Is there a simple tool that I can use to script Asterisk generating lots of calls according to a peak traffic curve, with random variance within a specified percentage around that curve, to test a number of DIDs at which I terminate voice recordings to test the audio and call quality? Any that will also give me a report of the actual traffic connections?
On Tue Feb 19 09:00:45 CST 2008 Atis Lezdins wrote: > On 2/19/08, Alex Balashov <abalashov at evaristesys.com> wrote: >> Or, you can write your own scripts to generate calls via the Manager >> API, or use Asterisk call files (see voip-info.org on this topic). >> >> But, all other things being equal, it is probably preferred to use some >> sort of testing framework of the sort mentioned below. > > The PBX Testing Framework i mentioned (and also developed) provides > call-generation trough call-files so all you have to do is code action > scripts (answer, talk for 3-10 minutes, transfer to other extension, > etc..) and call generation scripts (random agent call every 10-20 > seconds, and random customer call every 20-30 seconds), all in PHP > with some functions and objects to make interaction easy. > Atis > >> Atis Lezdins wrote: >>> On 2/18/08, Khaled Chehab <kchehab at xplorium.com> wrote: >>>> >>>> >>>> I want to have a PC-based real-time VoIP bulk call generator (including >>>> both >>>> SIP signaling and RTP generation) >>>> >>>> for stress testing and precise analysis of the VoIP network equipment. >>>> >>>> >>>> >>>> Do any one knows a free program can do that . >>> If you want just simple calls, i suppose SIPP can do that. >>> http://sipp.sourceforge.net/ >>> >>> If you want to have those calls perform some actions (send DTMF, etc), >>> you can try to write your own scripts based on PBX Testing Framework. >>> http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's >>> designed for testing queue-agents scenarios but i'm sure you can >>> adapt. >>> Atis >> Alex Balashov -- Alex Balashov -- (C) Matthew Rubenstein _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
