Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered.I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *-------------------------------------------------------* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110212/3513ffa1/attachment.htm>
as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers, () and divert it to the defined context for queue. for example test.call Channel: SIP/customer number at your carrier Context: your queue context ..... ask if you need more info best On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada <listas_quijada at hotmail.com>wrote:> Hi! > I have a script to generate calls from a database using .call files and > giving a message. If works great! but now I need to do the same but instead > of play a recorded message I need transfer this call to live person in a > specfic extension. > This is the scenario > I have a webpage with information about a customer so in this page the > agent click a phone number and asterisk do the call and transfer the call to > agent if this call is answered. > I did the page and everything but when I do the clicktodial I dont know how > transfer the call to this agent. I ask the extension and user before login > so I know what agent is in each extension to transfer the call to rigth > agent. > > Anybody can give an idea ? > TIA > > > *-------------------------------------------------------* > *-Edwin Quijada > *-Developer DataBase > *-JQ Microsistemas > *-Soporte PostgreSQL > *-www.jqmicrosistemas.com > *-809-849-8087 > *-------------------------------------------------------* > > > > > -- > _____________________________________________________________________ > -- 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/20110212/c0d2f329/attachment.htm>
On Sat, Feb 12, 2011 at 04:23:00PM +0000, Edwin Quijada wrote:>I have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered.Usually it's the other way round: the agent's phone rings, and when he picks it up the other end gets dialled. That's trivial with call files: Channel: (local channel ID for agent) Context: (context for calling local channel) Extension: (remote party's phone number)
My problem is that I dont know how to do for transfer the call to agentExample, I have this .call Channel: Zap/g1/8652323454MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 10 So my context is this [call-file-test ]exten => 10,1,Dial(SIP/2031,tT)exten => 10,2,hangup In this case I call the number 8652323454 if the call is connect this call in the context call-file-test uisng extension 10 for tranfering this call to extension 2031, but this doesnt work. The call file works fine but when I try to transfer the call I get an error Any help ? *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *-------------------------------------------------------* From: lopl at lopl.net Date: Sat, 12 Feb 2011 21:22:50 +0330 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers, () and divert it to the defined context for queue. for example test.call Channel: SIP/customer number at your carrier Context: your queue context..... ask if you need more infobest On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada <listas_quijada at hotmail.com> wrote: Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *-------------------------------------------------------* -- _____________________________________________________________________ -- 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/20110212/a6d30430/attachment.htm>