, but I still think res_jabber.

Why havn't I changed to res_xmpp, I have no answer for that.

Alec


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[asterisk-users] Queues and distributed device state over WAN

[Olivier]

Hello,

I've read a couple of docs on the web about distributed call centers
(mainly this one
http://www.google.com/url?sa=trct=jq=esrc=ssource=webcd=2cad=rjaved=0CDgQFjABurl=http%3A%2F%2Fleifmadsen.com%2Fsites%2Fdefault%2Ffiles%2Fcc01-madsen-digium.pdfei=wcT9ULmWLIKIhQf6zIB4usg=AFQjCNF1ehGNvK63uDyq4FDS6_B9OsjZDQbvm=bv.41248874,d.ZG4
).

Before diving into this, I've got the following question :

- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
- Alice's phone can be reached from server A phones dialing Local/6789
- Alice's phone can also be reached from server B phones dialing
Local/00123456789


1. How do you configure both servers so that Alice's phone becomes a Queue
Member from a server B given queue ?
Simply calling AddQueueMember on server B, passing Local/00123456789 as
interface value (ie AddQueueMember second argument) ?

2. Then, how should server A publish Alice's phone status ?
How should server B consume Alice's phone status and associate it with the
Queue member activity ? Using AddQueueMember stateinterface argument ?

Regards
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[asterisk-users] queues show some agents (In use) from the start

[Rafael Visser]

Hi gurus
Some of my agents  are in use with no call involved.


   Members:
  Local/0971230031@internal/n (In use) has taken no calls yet
  Local/0972105500@internal/n (In use) has taken no calls yet
   No Callers


Is there a workarround so solve this?
Wham am i doing wrong?

If i change agent Local/0971230031 for an other mobile line works
fine :(


This is my config
Asterisk 1.6.1.6
DAHDI Version: 2.6.1
libss7 version: 1.0.2

queues.conf
[RCEN]
musicclass = default
strategy = rrmemory
weight=0
wrapuptime=15
autopause=no
setinterfacevar=yes
setqueueentryvar=yes
setqueuevar=yes
eventwhencalled = yes
ringinuse = no
joinempty=yes
member = Local/0972105500@internal/n
member = Local/0971230031@internal/n

Thanks in advance.
rv
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Re: [asterisk-users] Queues and Distinctive Ring with Alert-Info

[Larry Moore]

On 26/11/2012 10:14 AM, Klaverstyn, David C wrote:

Hi All,

I’m new to Queues and I have created one as follows which seems to work ok.

[david-test]

strategy = rrmemory

timeout = 10

retry = 0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

member = SIP/121

member = SIP/122

member = SIP/123

I’m wondering how do you change the SipAddHeader/Alert-Info when a call
comes from a queue so users know it is a queue that is calling?

Is something like the following supposed to work?

exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4)

exten = 0453451564,2,Queue(david-test)




Seems to work with Asterisk 1.8.18.0.

I'm using extensions.ael and have tested the following;

400 = {
SIPAddHeader(Alert-Info: n=Classic-4;w=3;c=4);
Queue(400,inrt,,,30);
Hangup();
};


Larry.


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[asterisk-users] Queues and Distinctive Ring with Alert-Info

[Klaverstyn, David C]

Hi All,

I'm new to Queues and I have created one as follows which seems to work ok.

[david-test]
strategy = rrmemory
timeout = 10
retry = 0
maxlen = 0
announce-frequency = 0
announce-holdtime = no
member = SIP/121
member = SIP/122
member = SIP/123


I'm wondering how do you change the SipAddHeader/Alert-Info when a call comes 
from a queue so users know it is a queue that is calling?

Is something like the following supposed to work?

exten = 0453451564,1,SipAddHeader(Alert-Info: n=Classic-4;w=3;c=4)
exten = 0453451564,2,Queue(david-test)


Regards
David Klaverstyn
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[asterisk-users] Queues, pickup and transfers

[Olivier]

Hi,

If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
the application from working some other apps such as PickUp.

I wonder if such things are possible (and if possible, still keep useful
Queue Logs ie logs in which picked up or transfered calls are shown as
such):
1- a call enters a queue, a phone rings, and a non-Queue member dials some
digits and speaks with caller
2- a call enters a queue, a phone rings, basically a BLF attached to this
phone blinks and someone pressing this BLS would pick the call up

Which method should be used for case 1 ? Using Transfer or PickUp ?

Regards
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Re: [asterisk-users] Queues, pickup and transfers

[Olivier]

2011/5/5 Olivier oza_4...@yahoo.fr

 Hi,

 If my memory serves me right, up to Asterisk 1.6, Queue app internals kept
 the application from working some other apps such as PickUp.

 I wonder if such things are possible (and if possible, still keep useful
 Queue Logs ie logs in which picked up or transfered calls are shown as
 such):
 1- a call enters a queue, a phone rings, and a non-Queue member dials some
 digits and speaks with caller
 2- a call enters a queue, a phone rings, basically a BLF attached to this
 phone blinks and someone pressing this BLS would pick the call up

 Which method should be used for case 1 ? Using Transfer or PickUp ?

 Regards



 Thinking over this, I told myself that in some situations, MeetMe (or
ConfBridge) or Parking could be used instead of Queue : caller would hear
MusicOnHold while waiting to be answered and while at the same time, a call
could be originated and then transfered to appropriate channel.

Anyway, my questions remain as Queue fit my other requirements.
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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

[Mike]

  The Queue() application can automatically pause members who fail to
  answer; this would be the solution to your problem. With that solution
  in place, though, the agent will still need to be able to un-pause
  when they return to their desk, and since that is the case, they
  really should be taught to go on pause when they leave their desk as
  well :-)
 
 Not to mention that your caller has to wait for however long your agent
 timeout is when this happens the first time, which is bad customer
 service.
 
 I am a little confused as to what the OP wants the system to do? Call the
 proper agent, but when they don't answer, on the next call, it shouldn't
 call the same agent? OK, but for how long? 5 minutes? Until they manually
 unpause (current option as described by Kevin), 30 minutes? Should it then
 up their penalty? For how long?


I thought I replied back, but can`t find my own reply on the list.

I just want a sequenced queue. I DO know that it means a bit longer waiting
time if the first agent is unavailable, but I`m willing to live with it. I
know I could write the extension to ring phones in sequence, but the queue
includes other useful functionality (and logs that can be parsed and
statistics created from it).  I would be surprised if a queue cannot be made
to ring phones always in the same order (phone A, B and C, back to A, B,
C...). Is this simple use case possible?

Mike



 




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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

[Mike]

 
 I am a little confused as to what the OP wants the system to do? Call the
 proper agent, but when they don't answer, on the next call, it shouldn't
 call the same agent? OK, but for how long? 5 minutes? Until they manually
 unpause (current option as described by Kevin), 30 minutes? Should it then
 up their penalty? For how long?

I should have been more precise.  I don't actually expect all this to
happen, but here's what I wish it did:

1) Ring agents in Round Robin fashion, but always in the same order (could
simply use the already existing penalty value)
2) Always start from the top (taking into account the ringinuse value)

Basically, a simple _pre-ordered_ Roundrobin.

I could make this even better by (as you hinted at yourself) by using
autopause and asking for an autounpause after x minutes feature.  But
those two things above would be wonderful, and I was actually surprised that
it wasn't a possible setting.  Unless I can order the agents somehow, but I
seem to understand that dynamic agents are sequenced in the order in which
they joined the queue, not according to some easily defined position value.

How I would envision this being configured? A queue setting that would
define how it handles penalty.  Either in the current Ring the best
agent(s) over and over again or try the good agents first, but then move
on. Just a yes/no value.

Mike




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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

[Doug Lytle]

Mike wrote:

I was hoping to use this Queue not for professional agents in a call center,
but for reception.  When the receptionist (lowest penalty) is not at the
desk, then some junior sales person can pick up those calls.
   


We have our receptionist setup in a front-desk queue that has 2 phones 
in it.


The incoming call rings directly to the phone for 30 seconds, if not 
answered, plays the, Please wait while we find someone and then drops 
them into a queue.  At this point, it rings the operator phone again and 
if that fails, the 2nd phone.


This will bounce back and forth between phones, until finally dropping 
the call into our dial-by-name directory if nobody answers.


We also have both phones in a call group/pickup group, allowing to grab 
a call by doing a *7


Doug



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deserve neither Liberty nor Safety.


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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

[Mike]

 We have our receptionist setup in a front-desk queue that has 2 phones in
 it.
 
 The incoming call rings directly to the phone for 30 seconds, if not
 answered, plays the, Please wait while we find someone and then drops
 them into a queue.  At this point, it rings the operator phone again and
 if that fails, the 2nd phone.
 
 This will bounce back and forth between phones, until finally dropping the
 call into our dial-by-name directory if nobody answers.
 
 We also have both phones in a call group/pickup group, allowing to grab a
 call by doing a *7
 

Thanks Doug.  I realize there are many things I can do, I was just hoping to
use an application command to do it all.  What you described might just be
what I end up doing.

Regards,

Mike


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Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer

[Tom Rymes]

On Feb 3, 2011, at 11:12 AM, Kevin P. Fleming wrote:

 The Queue() application can automatically pause members who fail to answer; 
 this would be the solution to your problem. With that solution in place, 
 though, the agent will still need to be able to un-pause when they return to 
 their desk, and since that is the case, they really should be taught to go on 
 pause when they leave their desk as well :-)

Not to mention that your caller has to wait for however long your agent timeout 
is when this happens the first time, which is bad customer service. 

I am a little confused as to what the OP wants the system to do? Call the 
proper agent, but when they don't answer, on the next call, it shouldn't call 
the same agent? OK, but for how long? 5 minutes? Until they manually unpause 
(current option as described by Kevin), 30 minutes? Should it then up their 
penalty? For how long? 

Maybe some more specifics would help here.

Tom
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[asterisk-users] Queues with ringinuse=yes

[Vinícius Fontes]

I'm setting up a queue for two independent operator phones that are capable of 
answering multiple calls at once. It's currently working with the following 
settings and Asterisk 1.4:

queues.conf:

[telefonistas]
strategy=roundrobin
;strategy=leastrecent
music=default
timeout=60
retry=0
maxlen=0
wrapuptime=0
ringinuse=yes
autofill=yes
joinempty=yes
member = SIP/8899
member = SIP/8898



extensions.conf:

exten = _900[0-2],1,Answer()
exten = _900[0-2],n,Queue(telefonistas,r,,,45)
exten = _900[0-2],n,Dial(DAHDI/g3/9072) ;If no operator answers



Both SIP phones are set up with call-limit=4 so each phone will receive a 
maximum of 4 calls before Asterisk considers the phone busy and don't sends 
calls to it anymore.

The tricky part comes now: the customer asked me to load balance the phones, 
so the next incoming call should be sent to the phone with the least calls. For 
example:

a) Phone1 has one call, Phone2 has two. Next incoming call should be routed to 
Phone1.
b) Phone1 has two calls, Phone2 has zero. Next incoming call should be routed 
to Phone2.
c) Both Phone1 and Phone2 have one call. Next incoming call can be routed to 
any of them.

And so on.

I have never done something similar before. I tried changing the queue strategy 
to leastrecent but that didn't solve the issue, because both Phone1 and Phone2 
must also make calls in addition to answering them, and that should be taken in 
consideration.

I'm not sure if such a thing is possible using Queue() at all. As a last resort 
I'll write a dialplan that will check how many calls each phone has and route 
appropriately, but I really would like to implement that using Queue() if 
possible.

Any ideas? Thanks in advance.




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[asterisk-users] Queues, priorities and (miscalculated) holdtimes

[Daniel Tryba]

Anyone ever noticed that the reported holdtime is wrong when there are
different priorities? Also talktime is 0, but for the moment I don't
care.

queue show test reports:
test has 23 calls (max unlimited) in 'ringall' strategy (193s holdtime,
0s talktime)
[...]
   Callers: 
  1. Local/3...@default-8828;2 (wait: 3:32, prio: 15)
  2. Local/3...@default-8361;2 (wait: 3:32, prio: 15)
[...]
  6. Local/1...@default-b575;2 (wait: 9:25, prio: 10)
  7. Local/1...@default-fce5;2 (wait: 9:21, prio: 10)
[...]
  22. Local/3...@default-89fb;2 (wait: 6:45, prio: 5)
  23. Local/3...@default-7264;2 (wait: 0:08, prio: 5)

The reported holdtime to the caller is 3 minutes! The wait time of
caller number 1, instead of the 9 minutes holdtime of caller number 6.
This is a realtime Queue on 1.6.2.13.


Also can the priority be changed dynamically (using AMI)? For now I'm 
using:
  exten = s,n,Set(QUEUE_PRIO=5)
  exten = s,n,Queue(test,twrC,,,900)
  exten = s,n,Set(QUEUE_PRIO=15)
  exten = s,n,Queue(test,twr,,,3600)
some callers and 
  exten = s,n,Set(QUEUE_PRIO=10)
  exten = s,n,Queue(test,twr,,,3600)
for all others. Changing the the priority dynamically depending on
context and waittime should avoid the wrong reported holdtime.

-- 

   Daniel Tryba

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Re: [asterisk-users] Queues without agent login

[jonas kellens]

Barry,

I'm using the Asterisk GUI. When defining a User extension (menu 'user')
the only option I have is Is Agent.
The SIP extension (11) is automatically created as an agent that needs
to log in.

In the advanced options I can manually edit queues.conf and change to
member=SIP/11.

This way of defining queues and members is a 2-step work. Can it be done
in 1 step using the Asterisk GUI ??

Kind regards,
Jonas.

On Wed, 2009-11-18 at 10:42 -0500, Barry L. Kline wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1
 
 jonas kellens wrote:
  Is it possible to make use of queues for incoming calls but to have
  agents that do not need to log in ?
  
 
 Make the phones members of the queue.  In queues.conf:
 
 
 [MY_QUEUE]
 
 member = SIP/1234
 member = SIP/5678
 
 etc.
 
 Barry


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[asterisk-users] Queues without agent login

[jonas kellens]

Is it possible to make use of queues for incoming calls but to have
agents that do not need to log in ?

If I create a queue and make certain SIP-users member of the queue, do
these SIP-users always need to log in to the queue to be able to receive
calls that are in the queue ??

Can't a member be just available when the phone is registered to the
Asterisk-server ? In stead of also having to call an extension to log in
(and having to give some PIN).

I just want a queue (with MoH) to collect multiple incoming calls and
then one at a time transfer them to an available SIP-phone.

Is this possible ?

Thanks you.

Jonas.
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Re: [asterisk-users] Queues without agent login

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

jonas kellens wrote:
 Is it possible to make use of queues for incoming calls but to have
 agents that do not need to log in ?
 

Make the phones members of the queue.  In queues.conf:


[MY_QUEUE]

member = SIP/1234
member = SIP/5678

etc.

Barry

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFLBBYACFu3bIiwtTARAg5oAKClAtJ98LaSXnjCDBx4xlRcLQ9l/wCgoeI+
BAi7wu2nQ6vNPZSaLCDB4DA=
=egpe
-END PGP SIGNATURE-

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Re: [asterisk-users] Queues without agent login

[Tarek Sawah]

Simply use 
member=SIP/Tarek
member=IAX2/JONAS
member=LOCAL/whatever

simple and good.. 
with member=SIP/extension  i'm facing a CALL WAITING issue.. the agent hears a 
callwaiting signal whenever the queue tries to call .. so i woul dsuggest using 
call-limit and busy limite with all your Agents

--
AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308






From: jonas.kell...@telenet.be
To: asterisk-users@lists.digium.com
Date: Wed, 18 Nov 2009 16:21:12 +0100
Subject: [asterisk-users] Queues without agent login






  
  


Is it possible to make use of queues for incoming calls but to have agents that 
do not need to log in ?



If I create a queue and make certain SIP-users member of the queue, do these 
SIP-users always need to log in to the queue to be able to receive calls that 
are in the queue ??



Can't a member be just available when the phone is registered to the 
Asterisk-server ? In stead of also having to call an extension to log in (and 
having to give some PIN).



I just want a queue (with MoH) to collect multiple incoming calls and then one 
at a time transfer them to an available SIP-phone.



Is this possible ?



Thanks you.



Jonas.
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[asterisk-users] Queues

[Travis Elsberry]

Hello Everyone, 

I'm looking for help/ideas on how to do the following: 

I have a couple of people out of many (the couple of people randomly change) 
who log into an on-call queue. A call comes in and it rings the on-call 
extensions, but no one answers. I would like the call to then try the 
cell-phones of just the people that are logged into the on-call queue. 

I've got the queue setup and the people log into and out of it by dialing 
extensions that use AddQueueMember() and RemoveQueueMember() respectively. I 
tried using QUEUE_MEMBER_LIST to write to a database list when the call comes 
in however it keeps adding duplicates each time the call goes into the queue. 
I'm just not seeing how to pass the call that goes into the queue to a dynamic 
list on the way out. Is attempting something like this even realistic? 

Thanks in advance, 
Travis 
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Re: [asterisk-users] Queues

[Danny Nicholas]

It should be realistic, but have you considered just using followme to add
the cell phones to the queue list?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis
Elsberry
Sent: Monday, November 16, 2009 3:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queues

 

Hello Everyone,

I'm looking for help/ideas on how to do the following:

I have a couple of people out of many (the couple of people randomly change)
who log into an on-call queue.  A call comes in and it rings the on-call
extensions, but no one answers.  I would like the call to then try the
cell-phones of just the people that are logged into the on-call queue.

I've got the queue setup and the people log into and out of it by dialing
extensions that use AddQueueMember() and RemoveQueueMember() respectively.
I tried using QUEUE_MEMBER_LIST to write to a database list when the call
comes in however it keeps adding duplicates each time the call goes into the
queue.  I'm just not seeing how to pass the call that goes into the queue to
a dynamic list on the way out.  Is attempting something like this even
realistic?

Thanks in advance,
Travis

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Re: [asterisk-users] Queues

[Travis Elsberry]

I had looked at followme as a solution but ran into the same stumbling block of 
having to hard code the cell phone list. I didn't see a dynamic way of the list 
being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on 
Tuesday without editing the extensions.conf file manually each day. Did I 
overlook something in how followme works? 

- Original Message - 
From: Danny Nicholas da...@debsinc.com 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Monday, November 16, 2009 1:37:04 PM 
Subject: Re: [asterisk-users] Queues 




It should be realistic, but have you considered just using followme to add the 
cell phones to the queue list? 






From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry 
Sent: Monday, November 16, 2009 3:25 PM 
To: asterisk-users@lists.digium.com 
Subject: [asterisk-users] Queues 




Hello Everyone, 

I'm looking for help/ideas on how to do the following: 

I have a couple of people out of many (the couple of people randomly change) 
who log into an on-call queue. A call comes in and it rings the on-call 
extensions, but no one answers. I would like the call to then try the 
cell-phones of just the people that are logged into the on-call queue. 

I've got the queue setup and the people log into and out of it by dialing 
extensions that use AddQueueMember() and RemoveQueueMember() respectively. I 
tried using QUEUE_MEMBER_LIST to write to a database list when the call comes 
in however it keeps adding duplicates each time the call goes into the queue. 
I'm just not seeing how to pass the call that goes into the queue to a dynamic 
list on the way out. Is attempting something like this even realistic? 

Thanks in advance, 
Travis 
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Re: [asterisk-users] Queues

[Danny Nicholas]

Since followme is extension-based, you have at least two options.  Option
1 is to have a few extensions designated for following where you punch in
the cell numbers as you wish.  Option 2 is to use day logic to point to
the following guys based on days.If I were doing option 2, I'd try to
use ASTDB to control this instead of having to code a lot of dialplan, but
that's just me.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis
Elsberry
Sent: Monday, November 16, 2009 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues

 

I had looked at followme as a solution but ran into the same stumbling block
of having to hard code the cell phone list.  I didn't see a dynamic way of
the list being extensions 12 and 14 on Monday, but changing to extensions 13
and 19 on Tuesday without editing the extensions.conf file manually each
day.  Did I overlook something in how followme works?

- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 1:37:04 PM
Subject: Re: [asterisk-users] Queues

It should be realistic, but have you considered just using followme to add
the cell phones to the queue list?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis
Elsberry
Sent: Monday, November 16, 2009 3:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queues

 

Hello Everyone,

I'm looking for help/ideas on how to do the following:

I have a couple of people out of many (the couple of people randomly change)
who log into an on-call queue.  A call comes in and it rings the on-call
extensions, but no one answers.  I would like the call to then try the
cell-phones of just the people that are logged into the on-call queue.

I've got the queue setup and the people log into and out of it by dialing
extensions that use AddQueueMember() and RemoveQueueMember() respectively.
I tried using QUEUE_MEMBER_LIST to write to a database list when the call
comes in however it keeps adding duplicates each time the call goes into the
queue.  I'm just not seeing how to pass the call that goes into the queue to
a dynamic list on the way out.  Is attempting something like this even
realistic?

Thanks in advance,
Travis


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Re: [asterisk-users] Queues

[Michael Wyres]

Hi Travis,

There's lots of different ways to attack on-call roster solutions in Asterisk 
- as Danny suggested, FollowMe() is definitely an option (and normally the 
best), but it doesn't always suit the business need.  However, also as Danny 
suggested, in most cases using ASTDB in some way to simplify dialling plans is 
the way to go - then you just have to decide how you want to update the 
information as to the number to call, in ASTDB.

For example, I had a customer a couple of years back who desperately wanted to 
manage his on-call roster routing using a web interface.  I dollied up a 
simple PHP/MySQL web interface with a list of all the people (and their 
mobile/cell numbers) in a drop down list - they could simply select the right 
person, and click a First Call button to make that person the first in the 
roster, select another person and click a Second Call button to make that 
person the second in the roster, and so on.

Using the Asterisk manager interface - (or even asterisk -rx command if 
you're not comfortable using the AMI) - you get the numbers selected into ASTDB.

The dialplan just comes along then and reads the appropriate numbers from ASTDB 
as it steps through, and dials the people in order.

As with many things in Asterisk - there is more than one way to hump the leg.


Cheers
Michael

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, 17 November 2009 08:57
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Queues

Since followme is extension-based, you have at least two options.  Option 1 
is to have a few extensions designated for following where you punch in the 
cell numbers as you wish.  Option 2 is to use day logic to point to the 
following guys based on days.If I were doing option 2, I'd try to use 
ASTDB to control this instead of having to code a lot of dialplan, but that's 
just me...


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues

I had looked at followme as a solution but ran into the same stumbling block of 
having to hard code the cell phone list.  I didn't see a dynamic way of the 
list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 
on Tuesday without editing the extensions.conf file manually each day.  Did I 
overlook something in how followme works?

- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 1:37:04 PM
Subject: Re: [asterisk-users] Queues
It should be realistic, but have you considered just using followme to add the 
cell phones to the queue list?


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queues

Hello Everyone,

I'm looking for help/ideas on how to do the following:

I have a couple of people out of many (the couple of people randomly change) 
who log into an on-call queue.  A call comes in and it rings the on-call 
extensions, but no one answers.  I would like the call to then try the 
cell-phones of just the people that are logged into the on-call queue.

I've got the queue setup and the people log into and out of it by dialing 
extensions that use AddQueueMember() and RemoveQueueMember() respectively.  I 
tried using QUEUE_MEMBER_LIST to write to a database list when the call comes 
in however it keeps adding duplicates each time the call goes into the queue.  
I'm just not seeing how to pass the call that goes into the queue to a dynamic 
list on the way out.  Is attempting something like this even realistic?

Thanks in advance,
Travis

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Re: [asterisk-users] Queues

[Travis Elsberry]

Hi Michael, 

Your web interface for the on-call roster is pretty close to what we're 
trying to trying to achieve. I would like to have people signing into the 
on-call queue be the method that determined whose cell phone to call. I was 
hoping there was a way to pass the call exiting the queue to a variable or two 
that was composed of the extensions currently logged into the queue. 

I set up an extension number for people to call into and enter a forwarding 
number which writes an entry into the ASTDB. I have my dialplan check to see if 
there is an ASTDB entry for that extension before it tries to dial their 
deskphone, and if there is an entry it dials the forwarded number stored in the 
database instead. The closest thing so far I have found to what I am trying to 
achieve is to hard code a couple of spare extensions into the dialplan, and 
then have whoever is on-call set one of those extensions to their cell phone 
number. 

I'll definitely take another look at followme to see if I can adapt that what 
I'm trying to achieve. 

Thanks Danny  Michael, 
Travis 
- Original Message - 
From: Michael Wyres mwy...@cdm.com.au 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Monday, November 16, 2009 2:23:49 PM 
Subject: Re: [asterisk-users] Queues 




Hi Travis, 



There’s lots of different ways to attack “on-call” roster solutions in Asterisk 
– as Danny suggested, FollowMe() is definitely an option (and normally the 
best), but it doesn’t always suit the “business need”. However, also as Danny 
suggested, in most cases using ASTDB in some way to simplify dialling plans is 
the way to go - then you just have to decide how you want to update the 
information as to the number to call, in ASTDB. 



For example, I had a customer a couple of years back who desperately wanted to 
manage his “on-call roster” routing using a web interface. I dollied up a 
simple PHP/MySQL web interface with a list of all the people (and their 
mobile/cell numbers) in a drop down list – they could simply select the right 
person, and click a “First Call” button to make that person the first in the 
roster, select another person and click a “Second Call” button to make that 
person the second in the roster, and so on. 



Using the Asterisk manager interface – (or even “asterisk –rx command” if 
you’re not comfortable using the AMI) – you get the numbers selected into 
ASTDB. 



The dialplan just comes along then and reads the appropriate numbers from ASTDB 
as it steps through, and dials the people in order. 



As with many things in Asterisk – there is more than one way to “hump the leg”. 





Cheers 

Michael 





From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
Sent: Tuesday, 17 November 2009 08:57 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Subject: Re: [asterisk-users] Queues 



Since followme is “extension-based”, you have at least two options. Option 1 is 
to have a few extensions designated for “following” where you punch in the cell 
numbers as you wish. Option 2 is to use “day logic” to point to the “following” 
guys based on days. If I were doing option 2, I’d try to use ASTDB to control 
this instead of having to code a lot of dialplan, but that’s just me… 






From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry 
Sent: Monday, November 16, 2009 3:50 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: Re: [asterisk-users] Queues 





I had looked at followme as a solution but ran into the same stumbling block of 
having to hard code the cell phone list. I didn't see a dynamic way of the list 
being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on 
Tuesday without editing the extensions.conf file manually each day. Did I 
overlook something in how followme works? 

- Original Message - 
From: Danny Nicholas da...@debsinc.com 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Monday, November 16, 2009 1:37:04 PM 
Subject: Re: [asterisk-users] Queues 

It should be realistic, but have you considered just using followme to add the 
cell phones to the queue list? 






From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry 
Sent: Monday, November 16, 2009 3:25 PM 
To: asterisk-users@lists.digium.com 
Subject: [asterisk-users] Queues 




Hello Everyone, 

I'm looking for help/ideas on how to do the following: 

I have a couple of people out of many (the couple of people randomly change) 
who log into an on-call queue. A call comes in and it rings the on-call 
extensions, but no one answers. I would like the call to then try the 
cell-phones of just the people that are logged into the on-call queue. 

I've got

Re: [asterisk-users] Queues

[Michael Wyres]

Again – lots of ways to do it – you could use a web interface to set the 
numbers in ASTDB for lookup – or you could create an IVR to ask for the number, 
and store it in ASTDB that way.

Good luck!



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Tuesday, 17 November 2009 10:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues

Hi Michael,

Your web interface for the on-call roster is pretty close to what we're 
trying to trying to achieve.  I would like to have people signing into the 
on-call queue be the method that determined whose cell phone to call.  I was 
hoping there was a way to pass the call exiting the queue to a variable or two 
that was composed of the extensions currently logged into the queue.

I set up an extension number for people to call into and enter a forwarding 
number which writes an entry into the ASTDB.  I have my dialplan check to see 
if there is an ASTDB entry for that extension before it tries to dial their 
deskphone, and if there is an entry it dials the forwarded number stored in the 
database instead.  The closest thing so far I have found to what I am trying to 
achieve is to hard code a couple of spare extensions into the dialplan, and 
then have whoever is on-call set one of those extensions to their cell phone 
number.

I'll definitely take another look at followme to see if I can adapt that what 
I'm trying to achieve.

Thanks Danny  Michael,
Travis
- Original Message -
From: Michael Wyres mwy...@cdm.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 2:23:49 PM
Subject: Re: [asterisk-users] Queues


Hi Travis,

There’s lots of different ways to attack “on-call” roster solutions in Asterisk 
– as Danny suggested, FollowMe() is definitely an option (and normally the 
best), but it doesn’t always suit the “business need”.  However, also as Danny 
suggested, in most cases using ASTDB in some way to simplify dialling plans is 
the way to go - then you just have to decide how you want to update the 
information as to the number to call, in ASTDB.

For example, I had a customer a couple of years back who desperately wanted to 
manage his “on-call roster” routing using a web interface.  I dollied up a 
simple PHP/MySQL web interface with a list of all the people (and their 
mobile/cell numbers) in a drop down list – they could simply select the right 
person, and click a “First Call” button to make that person the first in the 
roster, select another person and click a “Second Call” button to make that 
person the second in the roster, and so on.

Using the Asterisk manager interface – (or even “asterisk –rx command” if 
you’re not comfortable using the AMI) – you get the numbers selected into ASTDB.

The dialplan just comes along then and reads the appropriate numbers from ASTDB 
as it steps through, and dials the people in order.

As with many things in Asterisk – there is more than one way to “hump the leg”.


Cheers
Michael

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, 17 November 2009 08:57
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Queues

Since followme is “extension-based”, you have at least two options.  Option 1 
is to have a few extensions designated for “following” where you punch in the 
cell numbers as you wish.  Option 2 is to use “day logic” to point to the 
“following” guys based on days.If I were doing option 2, I’d try to use 
ASTDB to control this instead of having to code a lot of dialplan, but that’s 
just me…


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues

I had looked at followme as a solution but ran into the same stumbling block of 
having to hard code the cell phone list.  I didn't see a dynamic way of the 
list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 
on Tuesday without editing the extensions.conf file manually each day.  Did I 
overlook something in how followme works?

- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, November 16, 2009 1:37:04 PM
Subject: Re: [asterisk-users] Queues
It should be realistic, but have you considered just using followme to add the 
cell phones to the queue list?


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry
Sent: Monday, November 16, 2009 3

[asterisk-users] queues autopause

[Rilawich Ango]

Hi,
I have 3 queue set in the table as below.

name,autopause
1000,1
2000,1
3000,1

In queue 1000, the autopause works after member failed to answer call.
 However, other queues don't work for the autopause function.

queue 1000:
-- Nobody picked up in 25000 ms
-- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they
failed to answer.

queue 2000/3000:
-- Nobody picked up in 25000 ms
-- SIP/1234-1544cd90 is ringing

Is it the limitation of the asterisk to support one queue of autopause
function?  Or any setting I need to take care to make autopause
function works for all queue?

ango

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Re: [asterisk-users] queues autopause

[Miguel Molina]

Rilawich Ango escribió:
 Hi,
 I have 3 queue set in the table as below.

 name,autopause
 1000,1
 2000,1
 3000,1

 In queue 1000, the autopause works after member failed to answer call.
  However, other queues don't work for the autopause function.

 queue 1000:
 -- Nobody picked up in 25000 ms
 -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they
 failed to answer.

 queue 2000/3000:
 -- Nobody picked up in 25000 ms
 -- SIP/1234-1544cd90 is ringing

 Is it the limitation of the asterisk to support one queue of autopause
 function?  Or any setting I need to take care to make autopause
 function works for all queue?

 ango

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Hi,

Some months ago there was a discussion about this, with a simple 
solution involving minimal changes to the source (1 line of code). 
Search the archives of this list and you will find the answer. BTW, what 
version of asterisk are you using?

Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] queues autopause

[Rilawich Ango]

Thanks.  Finally, I find that it was caused by the use of the table wrongly.

On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina
mmol...@millenium.com.co wrote:
 Rilawich Ango escribió:
 Hi,
 I have 3 queue set in the table as below.

 name,autopause
 1000,1
 2000,1
 3000,1

 In queue 1000, the autopause works after member failed to answer call.
  However, other queues don't work for the autopause function.

 queue 1000:
     -- Nobody picked up in 25000 ms
     -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they
 failed to answer.

 queue 2000/3000:
     -- Nobody picked up in 25000 ms
     -- SIP/1234-1544cd90 is ringing

 Is it the limitation of the asterisk to support one queue of autopause
 function?  Or any setting I need to take care to make autopause
 function works for all queue?

 ango

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 Hi,

 Some months ago there was a discussion about this, with a simple
 solution involving minimal changes to the source (1 line of code).
 Search the archives of this list and you will find the answer. BTW, what
 version of asterisk are you using?

 Cheers,

 --
 Ing. Miguel Molina
 Grupo de Tecnología
 Millenium Phone Center


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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

C. Chad Wallace cwall...@lodgingcompany.com writes:

 OK, I decided to write it up in AEL.  It's incomplete and untested, but
 it probably gets the idea across a little better.

 context agentcalls {
   _2XX = {
 Set(AGENT=${EXTEN});  // Assuming agent ID is extension.
 
 if (${EPOCH}${DB(AgentPaused/${AGENT})}) {
   // Let the call through to the cell phone
   Dial(...);

   if (cell call was rejected) {
 // Flag agent as paused for the next 30 seconds.
 Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
   };
 }
 else {
   // Agent still paused.
 };
   };
 };

I was going in the same direction at the end of my first mail, but I
hadn't written any code. There is a problem though: The Queue
application will keep sending calls to the Local channel, which have to
be rejected, over and over.

Would it perhaps work to simply Wait(30) if the call is rejected by the
phone? If the Queue assumes that the phone is busy for those 30 seconds,
I have accomplished my goal. It's worth a shot.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

Benny Amorsen benny+use...@amorsen.dk writes:

 Would it perhaps work to simply Wait(30) if the call is rejected by the
 phone? If the Queue assumes that the phone is busy for those 30 seconds,
 I have accomplished my goal. It's worth a shot.

This works! Actually I tried out Wait(1000), but that worked fine. After
30 seconds (the timeout in the queue) the Local channel was closed, and
a short while later a new call attempt was made. Just as I was hoping.

It would still be neat to have a min_dial_interval option, so that Queue
never overwhelms the server with failing dial attempts.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 11:23 AM on 16 Oct 2009, Benny Amorsen wrote:

 I was going in the same direction at the end of my first mail, but I
 hadn't written any code. There is a problem though: The Queue
 application will keep sending calls to the Local channel, which have
 to be rejected, over and over.
 
 Would it perhaps work to simply Wait(30) if the call is rejected by
 the phone? If the Queue assumes that the phone is busy for those 30
 seconds, I have accomplished my goal. It's worth a shot.

It would only be trying one agent at a time for each waiting queue
member...  I don't know how expensive it is to open and close a Local
channel and do a DB lookup, but I wouldn't expect it to be a real
problem.  You are at least avoiding multiple calls out to the cellular
network.  

Also, if there is another agent available, the caller would be connected
immediately, and it wouldn't have to make any more attempts.  With the
Wait() solution, that caller would be waiting for 30 seconds regardless
of whether there's anyone else available.  

Of course, I don't know your business case, so you'll have to decide
which of the two problems is worse.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

C. Chad Wallace cwall...@lodgingcompany.com writes:

 It would only be trying one agent at a time for each waiting queue
 member...

Would it? Almost all our queues are on a ringall strategy.

 I don't know how expensive it is to open and close a Local channel and
 do a DB lookup, but I wouldn't expect it to be a real problem. You are
 at least avoiding multiple calls out to the cellular network.

Not that expensive, but still a bit of a waste when done every couple of
seconds. Especially if multiple agents are unavailable.

 Also, if there is another agent available, the caller would be connected
 immediately, and it wouldn't have to make any more attempts.  With the
 Wait() solution, that caller would be waiting for 30 seconds regardless
 of whether there's anyone else available.  

This bit is solved by the ringall strategy.

 Of course, I don't know your business case, so you'll have to decide
 which of the two problems is worse.

I'm fairly happy with the Wait(1000) solution for now. We'll see if
testing shows any problems with it.


/Benny

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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 7:35 PM on 16 Oct 2009, Benny Amorsen wrote:

 C. Chad Wallace cwall...@lodgingcompany.com writes:
 
  Also, if there is another agent available, the caller would be
  connected immediately, and it wouldn't have to make any more
  attempts.  With the Wait() solution, that caller would be waiting
  for 30 seconds regardless of whether there's anyone else
  available.  
 
 This bit is solved by the ringall strategy.
 
  Of course, I don't know your business case, so you'll have to decide
  which of the two problems is worse.
 
 I'm fairly happy with the Wait(1000) solution for now. We'll see if
 testing shows any problems with it.

Oh yeah, I hadn't even considered the ringall strategy!  With that,
your Wait() solution sounds perfect to me.  Congrats!

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

Elliot Otchet elliot.otc...@callingcircles.com writes:

 Have you tried autopause=yes in your queue configuration? You can then
 unpause the member by either the dialplan (e.g. having the cell phone
 user log back in) or using an AMI based program to change the
 paused state.

 You can read more about the latter here:
 http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html

That looks very interesting, thank you!

First of all though I need to avoid having them autopause just because
they don't answer their phone. It should only happen if the call to
their phone fails completely. I guess that could be done by not doing
autopause but instead pausing manually in the context that the Local
call passes through.

That would also solve my second problem, which is that I need to pause
it in all queues, not just one queue.

The last challenge is to somehow unpause them after a while. In
traditional programming that would be something like keeping a list of
timeout,queuemember ordered by timeout, and then when every call comes
in unpause and remove the ones where timeout expired... I'm not sure
that I can make an ordered list in the dialplan though. I may have to
resort to AGI, but I still need somewhere to actually store the list.
Tricky.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

Lenz Emilitri lenz.lo...@gmail.com writes:

 You could configure them as agents and have them log off automatically
 after a while they're not responding.

Agents have to log in and wait for calls though, don't they? There used
to be AgentCallbackLogin, but that has been replaced by dialplan code
and chan_local.

Otherwise a nice idea though.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[Elliot Otchet]

That shouldn't be too hard to accomplish.  If you've got the addons (and mysql) 
installed you could store them in a MySQL table (timestamp, device) and have a 
cron job set to run at X frequency that un-pauses the queue members via AMI.  
Don't want to go to MySQL?  Use system() to 'touch' files named after devices.  
Then have your cron script go through the files by creation date.  Either way 
gets you there.

-Elliot

-Original Message-
From: Benny Amorsen [mailto:benny+use...@amorsen.dk]
Sent: Thursday, October 15, 2009 5:06 AM
To: Elliot Otchet
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Queues with unavailable members

Elliot Otchet elliot.otc...@callingcircles.com writes:

 Have you tried autopause=yes in your queue configuration? You can then
 unpause the member by either the dialplan (e.g. having the cell phone
 user log back in) or using an AMI based program to change the
 paused state.

 You can read more about the latter here:
 http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html

That looks very interesting, thank you!

First of all though I need to avoid having them autopause just because
they don't answer their phone. It should only happen if the call to
their phone fails completely. I guess that could be done by not doing
autopause but instead pausing manually in the context that the Local
call passes through.

That would also solve my second problem, which is that I need to pause
it in all queues, not just one queue.

The last challenge is to somehow unpause them after a while. In
traditional programming that would be something like keeping a list of
timeout,queuemember ordered by timeout, and then when every call comes
in unpause and remove the ones where timeout expired... I'm not sure
that I can make an ordered list in the dialplan though. I may have to
resort to AGI, but I still need somewhere to actually store the list.
Tricky.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[Benny Amorsen]

Elliot Otchet elliot.otc...@callingcircles.com writes:

 That shouldn't be too hard to accomplish. If you've got the addons
 (and mysql) installed you could store them in a MySQL table
 (timestamp, device) and have a cron job set to run at X frequency that
 un-pauses the queue members via AMI. Don't want to go to MySQL? Use
 system() to 'touch' files named after devices. Then have your cron
 script go through the files by creation date. Either way gets you
 there.

This seems like a very heavyweight solution. Having a cron job running
every minute isn't particularly attractive, and making a daemon do the
job isn't my cup of tea either.

Perhaps the problem could be restated in a different way: After a queue
member rejects a call (instead of just not answering), the queue should
wait X amount of time before sending the next call. Queues.conf has a
million settings, but I can't find one which does this.


/Benny


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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:

 Perhaps the problem could be restated in a different way: After a
 queue member rejects a call (instead of just not answering), the
 queue should wait X amount of time before sending the next call.
 Queues.conf has a million settings, but I can't find one which does
 this.

To pause an agent, store the unpause time per agent in the AstDB.
Then when you're deciding whether to give out a call (in the Local
channel), look up ${DB(AgentPaused/agentid)} and compare it to the
current time.  If there is no record or the time has passed, put the
call through; otherwise, skip that agent.

Sorry, no example code yet...  I just wanted to get the idea out there.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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Re: [asterisk-users] Queues with unavailable members

[C. Chad Wallace]

At 11:32 AM on 15 Oct 2009, C. Chad Wallace wrote:

 At 3:37 PM on 15 Oct 2009, Benny Amorsen wrote:
 
  Perhaps the problem could be restated in a different way: After a
  queue member rejects a call (instead of just not answering), the
  queue should wait X amount of time before sending the next call.
  Queues.conf has a million settings, but I can't find one which does
  this.
 
 To pause an agent, store the unpause time per agent in the AstDB.
 Then when you're deciding whether to give out a call (in the Local
 channel), look up ${DB(AgentPaused/agentid)} and compare it to the
 current time.  If there is no record or the time has passed, put the
 call through; otherwise, skip that agent.
 
 Sorry, no example code yet...  I just wanted to get the idea out
 there.

OK, I decided to write it up in AEL.  It's incomplete and untested, but
it probably gets the idea across a little better.

context agentcalls {
  _2XX = {
Set(AGENT=${EXTEN});  // Assuming agent ID is extension.

if (${EPOCH}${DB(AgentPaused/${AGENT})}) {
  // Let the call through to the cell phone
  Dial(...);

  if (cell call was rejected) {
// Flag agent as paused for the next 30 seconds.
Set(DB(AgentPaused/${AGENT})=$[${EPOCH}+30]);
  };
}
else {
// Agent still paused.
};
  };
};


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0



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[asterisk-users] Queues with unavailable members

[Benny Amorsen]

We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP registration for those unfortunately
doesn't go away just because the phone is off, since the registration is
done by our cell-phone=SIP gateway, and that gateway has no way of
knowing whether the phone is on or off.

This is usually ok, but it gets problematic if the cell phone is a
member of a queue. In that case Queue() keeps sending the call to the
phone, and the cell-phone=SIP gateway dutifully makes a call, which is
then rejected by the cellular network. A few seconds later, Queue()
tries again. This needlessly wastes resources both in Asterisk and in
the cellular network.

One idea is to run the call through chan_local (we do this anyway
because we need to format the caller-ID differently for different
phones) and then record if a call is rejected, and for the next
30 seconds just abort if we are asked to send a call to that particular
phone. The downside is that we are still running a call through part of
the dial plan, but at least it can be done in perhaps 3 lines of code.

I would very much like to hear about smarter ways to do it.


/Benny



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Re: [asterisk-users] Queues with unavailable members

[Lenz Emilitri]

You could configure them as agents and have them log off automatically after
a while they're not responding.
l.



2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk


 We have the possibly rather unique setup where we have cell phones
 posing as SIP devices. The SIP registration for those unfortunately
 doesn't go away just because the phone is off, since the registration is
 done by our cell-phone=SIP gateway, and that gateway has no way of
 knowing whether the phone is on or off.

 This is usually ok, but it gets problematic if the cell phone is a
 member of a queue. In that case Queue() keeps sending the call to the
 phone, and the cell-phone=SIP gateway dutifully makes a call, which is
 then rejected by the cellular network. A few seconds later, Queue()
 tries again. This needlessly wastes resources both in Asterisk and in
 the cellular network.

 One idea is to run the call through chan_local (we do this anyway
 because we need to format the caller-ID differently for different
 phones) and then record if a call is rejected, and for the next
 30 seconds just abort if we are asked to send a call to that particular
 phone. The downside is that we are still running a call through part of
 the dial plan, but at least it can be done in perhaps 3 lines of code.

 I would very much like to hear about smarter ways to do it.


 /Benny



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Re: [asterisk-users] Queues with unavailable members

[RSCL Mumbai]

What is the command to log off the agents ?

Thx


On Wed, Oct 14, 2009 at 6:45 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:

 You could configure them as agents and have them log off automatically
 after a while they're not responding.
 l.



 2009/10/14 Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
 

 We have the possibly rather unique setup where we have cell phones
 posing as SIP devices. The SIP registration for those unfortunately
 doesn't go away just because the phone is off, since the registration is
 done by our cell-phone=SIP gateway, and that gateway has no way of
 knowing whether the phone is on or off.

 This is usually ok, but it gets problematic if the cell phone is a
 member of a queue. In that case Queue() keeps sending the call to the
 phone, and the cell-phone=SIP gateway dutifully makes a call, which is
 then rejected by the cellular network. A few seconds later, Queue()
 tries again. This needlessly wastes resources both in Asterisk and in
 the cellular network.

 One idea is to run the call through chan_local (we do this anyway
 because we need to format the caller-ID differently for different
 phones) and then record if a call is rejected, and for the next
 30 seconds just abort if we are asked to send a call to that particular
 phone. The downside is that we are still running a call through part of
 the dial plan, but at least it can be done in perhaps 3 lines of code.

 I would very much like to hear about smarter ways to do it.


 /Benny



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 --
 Loway - home of QueueMetrics - http://queuemetrics.com


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Re: [asterisk-users] Queues with unavailable members

[Elliot Otchet]

Have you tried autopause=yes in your queue configuration?  You can then unpause 
the member by either the dialplan (e.g. having the cell phone user log back 
in) or using an AMI based program to change the paused state.

You can read more about the latter here:  
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-F-30.html

-Elliot

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
Sent: Wednesday, October 14, 2009 7:58 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Queues with unavailable members

We have the possibly rather unique setup where we have cell phones
posing as SIP devices. The SIP registration for those unfortunately
doesn't go away just because the phone is off, since the registration is
done by our cell-phone=SIP gateway, and that gateway has no way of
knowing whether the phone is on or off.

This is usually ok, but it gets problematic if the cell phone is a
member of a queue. In that case Queue() keeps sending the call to the
phone, and the cell-phone=SIP gateway dutifully makes a call, which is
then rejected by the cellular network. A few seconds later, Queue()
tries again. This needlessly wastes resources both in Asterisk and in
the cellular network.

One idea is to run the call through chan_local (we do this anyway
because we need to format the caller-ID differently for different
phones) and then record if a call is rejected, and for the next
30 seconds just abort if we are asked to send a call to that particular
phone. The downside is that we are still running a call through part of
the dial plan, but at least it can be done in perhaps 3 lines of code.

I would very much like to hear about smarter ways to do it.


/Benny



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which it is addressed and may contain information that is privileged, 
confidential, and/or proprietary to Calling Circles LLC and its affiliates. If 
the reader of this message is not the intended recipient, you are hereby 
notified that any dissemination, distribution, forwarding or copying of this 
communication is prohibited without the express permission of the sender. If 
you have received this communication in error, please notify the sender 
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[asterisk-users] queues load balancing

[Joao Gomes Pereira]

Hello
I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to 
send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2
How can I do that load balancing in extensions.conf?

I have something like this:
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer

;  2 in 3 calls go to queue_1
exten = 123,x,Queue(queue_1)

; 1 in 3 calls go to queue_2
exten = 123,x,Queue(queue_2)

But how can I configure this call distribution?
Thanks
Regards
Joao Pereira

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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Re: [asterisk-users] queues load balancing

[Geraint Lee]

Take a look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random

You should be able to do what you want with this, it obviously won't take in
to account the actual amount of people still in the queue (for example if
someone hangs up while on hold). I'm sure there'd be a way of integrating
this in to it using some different functions, but for a quick fix random
will do just fine.

Cheers

2009/7/20 Joao Gomes Pereira gomespere...@startel.pt

 Hello
 I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to
 send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2
 How can I do that load balancing in extensions.conf?

 I have something like this:
 exten = 123,1,Ringing
 exten = 123,2,Wait(1)
 exten = 123,3,Answer

 ;  2 in 3 calls go to queue_1
 exten = 123,x,Queue(queue_1)

 ; 1 in 3 calls go to queue_2
 exten = 123,x,Queue(queue_2)

 But how can I configure this call distribution?
 Thanks
 Regards
 Joao Pereira

 --
 StarTel - A Rede Livre
 Joao Gomes Pereira
 www.startel.pt
 +351 304500650
 sip: gomespere...@startel.pt


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Re: [asterisk-users] queues load balancing

[Joao Gomes Pereira]

Thanks for the idea.
I will try it this way:

exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Random(33:123,10)
exten = 123,5,Queue(queue_1)
exten = 123,6,Hangup

exten = 123,10,Queue(queue_2)
exten = 123,11,Hangup


Joao Pereira

-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt





Geraint Lee wrote:
 Take a look at:
 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random

 You should be able to do what you want with this, it obviously won't 
 take in to account the actual amount of people still in the queue (for 
 example if someone hangs up while on hold). I'm sure there'd be a way 
 of integrating this in to it using some different functions, but for a 
 quick fix random will do just fine.

 Cheers

 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt 
 mailto:gomespere...@startel.pt

 Hello
 I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to
 send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2
 How can I do that load balancing in extensions.conf?

 I have something like this:
 exten = 123,1,Ringing
 exten = 123,2,Wait(1)
 exten = 123,3,Answer

 ;  2 in 3 calls go to queue_1
 exten = 123,x,Queue(queue_1)

 ; 1 in 3 calls go to queue_2
 exten = 123,x,Queue(queue_2)

 But how can I configure this call distribution?
 Thanks
 Regards
 Joao Pereira

 --
 StarTel - A Rede Livre
 Joao Gomes Pereira
 www.startel.pt http://www.startel.pt
 +351 304500650
 sip: gomespere...@startel.pt mailto:gomespere...@startel.pt


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-- 
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt


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Re: [asterisk-users] queues load balancing

[Matt Riddell]

On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
 Thanks for the idea.
 I will try it this way:

 exten =  123,1,Ringing
 exten =  123,2,Wait(1)
 exten =  123,3,Answer
 exten =  123,4,Random(33:123,10)
 exten =  123,5,Queue(queue_1)
 exten =  123,6,Hangup

 exten =  123,10,Queue(queue_2)
 exten =  123,11,Hangup

Bear in mind that the Random application has been deprecated in favour 
of the RANDOM function:

asterisk -rx 'show application random'

   -= Info about application 'Random' =-

[Synopsis]
Conditionally branches, based upon a probability

[Description]
Random([probability]:[[context|]extension|]priority)
   probability := INTEGER in the range 1 to 100
DEPRECATED: Use GotoIf($[${RAND(1,100)}  number]?label)

-- 
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] queues load balancing

[Danny Nicholas]

Here is a brute force solution:
[global]
CALLCOUNT=0
 exten =  123,1,Ringing
 exten =  123,2,Wait(1)
 exten =  123,3,Answer
 exten =  123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
 exten =  123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
 exten =  123,6,Queue(queue_1)
 exten =  123,7,Hangup
 exten =  123,8(queue2),Set(CALLCOUNT=0)
 exten =  123,9,Queue(queue_2)
 exten =  123,10,Hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, July 20, 2009 7:37 AM
To: gomespere...@startel.pt; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] queues load balancing

On 21/7/09 12:08 AM, Joao Gomes Pereira wrote:
 Thanks for the idea.
 I will try it this way:

 exten =  123,1,Ringing
 exten =  123,2,Wait(1)
 exten =  123,3,Answer
 exten =  123,4,Random(33:123,10)
 exten =  123,5,Queue(queue_1)
 exten =  123,6,Hangup

 exten =  123,10,Queue(queue_2)
 exten =  123,11,Hangup

Bear in mind that the Random application has been deprecated in favour 
of the RANDOM function:

asterisk -rx 'show application random'

   -= Info about application 'Random' =-

[Synopsis]
Conditionally branches, based upon a probability

[Description]
Random([probability]:[[context|]extension|]priority)
   probability := INTEGER in the range 1 to 100
DEPRECATED: Use GotoIf($[${RAND(1,100)}  number]?label)

-- 
Cheers,

Matt Riddell
Director
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Re: [asterisk-users] queues load balancing

[Philipp Kempgen]

Danny Nicholas schrieb:
 Here is a brute force solution:
 [global]
 CALLCOUNT=0
 exten =  123,1,Ringing
 exten =  123,2,Wait(1)
 exten =  123,3,Answer
 exten =  123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)

...,Set(CALLCOUNT=$[${CALLCOUNT} + 1])
or
...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)})

 exten =  123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
 exten =  123,6,Queue(queue_1)
 exten =  123,7,Hangup
 exten =  123,8(queue2),Set(CALLCOUNT=0)
 exten =  123,9,Queue(queue_2)
 exten =  123,10,Hangup


Philipp Kempgen
-- 
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -  http://www.amoocon.de
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Re: [asterisk-users] Queues recording CDR

[Kurian Thayil]

Hi,

My apologies Nicolas, a mistake from my part. And I appreciate for
correcting me. Asternic is a  good piece of work.

Regards,

Kurian Thayil.

On Mon, 2009-07-06 at 09:41 -0300, Nicolás Gudiño wrote:
 Hello,
 
 Just a correction, Asternic Call Center Stats is not from
 asteriskguru. Asteriskguru has its own statistic program that is not
 open source, but free to use. Asternic was written by me (not
 asteriskguru) and has an open source version and a commercial one.
 
 Best regards,
 
 --
 Nicolás Gudiño
 Buenos Aires - Argentina
 
 
 
 On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote:
  Hi Sriram,
 
  1. Set the channel variable MonitorFilename before Queue() in dialplan
  and you can give some meaningful filename for record.
  2. I guess you can use an AGI to capture events and then integrate this
  with a DB in the Backend. This should help you to track the activity.
  3. asternic from asteriskguru is kind of OK. Gives you a live and
  detailed report. Parses the queue_log to the MySQL DB and works. This
  parse program could be used in your AGI which I mentioned in point 2.
 
  Hope this helps.
 
  Regards,
 
  Kurian Thayil.
 
  On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
  Hi
 
  1. I want to record all calls that land to an agent via a queue using
  a meaningful name - as of now i name the recorded file on the fly
  using {CALLERID} variable so that the file gets stored using the
  caller id iunder /var/spool/asterisk/monitor , now if i want to store
  it as CALLERIDEXTEN where call landed from queue how can i do
  this ?
  2. I have a CDR issue - when A calls he is put in Queue and say he is
  answered by Agent B ..Agent B transfers the Call to agent C as it is
  to Agent C whom A wants to talk..when the call gets d/c the CDR for
  that call shows the destination field as B whereas it shd be C...how
  do i take care of this ...in my call center agents are paid on the
  basis of talk time on inbound calls - this way an agent who just
  transfers calls is at merry !!
  3. Are their any GPL based queue reporting software - hows the
  asterisk queue statistics program from asteriskguru.com has anyone
  tried it ?
 
  Thanks
  Sriram
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Re: [asterisk-users] Queues recording CDR

[Nicolás Gudiño]

Hello,

Just a correction, Asternic Call Center Stats is not from
asteriskguru. Asteriskguru has its own statistic program that is not
open source, but free to use. Asternic was written by me (not
asteriskguru) and has an open source version and a commercial one.

Best regards,

--
Nicolás Gudiño
Buenos Aires - Argentina



On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote:
 Hi Sriram,

 1. Set the channel variable MonitorFilename before Queue() in dialplan
 and you can give some meaningful filename for record.
 2. I guess you can use an AGI to capture events and then integrate this
 with a DB in the Backend. This should help you to track the activity.
 3. asternic from asteriskguru is kind of OK. Gives you a live and
 detailed report. Parses the queue_log to the MySQL DB and works. This
 parse program could be used in your AGI which I mentioned in point 2.

 Hope this helps.

 Regards,

 Kurian Thayil.

 On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
 Hi

 1. I want to record all calls that land to an agent via a queue using
 a meaningful name - as of now i name the recorded file on the fly
 using {CALLERID} variable so that the file gets stored using the
 caller id iunder /var/spool/asterisk/monitor , now if i want to store
 it as CALLERIDEXTEN where call landed from queue how can i do
 this ?
 2. I have a CDR issue - when A calls he is put in Queue and say he is
 answered by Agent B ..Agent B transfers the Call to agent C as it is
 to Agent C whom A wants to talk..when the call gets d/c the CDR for
 that call shows the destination field as B whereas it shd be C...how
 do i take care of this ...in my call center agents are paid on the
 basis of talk time on inbound calls - this way an agent who just
 transfers calls is at merry !!
 3. Are their any GPL based queue reporting software - hows the
 asterisk queue statistics program from asteriskguru.com has anyone
 tried it ?

 Thanks
 Sriram
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 --
 Kurian Mathew Thayil.
 (GPG KeyID: E232394F)

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[asterisk-users] Queues recording CDR

[Sriram]

Hi

1. I want to record all calls that land to an agent via a queue using a 
meaningful name - as of now i name the recorded file on the fly using 
{CALLERID} variable so that the file gets stored using the caller id iunder 
/var/spool/asterisk/monitor , now if i want to store it as CALLERIDEXTEN 
where call landed from queue how can i do this ?
2. I have a CDR issue - when A calls he is put in Queue and say he is answered 
by Agent B ..Agent B transfers the Call to agent C as it is to Agent C whom A 
wants to talk..when the call gets d/c the CDR for that call shows the 
destination field as B whereas it shd be C...how do i take care of this ...in 
my call center agents are paid on the basis of talk time on inbound calls - 
this way an agent who just transfers calls is at merry !!
3. Are their any GPL based queue reporting software - hows the asterisk queue 
statistics program from asteriskguru.com has anyone tried it ?

Thanks
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Re: [asterisk-users] Queues recording CDR

[Kurian Thayil]

Hi Sriram,

1. Set the channel variable MonitorFilename before Queue() in dialplan
and you can give some meaningful filename for record.
2. I guess you can use an AGI to capture events and then integrate this
with a DB in the Backend. This should help you to track the activity.
3. asternic from asteriskguru is kind of OK. Gives you a live and
detailed report. Parses the queue_log to the MySQL DB and works. This
parse program could be used in your AGI which I mentioned in point 2.

Hope this helps.

Regards,

Kurian Thayil.

On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
 Hi
  
 1. I want to record all calls that land to an agent via a queue using
 a meaningful name - as of now i name the recorded file on the fly
 using {CALLERID} variable so that the file gets stored using the
 caller id iunder /var/spool/asterisk/monitor , now if i want to store
 it as CALLERIDEXTEN where call landed from queue how can i do
 this ?
 2. I have a CDR issue - when A calls he is put in Queue and say he is
 answered by Agent B ..Agent B transfers the Call to agent C as it is
 to Agent C whom A wants to talk..when the call gets d/c the CDR for
 that call shows the destination field as B whereas it shd be C...how
 do i take care of this ...in my call center agents are paid on the
 basis of talk time on inbound calls - this way an agent who just
 transfers calls is at merry !!
 3. Are their any GPL based queue reporting software - hows the
 asterisk queue statistics program from asteriskguru.com has anyone
 tried it ?
  
 Thanks
 Sriram
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[asterisk-users] Queues in memory after startup

[Gabriel Ortiz Lour]

Hi all,

  After * starts the command queue show would not show any of the realtime
queues, but just the ones that are in the queues.conf file. In this state de
AMI would not send any QueueMemberStatus for that queues until a call is
received by that realtime queue.

  Anyone knows any whay to load this information in *'s memory without the
need of the queue receiving a call?

Thanks,
Gabriel Ortiz
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Re: [asterisk-users] Queues in memory after startup

[Mark Michelson]

Gabriel Ortiz Lour wrote:
 Hi all,
 
   After * starts the command queue show would not show any of the 
 realtime queues, but just the ones that are in the queues.conf file. In 
 this state de AMI would not send any QueueMemberStatus for that queues 
 until a call is received by that realtime queue.
 
   Anyone knows any whay to load this information in *'s memory without 
 the need of the queue receiving a call?
 
 Thanks,
 Gabriel Ortiz
 

There is a bit of a hack you can use. If you instead use the command queue 
show 
some_specific_queue_name then Asterisk will load from realtime. Then type 
queue show again and you'll see all the queues. I'm not sure why it was 
written this way. If you use any 1.6 version of Asterisk, you will find that it 
does not behave this way. queue show will always show all queues.

Mark Michelson

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Re: [asterisk-users] Queues in memory after startup

[Miguel Molina]

Mark Michelson escribió:

Gabriel Ortiz Lour wrote:
  

Hi all,

  After * starts the command queue show would not show any of the 
realtime queues, but just the ones that are in the queues.conf file. In 
this state de AMI would not send any QueueMemberStatus for that queues 
until a call is received by that realtime queue.


  Anyone knows any whay to load this information in *'s memory without 
the need of the queue receiving a call?


Thanks,
Gabriel Ortiz




There is a bit of a hack you can use. If you instead use the command queue show 
some_specific_queue_name then Asterisk will load from realtime. Then type 
queue show again and you'll see all the queues. I'm not sure why it was 
written this way. If you use any 1.6 version of Asterisk, you will find that it 
does not behave this way. queue show will always show all queues.


Mark Michelson

  
I was noticing the same behavior with 1.4 realtime queues. I thought 
this was the standard, but it's good to know that 1.6 behaves well, no 
matter if the queue is static from the queues.conf file or defined in 
realtime way. Some of this little improvements makes 1.6 a bit more 
attractive to try for us (1.4 users). Moreover, some of the changes made 
to trunk to improve IAX2 performance, and some other big changes are 
just awesome.


Thanks.

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 

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[asterisk-users] Queues Announce help request.

[Cary Fitch]

I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some comforting voice announcements would be nice.

Below is the queues.conf file relevant portions.

Member phone number is munged to protect the guilty.

We shouldn't need the announcement source info, but I have been trying
everything.

The problem is with the member busy, we get no voice announcements.
(For test purposes is being on hold busy?  We have also just laid the
phone on the desk.)

We will settle for expected hold time, Thank you announcements, Position in
queue, or Dow Jones 30 Industrials news. :-)

Anyone have a tip?

Cary Fitch

Here are some relevant errors from console.

[Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite:
Re-invite to non-existing call leg on other UA. SIP dialog
'495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up.
-- SIP/3617001000-009a3930 is circuit-busy
-- Nobody picked up in 6 ms
-- Exiting on time-out cycle
-

[Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12949 handle_response: Remote
host can't match request CANCEL to call
'495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up.
-- Stopped music on hold on SIP/3617001401-fc0359f0


[usa-queue]
queue-youarenext = queue-youarenext
queue-thereare =  there-thereare
queue-callswaiting = queue-callswaiting
queue-holdtime = queue-holdtime
queue-thankyou =  queue-thankyou
music = default
maxlen = 0
strategy = ringall
context = leave-message
periodic-announce = thank-you-message
periodic-announce-frequency = 30
announce-frequency = 30
announce-holdtime = yes
announce-round-seconds = 10
joinempty = no
leavewhenempty = yes
retry = 30
timeout = 300
member = SIP/3617001402



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Re: [asterisk-users] Queues Announce help request.

[Danny Nicholas]

The most popular answer I've seen here is to replace the regular music with
a streamed audio feed which can be anything you have access to.  I'd try and
give you details, but they wouldn't be correct.  This information is pretty
easy to locate in the digium site, viop-info.org or google.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Friday, March 20, 2009 12:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Queues Announce help request.

I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some comforting voice announcements would be nice.

Below is the queues.conf file relevant portions.

Member phone number is munged to protect the guilty.

We shouldn't need the announcement source info, but I have been trying
everything.

The problem is with the member busy, we get no voice announcements.
(For test purposes is being on hold busy?  We have also just laid the
phone on the desk.)

We will settle for expected hold time, Thank you announcements, Position in
queue, or Dow Jones 30 Industrials news. :-)

Anyone have a tip?

Cary Fitch

Here are some relevant errors from console.

[Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12406 handle_response_invite:
Re-invite to non-existing call leg on other UA. SIP dialog
'495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up.
-- SIP/3617001000-009a3930 is circuit-busy
-- Nobody picked up in 6 ms
-- Exiting on time-out cycle
-

[Mar 20 12:16:16] WARNING[4317]: chan_sip.c:12949 handle_response: Remote
host can't match request CANCEL to call
'495f59f65dec70c67849cf4f0cd09...@72.49.176.4'. Giving up.
-- Stopped music on hold on SIP/3617001401-fc0359f0


[usa-queue]
queue-youarenext = queue-youarenext
queue-thereare =  there-thereare
queue-callswaiting = queue-callswaiting
queue-holdtime = queue-holdtime
queue-thankyou =  queue-thankyou
music = default
maxlen = 0
strategy = ringall
context = leave-message
periodic-announce = thank-you-message
periodic-announce-frequency = 30
announce-frequency = 30
announce-holdtime = yes
announce-round-seconds = 10
joinempty = no
leavewhenempty = yes
retry = 30
timeout = 300
member = SIP/3617001402



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Re: [asterisk-users] Queues Announce help request.

[Jared Smith]

- Cary Fitch ca...@usawide.net wrote:
 I am trying to get a queue to do more than just play music and hold
 calls.
 Specifically, making some comforting voice announcements would be
 nice.

You may want to take the quotes off of the filenames in your queues.conf config 
file... they're not needed, and could very well be causing your problems.

--
Jared Smith
Training Manager
Digium, Inc.

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Re: [asterisk-users] Queues Announce help request.

[Matt Florell]

On 3/20/09, Cary Fitch ca...@usawide.net wrote:
 I am trying to get a queue to do more than just play music and hold calls.
  Specifically, making some comforting voice announcements would be nice.

  Below is the queues.conf file relevant portions.

  Member phone number is munged to protect the guilty.

  We shouldn't need the announcement source info, but I have been trying
  everything.

  The problem is with the member busy, we get no voice announcements.
  (For test purposes is being on hold busy?  We have also just laid the
  phone on the desk.)

  We will settle for expected hold time, Thank you announcements, Position in
  queue, or Dow Jones 30 Industrials news. :-)

  Anyone have a tip?

  Cary Fitch

I just thought I'd mention that ViciDial has the ability to play a
periodic announcement on inbound queue calls, as well as music on
hold, place in line, estimated hold time and lots of other
inbound-only features.

ViciDial does not use Asterisk Queues so the way we got it working
probably wouldn't help you much, but I just wanted to mention it's
functionality and that it is open source if you wanted to give it a
try.

Thanks,

MATT---

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[asterisk-users] * Queues with legacy pbx extensions ?

[Sriram]

Hello Everybody

I am using Trixbox 2.4 (with TE420P  PRI lines) .. my setup is like Calls 
--Asterisk--legacy pbx---analog extensions(agents). Whenver a call comes in 
, asterisk dials the ACD number of the legacy pbx which in turn decides to 
route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span 
and 5432 is the ACD number of legacy pbx under which agents like 102,103,104 
are present. Now my question is  (might be very silly) :

1. I dont have a queue stats software for my legacy pbx, the agents stil logon 
and logout on their legacy pbx..I was wondering if its possible that all these 
102,103,104 etc login to a queue which resides on asterisk ? Can they login and 
logout of the asterisk queue..I am asking this since there are very nice 
asterisk queue reporting utilities out there for free..and i wud like to use 
them ...this will also help me using screen pop-ups for my agents...i know SIP 
extensions can simplify my setup but echo problems scare me also i dont want to 
throw my legacy pbx on which i invested heavily..

Can anyone throw some pointers ?

Thanks in advance

Sriram  ___
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Re: [asterisk-users] * Queues with legacy pbx extensions ?

[David fire]

hi
we ALWAYS use
sip phone IP*-E1-pstn
or
sip phone-IP-*-E1-legasypbx-?- pstn

we use the digiums cards whit echo canceller and we havent any echo problem.

more than 20 or 30 installations.
whit almost every provider in the country.

dont be SO scared.
David

2009/1/26 Sriram d_r_sri...@hotmail.com

  Hello Everybody

 I am using Trixbox 2.4 (with TE420P  PRI lines) .. my setup is like Calls
 --Asterisk--legacy pbx---analog extensions(agents). Whenver a call comes
 in , asterisk dials the ACD number of the legacy pbx which in turn decides
 to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the
 4th span and 5432 is the ACD number of legacy pbx under which agents like
 102,103,104 are present. Now my question is  (might be very silly) :

 1. I dont have a queue stats software for my legacy pbx, the agents stil
 logon and logout on their legacy pbx..I was wondering if its possible that
 all these 102,103,104 etc login to a queue which resides on asterisk ? Can
 they login and logout of the asterisk queue..I am asking this since there
 are very nice asterisk queue reporting utilities out there for free..and i
 wud like to use them ...this will also help me using screen pop-ups for my
 agents...i know SIP extensions can simplify my setup but echo problems scare
 me also i dont want to throw my legacy pbx on which i invested heavily..

 Can anyone throw some pointers ?

 Thanks in advance

 Sriram

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-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
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[asterisk-users] Queues, SIP channel and In Use

[Benoit]

Hi,

I'm a little surprised, up until 1.4.22 my agents where using an IAX
channel to ZoIPer Softphone,
however since after the upgrade to .22 we experienced a problem with
hangup failure between zoiper
 and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
made them switch to SIP
instead.

Weird thing is that the 'Not In Use' warning message keep showing
(WARNING[1863]: app_queue.c:3136 try_calling: The device state of this
queue member, Agent/136, is still 'Not in Use' when it probably should
not be! Please check UPGRADE.txt for correct configuration settings.)
However, when i look at the queue a few sec after the Agent is marked
'in use' which wasn't the case with IAX iirc

Agent are defined using a Local channel, but we used the '/n' flag to
passthru the status:
Agent/136 (Local/1...@queues/n) ..

As for the SIP peer definition i have the limitonpeer=yes in the general
section and all peers are templated based on this:

[poste](!)
type=friend
host=dynamic
qualify=yes
call-limit=6


Is their something more in can do to avoid the warning ?

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Re: [asterisk-users] Queues, SIP channel and In Use

[Mark Michelson]

Benoit wrote:
 Hi,
 
 I'm a little surprised, up until 1.4.22 my agents where using an IAX
 channel to ZoIPer Softphone,
 however since after the upgrade to .22 we experienced a problem with
 hangup failure between zoiper
  and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
 made them switch to SIP
 instead.
 
 Weird thing is that the 'Not In Use' warning message keep showing
 (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this
 queue member, Agent/136, is still 'Not in Use' when it probably should
 not be! Please check UPGRADE.txt for correct configuration settings.)
 However, when i look at the queue a few sec after the Agent is marked
 'in use' which wasn't the case with IAX iirc
 
 Agent are defined using a Local channel, but we used the '/n' flag to
 passthru the status:
 Agent/136 (Local/1...@queues/n) ..
 
 As for the SIP peer definition i have the limitonpeer=yes in the general
 section and all peers are templated based on this:
 
 [poste](!)
 type=friend
 host=dynamic
 qualify=yes
 call-limit=6
 
 
 Is their something more in can do to avoid the warning ?

I believe the use of the Local channel is what is causing the warning to 
appear. 
The problem is that the device state is not updated until after app_queue 
checks 
to see if it should display that warning. This has been brought up before, but 
since it doesn't actually adversely affect the operation of the queue, not much 
has been done. The worst you have to worry about is that warning. If you find 
the warning to be annoying, the best I can offer you is to either not log 
warnings (a bad idea, imho) or just remove that line of code from the source.

Mark Michelson

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Re: [asterisk-users] Queues, SIP channel and In Use

[Benoit]

Mark Michelson a écrit :
 Benoit wrote:
   
 Hi,

 I'm a little surprised, up until 1.4.22 my agents where using an IAX
 channel to ZoIPer Softphone,
 however since after the upgrade to .22 we experienced a problem with
 hangup failure between zoiper
  and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i
 made them switch to SIP
 instead.

 Weird thing is that the 'Not In Use' warning message keep showing
 (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this
 queue member, Agent/136, is still 'Not in Use' when it probably should
 not be! Please check UPGRADE.txt for correct configuration settings.)
 However, when i look at the queue a few sec after the Agent is marked
 'in use' which wasn't the case with IAX iirc

 Agent are defined using a Local channel, but we used the '/n' flag to
 passthru the status:
 Agent/136 (Local/1...@queues/n) ..

 As for the SIP peer definition i have the limitonpeer=yes in the general
 section and all peers are templated based on this:

 [poste](!)
 type=friend
 host=dynamic
 qualify=yes
 call-limit=6


 Is their something more in can do to avoid the warning ?
 

 I believe the use of the Local channel is what is causing the warning to 
 appear. 
 The problem is that the device state is not updated until after app_queue 
 checks 
 to see if it should display that warning. This has been brought up before, 
 but 
 since it doesn't actually adversely affect the operation of the queue, not 
 much 
 has been done. The worst you have to worry about is that warning. If you find 
 the warning to be annoying, the best I can offer you is to either not log 
 warnings (a bad idea, imho) or just remove that line of code from the source.

 Mark Michelson
   
No it's fine for me, i was just wondering if i missed something
thanks

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[asterisk-users] queues and MEMBERINTERFACE for AGI script

[Thomas Winter]

Hi,
iam using and queue and starting an AGI script after caller connected to 
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to 
be not work, either as variable in the queue command and also not in the AGI 
script.
How to found out which agent is connected to calling channel?

I try to avoid to using LOCAL channels, because I like the function ringinuse.

regards
Thomas


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Re: [asterisk-users] queues and MEMBERINTERFACE for AGI script

[David Van Ginneken]

Thomas Winter wrote:
 Hi,
 iam using and queue and starting an AGI script after caller connected to 
 agent.
 How to find out in the script the connected agent, MEMBERINTERFACE seemed to 
 be not work, either as variable in the queue command and also not in the AGI 
 script.
 How to found out which agent is connected to calling channel?

 I try to avoid to using LOCAL channels, because I like the function ringinuse.

 regards
 Thomas


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Thomas,

In queues.conf you need to set the variable setinterfacevar=yes

You'd then pass the AGI to the Queue application with something like this:

exten = some_extension,n,Queue(somequeue,some.agi)

Then within the AGI you'd retrieve the variable like this:

my $memberinterface = $AGI-get_variable('MEMBERINTERFACE'); # Perl example.

Hope this helps.

-Dave


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Re: [asterisk-users] Queues, monitor-join=yes, and volume

[Asterisk]

Thanks. If I find out some settings for soxmix, do you maybe know where can I 
change Asterisk settings for soxmix (parameters)?

Regards, Alex

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Backeberg
Sent: Tuesday, May 13, 2008 5:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume

On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
  Is there any way to modify the volume (either lower the volume of the 
 clients, or increase the volume of the agents) while doing the join of the 
 -in and -out files into one recording?

Uh-huh. Read the documentation for soxmix.

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Re: [asterisk-users] Queues, monitor-join=yes, and volume

[David Backeberg]

You can call the sox binary directly from your dialplan, or any other
binary that fits your needs. If you post your dialplan where you're
doing the recording, we can give input about where to put the calls to
sox.

On Wed, May 14, 2008 at 4:16 AM, Asterisk [EMAIL PROTECTED] wrote:
 Thanks. If I find out some settings for soxmix, do you maybe know where can I 
 change Asterisk settings for soxmix (parameters)?

  Regards, Alex



  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David 
 Backeberg
  Sent: Tuesday, May 13, 2008 5:35 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Queues, monitor-join=yes, and volume

  On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the 
 clients, or increase the volume of the agents) while doing the join of the 
 -in and -out files into one recording?

  Uh-huh. Read the documentation for soxmix.



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[asterisk-users] Queues, monitor-join=yes, and volume

[Asterisk]

Hi guys,

I have few queues configured on my PBX that have:

monitor-format=wav
monitor-join=yes

My problem is that the volume of the clients who are calling into these queues 
is slightly higher than the volume of the agents.

Is there any way to modify the volume (either lower the volume of the clients, 
or increase the volume of the agents) while doing the join of the -in and 
-out files into one recording?

I'm using version 1.2.9, which I think uses soxmix for joining the files?

Thanks, Alex

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