[asterisk-users] Asterisk Queue's

[Atis Lezdins]

On Wed, Sep 3, 2008 at 9:42 AM, Tobias Ahlander <plyschen at gmail.com> wrote:
>>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
>>
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>>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.

You have to set it to any value, so that device state events are
generated, so set it to 10 or 20 to have no actual limit.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835