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.