Now i know there are many written articles about this and many different ways to do it from easy to complex way but i find my method quite easy to implement and involves no manually editing any file. The v's increase the level of verbosity, four v's are probably enough but I tend to go hog wild).One of the most required features of any PBX in any company is call restrictions. (If you are not familiar with the Asterisk CLI, from a Linux command prompt enter something like It reports online to figure out what it is really telling you. Also watch the Asterisk CLI as you register a client and as you try to make a call often it will tell you what the problem is, though sometimes in a rather cryptic way and you will need to search what If you want to try to figure this out, look in the SIP settings module and make sure you haven't accidentally blocked anything. But for what you are doing, you may find it easier to try something a little simpler than FreePBX, such as Wazo ( ) or Ombutel ( ) (yes I realize this is a FreePBX sub, but if someone doesn't want anything but basic intercom service there is no need for them to get into the complexities of FreePBX). The only reason you would need to pay $$$ is to have someone set up a working configuration for you, if you can't figure it out on your own. Basic FreePBX is perfectly capable of handling internal calls. The "First you need to do some configuration with this module you need to pay $XXX for" is bullshit. Asterisk has had a long-standing bug where if a trunk can't connect, it can take down internal sip connections. Make sure you have not inadvertently set up a firewall that blocks connections from addresses on your local network.Īlso, make sure no trunks are enabled, if you created one for some reason. The FreePBX guides all seem to start with "First you need to do some configuration with this module you need to pay $XXX for" which is generally unhelpful for users attempting to evaluate the software, or those operating non-commercially.Īre there any good places to start with FreePBX? I've scoured the web for help and guides but I've not found anything helpful. I've set up a PJSIP and a SIP extension with associated user accounts but neither worked.
They all connect and I can see their attempts at registering in the Asterisk Logfiles but I don't know why their registrations are being returned as unauthorised. I cannot get any SIP client (I've tried about 5 different ones) to authenticate with FreePBX. Before I even connect handsets though, I'm trying to connect a single SIP client from an Android phone to the FreePBX server, installed fresh on a VM bridged to my LAN.
They have no need to contact external numbers.
I'm trying to set up FreePBX to route calls between a few handsets on a LAN.
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