[Asterisk-Users] Help on queues

[Zach A]

Thanks johann for your reply and examples. I'll try those and will go
through wiki again.

Zach A


-----Original Message-----
From: Johann [mailto:johann.hoehn at ecommerce.com] 
Sent: Monday, February 06, 2006 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help on queues

See below...

Zach A wrote:
> I need practical examples showing solutions to various solutions, e.g.
> how can a caller leave a queue and go back to the main menu instead of
> hanging up and redialing,

extensions.conf:

[menu]
exten => s,1,Background(menu-message)
exten => 1,1,Queue(aqueue,,,,300)
exten => 2,1,Queue(someotherqueue,,,,300)
exten => t,1,Goto(s,1)

Above is a very simple approach.  Plays the menu and has options for 1
and 2. 
Both enter queues and have a timeout option of 300 seconds (if they dont
get 
anywhere).  The timeout extension plays the menu again and is hit if
they 
timeout of a queue and if they fail to hit an option.  There is nothing
to force 
the call to be hung up so use with caution for calls you pay for(or add
checks 
to force a call to be hung up after a certain time or after X amount of
menus 
prompts have been played).

Look at the output of "show application queue" for more information or
read the 
wiki.

> or how can a queue be started for an
> extension, i.e. if 3-4 callers dial 201 and 201 is busy, instead of
> sending calls to voice mails, start a queue and let them wait in
queue.

Afaik, you can not dynamically create queues.  So every extension that
wants to 
have a queue will need to have it defined in queues.conf (or use a few
queues 
and try to keep track of which extension they are going to be used for).
You 
can however dynamically add and remove members from a queue.  In
asterisk 
members of a queue are people that would get answer the call.  Below is
a very 
simple example with the extension 123 on a SIP phone.  Depending on how
your SIP 
phone treats calls this may not work well(ie if it does not return busy
for 
additional incoming calls).

extensions.conf:

[staff]
exten => 123,Queue(personalqueue)

In your queues.conf file have:

[personalqueue]
member => SIP/123

The phone SIP/123, is always a member of personalqueue.  Calls to staff
are 
handled by the staff context(or whatever you named it) that simply put
the 
caller into the queue.  If the person is available they call will ring
them 
right away, if not then they wait with music on hold.  There are lots of
options 
and plenty of documentation in the wiki and source for the config files
to see 
all the options that were left off.

These are very simply examples and may not work in all cases and will
likely 
need more added depending on what you want to do with them...


--johann

> 
> Zeeshan A Zakaria
> 
> 
> -----Original Message-----
> From: jan.sarin at securia.se [mailto:jan.sarin at securia.se] 
> Sent: Monday, February 06, 2006 12:52 PM
> To: asterisk-users at lists.digium.com
> Subject: SV: [Asterisk-Users] Help on queues
> 
> What kind of help do you need then?
> 
> Regards,
> Jan
> 
> 
> -----Ursprungligt meddelande-----
> Från: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] För Zach A
> Skickat: den 6 februari 2006 18:31
> Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Ämne: RE: [Asterisk-Users] Help on queues
> 
> There is no good help on wiki and voip-info.org, I've gone through it
> already.
> 
> Zach
> 
> 
> -----Original Message-----
> From: Dovid Bender [mailto:asteriskdigium at yahoo.com]
> Sent: Monday, February 06, 2006 11:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Help on queues
> 
> Yes. The wiki and voip-info.org
> --- Zach A <zeeshan at acabling.com> wrote:
> 
> 
>>Hi,
>>
>>Is there any detailed guide/tutorial source online on queues?
>>
>>Zach
>>
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