Well, PHP is language in which i'm coding most for last 5 years, so when i needed something fast, i took it. And maybe some day it will have web interface.
Regards, Atis On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote: > Just out of curiosity, why PHP? > > Atis Lezdins wrote: > > On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> 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. > > > > Regards, > > Atis > > > >> Atis Lezdins wrote: > >>> On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> 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. > >>> > >>> Regards, > >>> Atis > >>> > >> > >> -- > >> Alex Balashov > >> Evariste Systems > >> Web : http://www.evaristesys.com/ > >> Tel : (+1) (678) 954-0670 > >> Direct : (+1) (678) 954-0671 > >> Mobile : (+1) (706) 338-8599 > >> > >> _______________________________________________ > >> -- 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 > >> > > > > > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- 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 > -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 _______________________________________________ -- 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
