Mystery with busy tone any time a call is attempted

Have an old version that was working with old Polycom phones. A very simple set up.
This is used only for internal calls. Nothing to the outside world, trunks with ISP.
It is a set up in a classroom for students to grasp some concepts, ending in making a call to each other, only internally.
So what is happening now?
Polycom phones get a DCHP address, then extensions are created in Free Pbx (PJSIP). Students log in to web interace on their respective phones and set up extension number and password that matches the extensions created in the IP PBX with the correct PBX IP and default ports 5060.
After that we check phones one by one and Line status is registered.
Then we attempt to ring one extension (say as example 5090) and as soon as we press call, there is a busy tone straight away.
The IP PBX and the phones are all working in the same subnet at Layer 2, there is no routing to anywhere, very simple as stupid.
I have tried at my home with the latest Free PBX on a virtual machine but with old Cisco phones doing exactly the same and it works.
I am so lost…
Any ideas, any logs that might exist ? Yes, I should attempt wire shark as well.
A total mystery

The main Asterisk log is at
or you can view it in the GUI at Reports → Asterisk Logfiles.
At the Asterisk command prompt, type
pjsip set logger on
to include a SIP trace in (future) logs. If you have trouble interpreting them, paste the relevant section at and post the last eight characters of the URL.

What model and software version? I have some old VVX series and some really old Soundpoint IP series. AFAIK, they don’t ever display ‘call’. When entering digits on-hook, there is a Dial softkey; when entering off-hook a Send softkey is displayed.

Try calling *43 (echo test) from the phones. Enter *43 while on-hook and then press Dial.

If you still have trouble, set up an extension with a simple free softphone, e.g.

Then try calling from Polycom to softphone and vice versa.

You definitely should try to use Wireshark.

One big issue I had with Polycom phones was their multicast / broadcast protection mechanism. At some unspecified level of broadcast traffic they were blocking all broadcast, including incoming ARP request. This resulted in a very erratic behavior and lot of running between network switches. In my opinion this is very dumb feature and I would very much prefer if these phones were just crashing (and they did if this mechanism was turned of). Yealinks and Grandstreams were working fine in the same network.

Thanks, but there are other tests I have done.
I took one phone to my house and used Kerio Operator. The phone was up an running in no time.
I also downloaded Vital PBX and works.
So I will give Free PBX the rubbish bin treatment.
3CX is king, yes I have Yealink phones.
So, in summary I am not impressed with Free PBX and have not got time to waste.
I shall move to another product, after all many are based on Asterisx but depends on men hours of work and testing rather than launching it and good luck to you.
Yes, I hear what you say, I may use Wire Shark to see those conversations when I have time.
Vital PBX has also the same drama of Free PBX, first set up you log in with the Web Interface and then you are blocked out. Many say is a flaw in the Apache module with firewall or blacklist. Also the IPS = Intrusion prevention. I do understand all of that, why the user in the documentation is not warned about this in the first place?
How something is released without checking or testing ?
Yes, fail2ban command and then disable the firewall, also the documentation is as useful as a door bell in a graveyard.
Anyway, thanks.

One swallow should not make or break a summer.

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