asterisk users - Aug 2008 - Asterisk Queue's

Hello List,

I have a sample queue with two dynamic agents. When the first caller calls
in to the system, the first agents phone starts to ring. Then another caller
calls in to the queue, but the other phone doesn't start to ring until the
first agents pick up his queued call.

I want the second call to start ringing on the second agents phone right
away, since he's available.

Here's the output from the queue from the CLI:
kursk*CLI> queue show sales
sales        has 0 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
      SIP/1203 (dynamic) (Not in use) has taken no calls yet
      SIP/1003 (dynamic) (Not in use) has taken no calls yet
   No Callers

And here's the output from the CLI when the calls come in:
    -- Executing [726 at default:1] NoOp("SIP/1303-092637d0",
"Sales Queue") in
new stack
    -- Executing [726 at default:2] Queue("SIP/1303-092637d0",
"sales|t|||1800|") in new stack
    -- Started music on hold, class 'default', on SIP/1303-092637d0
    -- SIP/1003-0926ae50 is ringing
    -- Executing [726 at default:1] NoOp("SIP/1103-09279020",
"Sales Queue") in
new stack
    -- Executing [726 at default:2] Queue("SIP/1103-09279020",
"sales|t|||1800|") in new stack
    -- Started music on hold, class 'default', on SIP/1103-09279020
    -- SIP/1003-0926ae50 answered SIP/1303-092637d0
    -- Stopped music on hold on SIP/1303-092637d0
[Aug 15 14:01:04] ERROR[20347]: chan_sip.c:3192 update_call_counter: Call to
peer '1003' rejected due to usage limit of 1
    -- Couldn't call SIP/1003
[Aug 15 14:01:04] NOTICE[20347]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/1003-09276ef8' not posted
    -- SIP/1203-09290100 is ringing
  == Spawn extension (default, 726, 2) exited non-zero on
'SIP/1303-092637d0'
    -- SIP/1203-09290100 answered SIP/1103-09279020
    -- Stopped music on hold on SIP/1103-09279020
  == Spawn extension (default, 726, 2) exited non-zero on
'SIP/1103-09279020'

Has anyone seen this problem before or have a solution on it? Is it possible
somehow to tell Asterisk to only send one queue'd call to the Agent at the
time?

Thanks,
Best regards,
Tobias
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080828/220a793f/attachment.htm

Tobias Ahlander schrieb:
> I have a sample queue with two dynamic agents. When the first caller calls
> in to the system, the first agents phone starts to ring. Then another
caller
> calls in to the queue, but the other phone doesn't start to ring until
the
> first agents pick up his queued call.
> 
> I want the second call to start ringing on the second agents phone right
> away, since he's available.
> 
> Here's the output from the queue from the CLI:
[...]
> Has anyone seen this problem before or have a solution on it? Is it
possible
> somehow to tell Asterisk to only send one queue'd call to the Agent at
the
> time?
Did you set autofill=yes in queues.conf?


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--

Hello Philipp,

Yes, I have autofill set in queues.conf. I suspect that this behaviour is
because the Polycom phones I use have 2 lines. Has anyone used this function
with polycom phones before? Also, my agents are Dynamic, perhaps this works
better with Static agents?

Here's my queues.conf (with commented lines deleted for easier reading):

[general]
persistentmembers = yes
autofill = yes
monitor-type = MixMonitor

[sales]
strategy = rrmemory
wrapuptime=15


Date: Thu, 28 Aug 2008 13:56:49 +0200
From: Philipp Kempgen <philipp.kempgen at amooma.de>
Subject: Re: [asterisk-users] Asterisk Queue's
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <48B69281.2000804 at amooma.de>
Content-Type: text/plain; charset=ISO-8859-1

Tobias Ahlander schrieb:
> I have a sample queue with two dynamic agents. When the first caller calls
> in to the system, the first agents phone starts to ring. Then another
caller
> calls in to the queue, but the other phone doesn't start to ring until
the
> first agents pick up his queued call.
>
> I want the second call to start ringing on the second agents phone right
> away, since he's available.
>
> Here's the output from the queue from the CLI:
[...]
> Has anyone seen this problem before or have a solution on it? Is it
possible
> somehow to tell Asterisk to only send one queue'd call to the Agent at
the
> time?
Did you set autofill=yes in queues.conf?


  Philipp Kempgen

--
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
--
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080829/83a19fa0/attachment.htm

>Date: Fri, 29 Aug 2008 09:12:12 -0500
>From: Mark Michelson <mmichelson at digium.com>
>Subject: Re: [asterisk-users] Asterisk Queue's
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>    <asterisk-users at lists.digium.com>
>Message-ID: <48B803BC.5060309 at digium.com>
>Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>Tobias Ahlander wrote:
>> Hello Philipp,
>>
>> Yes, I have autofill set in queues.conf. I suspect that this behaviour
>> is because the Polycom phones I use have 2 lines. Has anyone used this
>> function with polycom phones before? Also, my agents are Dynamic,
>> perhaps this works better with Static agents?
>>
>> Here's my queues.conf (with commented lines deleted for easier
reading):
>>
>> [general]
>> autofill = yes
>> monitor-type = MixMonitor
>>
>> [sales]
>> strategy = rrmemory
>> wrapuptime=15
>>
>
>Depending on which Asterisk version you are using, there was a bug in the
queue
>application for some 1.4 releases where the autofill option would only be
set
>properly if it were placed inside a queue. In other words, you may want to
try
>putting autofill=yes inside the [sales] queue in your configuration.
>
>Also, if you're using a version of Asterisk 1.2, autofill is not a valid
option
>and you'll be stuck with the behavior you're seeing.
>
>Mark Michelson
Mark,

Unfortunately this didn't help at all... Anyone else has any tips? Is there
a way to limit the polycom phones to only take one call from the Queue at
the same time? Asterisk version running is 1.4.13

Best regards,
Tobias
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080901/b9f4acb0/attachment.htm

>Date: Tue, 02 Sep 2008 18:08:52 +1200
>From: Paul Crane <paul at venturevoip.com>
>Subject: Re: [asterisk-users] Asterisk Queue's
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>    <asterisk-users at lists.digium.com>
>Message-ID: <48BCD874.3040606 at venturevoip.com>
>Content-Type: text/plain; charset=ISO-8859-1
>
>-----BEGIN PGP SIGNED MESSAGE-----
>Hash: SHA1
>
>Philipp Kempgen wrote:
>> Tobias Ahlander schrieb:
>>
>>>> From: Mark Michelson <mmichelson at digium.com>
>>
>>>> Tobias Ahlander wrote:
>>
>>>>> Yes, I have autofill set in queues.conf. I suspect that
this behaviour
>>>>> is because the Polycom phones I use have 2 lines. Has
anyone used this
>>>>> function with polycom phones before? Also, my agents are
Dynamic,
>>>>> perhaps this works better with Static agents?
>>>>>
>>>>> Here's my queues.conf (with commented lines deleted for
easier
reading):
>>>>>
>>>>> [general]
>>>>> autofill = yes
>>>>> monitor-type = MixMonitor
>>>>>
>>>>> [sales]
>>>>> strategy = rrmemory
>>>>> wrapuptime=15
>>>>>
>>>> Depending on which Asterisk version you are using, there was a
bug in
the
>>> queue
>>>> application for some 1.4 releases where the autofill option
would only
be
>>> set
>>>> properly if it were placed inside a queue. In other words, you
may want
to
>>> try
>>>> putting autofill=yes inside the [sales] queue in your
configuration.
>>>>
>>>> Also, if you're using a version of Asterisk 1.2, autofill
is not a
valid
>>> option
>>>> and you'll be stuck with the behavior you're seeing.
>>
>>> Unfortunately this didn't help at all... Anyone else has any
tips? Is
there
>>> a way to limit the polycom phones to only take one call from the
Queue
at
>>> the same time? Asterisk version running is 1.4.13
>>
>> Maybe the phones have call-waiting enabled?
>> Does it work if you remove the second line?
>>
>>
>>    Philipp Kempgen
>>
>
>Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes.
>
>- --
>Paul Crane
>
>Technical Support Officer
>VentureVoIP Ltd
>John Wickliffe House
>265 Princes Street
>Dunedin
Paul,

This option doesn't help me that much. When I have it enabled, I can't
put a
call on hold and transfer it since Asterisk rejects usage limit to 1.

Philipp,

I'm using Polycom phones. When I set the "Calls Per Line" (which
I'm told is
Call Waiting) I seem to be able to transfer calls etc, but I'm still
noticing the same behaviour with the queues as before.


Any more tricks I can try?

Thanks,
Best regards,
Tobias
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080903/3d2eee7c/attachment.htm

Alex,

Unfortunately these two setting didn't change the behaviour either... Could
it be a bug in the 1.4.13 version I use?

Thanks,
Best regards,
Tobias



Date: Wed, 03 Sep 2008 03:27:26 -0500
From: Alejandro Kauffmann <akauffma at prodigy.net.mx>
Subject: Re: [asterisk-users] Asterisk Queue's
To: Asterisk Users Mailing List - Non-Commercial Discussion
    <asterisk-users at lists.digium.com>
Message-ID: <48BE4A6E.1060409 at prodigy.net.mx>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Tobias Ahlander wrote:
>  >Date: Tue, 02 Sep 2008 18:08:52 +1200
>  >From: Paul Crane <paul at venturevoip.com <mailto:paul at
venturevoip.com>>
>  >Subject: Re: [asterisk-users] Asterisk Queue's
>  >To: Asterisk Users Mailing List - Non-Commercial Discussion
>  >    <asterisk-users at lists.digium.com
> <mailto:asterisk-users at lists.digium.com>>
>  >Message-ID: <48BCD874.3040606 at venturevoip.com
> <mailto:48BCD874.3040606 at venturevoip.com>>
>  >Content-Type: text/plain; charset=ISO-8859-1
>  >
>  >-----BEGIN PGP SIGNED MESSAGE-----
>  >Hash: SHA1
>  >
>  >Philipp Kempgen wrote:
>  >> Tobias Ahlander schrieb:
>  >>
>  >>>> From: Mark Michelson <mmichelson at digium.com
> <mailto:mmichelson at digium.com>>
>  >>
>  >>>> Tobias Ahlander wrote:
>  >>
>  >>>>> Yes, I have autofill set in queues.conf. I suspect
that this
> behaviour
>  >>>>> is because the Polycom phones I use have 2 lines. Has
anyone used
> this
>  >>>>> function with polycom phones before? Also, my agents
are Dynamic,
>  >>>>> perhaps this works better with Static agents?
>  >>>>>
>  >>>>> Here's my queues.conf (with commented lines
deleted for easier
> reading):
>  >>>>>
>  >>>>> [general]
>  >>>>> autofill = yes
>  >>>>> monitor-type = MixMonitor
>  >>>>>
>  >>>>> [sales]
>  >>>>> strategy = rrmemory
>  >>>>> wrapuptime=15
>  >>>>>
>  >>>> Depending on which Asterisk version you are using, there
was a bug
> in the
>  >>> queue
>  >>>> application for some 1.4 releases where the autofill
option would
> only be
>  >>> set
>  >>>> properly if it were placed inside a queue. In other
words, you may
> want to
>  >>> try
>  >>>> putting autofill=yes inside the [sales] queue in your
configuration.
>  >>>>
>  >>>> Also, if you're using a version of Asterisk 1.2,
autofill is not a
> valid
>  >>> option
>  >>>> and you'll be stuck with the behavior you're
seeing.
>  >>
>  >>> Unfortunately this didn't help at all... Anyone else has
any tips?
> Is there
>  >>> a way to limit the polycom phones to only take one call from
the
> Queue at
>  >>> the same time? Asterisk version running is 1.4.13
>  >>
>  >> Maybe the phones have call-waiting enabled?
>  >> Does it work if you remove the second line?
>  >>
>  >>
>  >>    Philipp Kempgen
>  >>
>  >
>  >Try setting the call-limit to 1 in sip.conf as well as limitonpeer to
yes.
>  >
>  >- --
>  >Paul Crane
>  >
>  >Technical Support Officer
>  >VentureVoIP Ltd
>  >John Wickliffe House
>  >265 Princes Street
>  >Dunedin
>
> Paul,
>
> This option doesn't help me that much. When I have it enabled, I
can't
> put a call on hold and transfer it since Asterisk rejects usage limit to
1.
>
> Philipp,
>
> I'm using Polycom phones. When I set the "Calls Per Line"
(which I'm
> told is Call Waiting) I seem to be able to transfer calls etc, but I'm
> still noticing the same behaviour with the queues as before.
>
>
> Any more tricks I can try?
>
Have you tried ringinuse=no in the queue definition in queues.conf and
call-limit=2 in sip.conf?

Alex
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080904/0e19a2a0/attachment.htm