Re: [asterisk-users] Using files .call or AMI
On Mon, Feb 14, 2011 at 04:06:10AM +, Edwin Quijada wrote: >How would be the dialplan for this context from-lan ??? This list is for non-commercial support. If you want someone to do the work for you, I suggest you go elsewhere and offer money. -- _ -- 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
Re: [asterisk-users] Using files .call or AMI
How would be the dialplan for this context from-lan ??? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* > Date: Sat, 12 Feb 2011 23:20:11 + > From: ro...@firedrake.org > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Using files .call or AMI > > On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: > >This works for me.! but the agent has to dial the number ? > >How could be the context for do this ? U can give an example ? > > I'm using this to place calls from local IP-phones over the PSTN. So my > script will generate, say: > > Channel: SIP/lanphone > Context: from-lan > Extension: 08001234567 > > taking the 0800... from the list of customer details. > > SIP/lanphone is the ID of the "originating" phone. Extension is the > sequence the agent would dial if he were placing the call himself. > The "originating" phone rings; when it's picked up, the Asterisk server > calls the "Extension" number and bridges the two calls, so the local > agent hears ringing tones from the far end. All the agent has to do is > pick up the phone when it rings and put it down when the call is over. > > -- > _ > -- 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
Re: [asterisk-users] Using files .call or AMI
Thks, now I understand for your cooperation.TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* > Date: Sat, 12 Feb 2011 23:20:11 + > From: ro...@firedrake.org > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Using files .call or AMI > > On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: > >This works for me.! but the agent has to dial the number ? > >How could be the context for do this ? U can give an example ? > > I'm using this to place calls from local IP-phones over the PSTN. So my > script will generate, say: > > Channel: SIP/lanphone > Context: from-lan > Extension: 08001234567 > > taking the 0800... from the list of customer details. > > SIP/lanphone is the ID of the "originating" phone. Extension is the > sequence the agent would dial if he were placing the call himself. > The "originating" phone rings; when it's picked up, the Asterisk server > calls the "Extension" number and bridges the two calls, so the local > agent hears ringing tones from the far end. All the agent has to do is > pick up the phone when it rings and put it down when the call is over. > > -- > _ > -- 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
Re: [asterisk-users] Using files .call or AMI
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: >This works for me.! but the agent has to dial the number ? >How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the "originating" phone. Extension is the sequence the agent would dial if he were placing the call himself. The "originating" phone rings; when it's picked up, the Asterisk server calls the "Extension" number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- 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
Re: [asterisk-users] Using files .call or AMI
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: l...@lopl.net Date: Sat, 12 Feb 2011 21:22:50 +0330 To: asterisk-users@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@your carrier Context: your queue context. ask if you need more infobest On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada 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 -- _ -- 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
Re: [asterisk-users] Using files .call or AMI
> Date: Sat, 12 Feb 2011 21:35:29 + > From: ro...@firedrake.org > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Using files .call or AMI > > On Sat, Feb 12, 2011 at 04:23:00PM +, 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) This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? TIA -- _ -- 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
Re: [asterisk-users] Using files .call or AMI
On Sat, Feb 12, 2011 at 04:23:00PM +, 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) -- _ -- 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
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@your carrier Context: your queue context . ask if you need more info best On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada 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 > -- _ -- 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
[asterisk-users] Using files .call or AMI
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
