Re: [Asterisk-Users] Call simulators
Hi Rob, you could build a simple Perl or Python script to create incoming calls using callfiles. We have used such a strategy and it seems to be working. l. On Thu, 08 Dec 2005 14:15:50 +0100, Rob Hillis <[EMAIL PROTECTED]> wrote: I'm currently starting development of an add-on to a program designed to be used in a call-centre type environment that will interface very closely with Asterisk - quite possibly to the point that the add-on itself will be a softphone as well. In order to test this application properly, I find myself needing to generate a constant volume of calls to a queue. I can do this by dialling from the two test extensions I have set up on my system, but it would seem a better way of doing this would be to have an external application randomly generate calls at a certain volume. My budget is not big - this is a project for a non-profit volunteer organisation I do a lot of work with so I would obviously prefer something open source. The ability to randomly generate caller ID and intermittently suppress caller ID would be a *very* useful addition. Does anyone know of any software that would fit this bill? If such software doesn't exist, or is beyond my capacity to afford, what other options might I have? My test rig is my home PABX - a very small setup running with three ATAs and two VoIP trunks. It would seem that simulating a trunk would be the best way of doing this, but again, I don't know what is available. Any help would be gratefully received. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call simulators
Use asterisk itself to build a box which generates the calls. Maybe what some people misses (call simulators are quite a recurrent query on the list) is that you can move a text file with the equivalent of a manager API action "Originate" in the spool/asterisk/outgoing/ directory and the call will be placed, so it's quite simple to do some intensive test. http://www.asteriskguru.com/tutorials/astertest.html seems nice, never used and I read somewhere it wont compile out of the box with 1.2, but you have the source ... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call simulators
I'm currently starting development of an add-on to a program designed to be used in a call-centre type environment that will interface very closely with Asterisk - quite possibly to the point that the add-on itself will be a softphone as well. In order to test this application properly, I find myself needing to generate a constant volume of calls to a queue. I can do this by dialling from the two test extensions I have set up on my system, but it would seem a better way of doing this would be to have an external application randomly generate calls at a certain volume. My budget is not big - this is a project for a non-profit volunteer organisation I do a lot of work with so I would obviously prefer something open source. The ability to randomly generate caller ID and intermittently suppress caller ID would be a *very* useful addition. Does anyone know of any software that would fit this bill? If such software doesn't exist, or is beyond my capacity to afford, what other options might I have? My test rig is my home PABX - a very small setup running with three ATAs and two VoIP trunks. It would seem that simulating a trunk would be the best way of doing this, but again, I don't know what is available. Any help would be gratefully received. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
