Why can't I make a 2-way call with the Cisco spa303 (3-way call)?

Question

Asterisk

Colleagues!
Faced with a rake: there are two completely identical cisco SPA303 devices (well, except for MAC, they are no different). They connect to FreePBX 12.0.1beta11. Firmware version on the devices: 7.5.2 (as I understand it - the last one). 1 external line is configured on FreePBX, all calls to the outside are output through it. SLTs connect to the aster normally, calls can be made (both to the city and within the network)
The essence of the problem: when you try to make a call to two devices at once (collect the conf using the phone itself), on one device, let it be internal 111 (the first device), when you press the conf button again (i.e. the whole process looks like this: 111 make a call to 112, pick up the phone to 112, click on 111 conf, dial 115, pick up 115, press conf again) at the top of the device’s screen, the display name blinks for 112 and after a couple of seconds the call falls off, while 111 and 115 connect and can communicate.
The most interesting thing is that if you collect a conference from number 112 (the second device), the conference is assembled without problems, i.e. when you press conf again, a quick double beep sounds and you can communicate in three handsets.
The most hemorrhagic is that periodically, it happens, with 111 it is also possible to collect three people. With 112, it turns out to do this every time.
Can you please tell me what could be the reason for this strange behavior? At least tell me which way to dig, because. already completely lost in conjectures. Unfortunately, I do not have a complete understanding of the work of the asterisk (and even more so FreePBX), I will be glad for any hint! Thanks in advance.
PS. Tried to do mana from here(second comment from the top), after which they began to gather. But it still confuses me very much that, provided that the internal mechanism of the phones themselves is used, the master should, in theory, process each call from this device separately .... (i.e. when creating a conference from 111, a call is first made to 112, after which it is put on hold, a call is made to 115, after which the bridge mechanism of the phone itself is turned on and combines these two calls).

Answer 1

According to the manual, the link to which you provided, you just set up conferences on the PBX, and not on the device. Moreover, using the MeetMe application, which has long been considered obsolete and is recommended for replacement with ConfBridge.
De facto, all implementations of conferences in Asterisk at the moment are either rather difficult to implement or do not work very intuitively (and it can also be unstable at the same time).
There is experience in implementing conferences on the actual "pure" Asterisk (without FreePBX), if this is an urgently needed functionality, we can discuss cooperation.
PS By the way, the latest firmware version for this line is Cisco (Linksys) 7.5.5, but first you need to upgrade to 7.5.2b, because there is a bug in 7.5.2 that prevents you from updating dr 7.5.5, and 7.5.2b as once it fixes it.