[asterisk-users] SiP call generator

[Atis Lezdins]

On 2/20/08, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
> On Wed, Feb 20, 2008 at 01:35:20PM -0500, Matthew Rubenstein wrote:
>  >       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?
>
>
> Most of the things here are probably not that difficult to script within
>  Asterisk itself, or with a simple wrapper.
>
>  Test of audio quality is something I'm not really sure how to do.

Run tests, and ChanSpy() them? See at which point decrease of quality
becomes hearable.

Regards,
Atis

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