Minimalistic Asterisk configuration howto


#1

Hello,
I have installed samples but felt a bit overwhelmed by its amount so I moved sample config files to /etc/asterisk/mySamples directory.
Now I try to build Asterisk system using as simple configuration as possible and having only needed modules in /usr/lib/asterisk/modules.

What modules and config files are required to enable one SIP phone (from the same network) to call second SIP phone (from the same network)?

Thank you for the answers.


#2

I don’t have the answer to your specific question, but how “minimalistic” do you need the system to be? Is this due to system resources available like space, RAM, etc? Or, are you just looking for simple management, overall?


#3

Simple management and full control over all things.
It does not have anything to resources.
I want to start from even insecure very basic settings with only two SIP and only needed modules loaded.

The case is that I installed samples and was overwhelmed by its amout.
Instead of understanding EVERY config I want to simply omit of those not needed at the beginning.

I tried to remove nearly all configs but:
asterisk.conf
modules.conf -> here I switched off autoload and want to specify only required modules
sip.conf
extensions.conf
but then I met: viewtopic.php?f=1&t=76115 issue


#4

That makes perfect sense, yet i still don’t have the answer. I could suggest turning everything back on, get it working, then disabling one-by-one.

Also, I have found FreePBX to be pretty simple to use. That is where I started with Asterisk and it’s helped me learn quite a bit about Asterisk itself.


#5

Selecting a minimum set of modules increases the up front management cost and doesn’t really save you anything subsequently. Nearly all the security risk in a SIP system derives from misconfiguration of chan_sip.c and that is the core module for SIP.


#6

ok what I meant was:
modules.conf

[modules]
autoload=no
load => pbx_config.so
load => chan_sip.so
load => codec_ulaw.so
load => codec_alaw.so
load => codec_gsm.so
load => app_dial.so
load => res_rtp_asterisk.so

It gives the possibility to dial one SIP client to second SIP client and communicate with one codec from alaw, ulaw and gsm codec set.