New to Asterisk and FreePBX

so far, i have installed AsteriskNOW on an experimental server, and i have a 8-port analog (FXO) card, that is recognized in Connectivity -> DAHDI Configuration -> Analog Tab.
I have ordered a used Polycom SoundPoint IP 650 SIP phone for testing my system (not yet delivered).
my intent is to learn all i can about VoIP, and how to make my system work. i think i got the correct analog card to connect to a analog POTS line. i think what i am building is called a analog to VoIP gateway.
this is not for a company phone system, just part of my network admin hobby, to learn for the future. I am not completely fluent in the terminology around phone networks, but i will do my best to learn.
is there a sort of guide for setting my system up? so far, i have found for setting up a SIP trunk based system. i think SIP is a over-the-internet connection, but i just want to connect to an analog POTS line (or more than one). just trying to keep it self contained while i learn.


Iā€™d say install the official distro, which you can download at

Everything is very well documented on the wiki, from installation to troubleshooting

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i upgraded the system via the web gui, and it seems to be doing just that. it is into the post-upgrade portion, and it booted as SNG7.


I am downloading the official distro, just in case.

it seems i have issues with my install/upgrade.

exit: 255
Unable to continue. Call to a member function add_error() on null in /var/www/html/admin/modules/dahdiconfig/modules/sangoma.module on line 94
#0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, 'Call to a membe...', '/var/www/html/a...', 94)
#1 [internal function]: Whoops\Run->handleShutdown()
#2 {main}

I get this error whenever i hit the button for Apply Config on the web gui. it seems to occur regardless of what setting was changed.


upgrade to edge:

fwconsole ma upgrade dahdiconfig --edge
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