Run VitalPBX-Stress-Test on Asterisk 1.4.30

VitalPBX
1

Hi

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

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and make reference to these files in the files:

  • extensions.conf

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  • sip.conf

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the value of Asterisk calls is always 0.

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this is my values on my script:

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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.

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The script is not intended for old versions of Asterisk.

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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
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@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

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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.

2 Likes
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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?

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

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