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 <timotsm...@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.ge...@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...@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 >
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