[Asterisk-Users] Asterisk Queue strange behaviour

[Henry Devito]

I used agentcallbacklogin app with a extension macro, that way if he was on 
a conference call he could log out and the caller would be sent to his 
voicemail.   Here is another approach though, it is in the WIKI and does 
work
http://www.voip-info.org/wiki-Asterisk+tips+campon
----- Original Message ----- 
From: "Jan Marius Evang" <marius at evang.net>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Tuesday, March 15, 2005 7:26 AM
Subject: Re: [Asterisk-Users] Asterisk Queue strange behaviour


> Hi. Is this different depending on wether I am a "Member" or "Agent"?
>
> I have used "Memeber" and it does not work...
>
> Yours
> Jan Marius Evang
>
>> I just did this for a customer.  All I did was create a queue just for
>> him,
>> he is the only agent in the queue.  * acts just like you want if you are
>> the
>> only person in the queue.
>> ----- Original Message -----
>> From: "Jan Marius Evang" <marius at evang.net>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Tuesday, March 15, 2005 6:19 AM
>> Subject: [Asterisk-Users] Asterisk Queue strange behaviour
>>
>>
>>> Hi.
>>> I have a problem which I assume would be easy to fix, but I can't find
>>> anything about it...
>>>
>>> I wish to have people dialing my phone, and if it is busy, they are put
>>> into a queue. And then I am dialed back when the previous call is
>>> finished, and connected to the waiting caller.
>>>
>>> Easy enough?
>>> ----------exten
>>> exten => 6,1,Background(salesq-intro);
>>> exten => 6,2,Queue(salesq|tT|||300);
>>>
>>> ----------queue
>>> [salesq]
>>> music = default
>>> context=default
>>> Member => SIP/99559088 at future-out
>>>
>>> ----------
>>>
>>> So.. what is the problem... The first caller gets connected to my
>>> (mobile)
>>> phone. The second and all other caller gets transferred to my phone's
>>> mailbox... Which is not what I intended.
>>>
>>> I tried the same using a SIP softphone as my Member, And all calls kept
>>> being sent to the softphone, even when I was busy talking...
>>>
>>> So.. I guess I have a few options...
>>> - Instead of putting my SIP directly as a member, put an extension that
>>> will raise a flag when called, drop it when hanging up, and if called
>>> when
>>> the flag is up, give a busy()? or return to queue, is this possible?
>>> Problem: If my phone is busy from a call not coming through the asterisk
>>> server...
>>>
>>> - I could change my mobile phone mailbox to use asterisk as my "normal"
>>> mailbox. Could I then detect that this is a call commnig from the Queue
>>> and put it back into the queue or return busy()? (And use it as a normal
>>> mailbox when not coming from the queue?)
>>>
>>> - I could put some "magic" into my mobile phone voicemail welcome
>>> message
>>> that tells asterisk to put the call back into the Queue?
>>>
>>> Would any of these help? Or is there a way of specifying that an
>>> extension
>>> or a channel can only be used by one call at a time?
>>>
>>> Please Help...
>>>
>>> Yours
>>> Jan Marius Evang
>>>
>>>
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>>
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>
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