Run VitalPBX-Stress-Test on Asterisk 1.4.30


I am trying to make some stress tests between VitalPBX and an older Asterisk 1.4.30 for my final work school but even create the folder “vitalpbx”: to insert these two files


and make reference to these files in the files:

  • extensions.conf


  • sip.conf

the value of Asterisk calls is always 0.

this is my values on my script:

scritp edited:

I had remove protocol IAX and codec GSM nad add codec ulaw and alaw

Could you please provide some feedback to make some changes.

The script is not intended for old versions of Asterisk.

I adapted the script to run with Asterisk 1.4.30.

@rodrigocuadra Can I share the file in your GuitHub like a fork or a branch for people that want to run this script with lower versions of Asterisk?

I added the option of codec G. 711 a-law (a-law) and the μ-law (u-law).

1 Like

@miguel Is it possible to provide me the contact of Rodrigo Cuadra or give me the reason for you said that these values in script:

Calls Step (Recommended 5-20)… >
Seconds between each step (Recommended 5-30)… >
Reference: GitHub - VitalPBX/VitalPBX-Stress-Test: VitalPBX Stress Test

I choose your script for my final academic school project to create stress tests in Asterisk PBX 1.4.30 and compare them with VitalPBX 3.2.3-2, but I need the explanation of why you recommend these values that I put in bold. What is the main reason?

Thanks for or attention.

If you give me permission, I would like to share my version in your GhitHub

Answering your questions:
1.- Calls Step (Recommended 5-20)… >
A/ It is to see the behavior of the CPU as more calls are generated.
2.- Seconds between each step (Recommended 5-30)… >
A/ Many times when many calls are generated at the same time, there is a saturation peak in the CPU, this time helps to normalize this saturation peak and thus the behavior of the test is achieved to be more natural.


Thank you very much @admin for your replies.

This will help me with the all justifications for these timers in my final school report.

By the way. Can I share with you my script and if you approve, attach it or add side your script version in GitHub for others to try older Asterisk versions?

Is this possible to make tests with this script and configure encrypted calls, SRTP?

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