Test my SiP Trunk performance

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Test my SiP Trunk performance

Postby wilwil » Tue Jul 21, 2015 2:07 am

Hi everyone,

I'm a newbie to Asterisk, but I really like this amazing soft, so I want to exploit it as possible as I can to discover its advanced functionalities.

So, I'm using Asterisk 13.4 installed on 14.04 Ubuntu server. And I recently installed Freepbx 12 (by obligation because I must give a graphical solution in the end to a final user who don't know anything about this stuff).

So for me I want to know about :
-- How to choose my SIP Trunk Provider ?
-- Depending on what criteria ?
-- The most important is if Choose one, how can I test it using my "Asterisk server" (its Performance / Robustness /efficiency ... )?
Is there a special scenarios can I run it to measure the performance of my trunk?
Or might be Asterisk has applications to test Sound Quality, Signal Attenuation or Quality of Service or anything I can defend with it my choices in choosing a Sip Provider ??

One more thing please, Or if there is Providers Well Known and tested by "Digium" ? At least because the solution is intended to be used to different places in the world?

Please can you help me.
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