[asterisk-users] SiP call generator

[Atis Lezdins]

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.

Regards,
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.
> >
> > 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
>
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-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835