Asterisk basics and language, what to choose?

Question

Asterisk

Do not send to read "Telephony of the Future".
I'm actually training and I want to understand how to deploy telephony from scratch. I don't consider FreePBX. I want a "naked" asterisk. In all manuals on the network there are different programming languages ​​​​in it. I want to understand how "correct" and "for the future" so that in a year it will not become obsolete and it will not be turned off (I'm talking about the language of functions).
There are three options on the web:

• default SIP language (don't know what it is);
  
• AEL language
  
• and dialplan directly in the database (such as real-time changes)
  

So the question is how to properly deploy telephony, in what language is it better and correct to write extensions, and most importantly, it’s more convenient, because as I understand it, some support certain functions, and some don’t.

Answer 1

AEL is somewhat more convenient in writing dialplans, but when loaded, it is translated into a standard config, so if necessary, you still have to debug the dialplan in the standard. Plus, during translation, additional NoOp commands are added to transfer control structures if/else, case, while, etc.
The dialplan in the database is just a storage option. In fact, it is no different from the standard config, just one line of the dialplan - one line of the table in the database.

Answer 2

I also studied on a bare asterisk, and although I still use FreePBX at work, I miss that bare asterisk. First I coded on standard exten. But then I transferred all the logic to AEL, since it is much more flexible, but not at all more complicated. So I advise you to first make the simplest configuration in the standard language out of the box, then connect the AEL module, and try to rewrite the configuration in AEL, and you will notice how much more convenient it is.
If you know how to use LUA, then it seems like you can still use it, but I don’t know how to use it.

Answer 3

In general, the approach is correct, which is directly from the bare Aster. Also, start right away, of course, without the real-time config and database, but just the sip.conf and extensions.conf files (well, maybe also, asterisk.conf).
But also, note that during the study you will come across a recommendation to use the new set - pjsip (pjsip.conf file) instead of the outdated sip. Yes, it is cooler than the vulture, and yes, it is worth switching to it. But it can scare away with its volume of settings. What was done in sip in 3 lines, in pjsip is done in 5 blocks of 3 or more lines. That is, until pjsip is pushed in, when you get comfortable in practice with sip, then switch to pjsip, and possibly to realtime.