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

Tiago Geada tiago.geada at gmail.com
Sun Aug 4 19:12:58 CDT 2013


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 <timotsmith at 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.geada at 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_ml at 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
> >>>
> >>> --
> >>> _____________________________________________________________________
> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>>                 http://www.asterisk.org/hello
> >>>
> >>> asterisk-users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>     http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >>
> >> --
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>               http://www.asterisk.org/hello
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >                http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130805/9d98cec8/attachment.htm>


More information about the asterisk-users mailing list