How to organize IP telephony with connection to an analog line?

Question

Telephony

I am writing a term paper on the topic "Designing SCS IP telephony of one of the university buildings". I have never worked with IP telephony, I know only a little theory. A description of the equipment settings in this coursework is not necessary, so just a logical diagram is needed.
I plan to do this:
- 12 phones are connected to the switch (switch and phones with POE);
- the switch connects to the server on which Asterisk is installed;
- the server connects to the router;
- the router is connected to the university's IP telephony network.
Will this work and how can I connect this network to an analog line so that I can send and receive calls to simple phones?

Answer 1

It is not entirely clear what is meant by the switch. If the phones are analog, the circuit will be like this:

[ТА]<---->(FXO-порт)[VoIP-шлюз](FXS-порт)<----->[ТФОП]
                          ^
                          |
                          v
                     [Asterisk]<----->[Маршрутизатор]<---->{Внутренняя сеть}

If the phones are capable of VoIP, it makes sense to register them directly on Asterisk, bypassing the FXO port. In general, if there is no task of very cunning routing from the internal network, Asterisk is not needed in this scheme: the gateway itself will quite cope after resp. settings.

Answer 2

There are FXO-gateways that drive the analog from the PBX to IP pretending to be a regular phone. But the dtuka is quite expensive and when a certain number of outgoing lines is increased, it turns out to be more profitable to take a stream from a SIP or E1 operator. And not necessarily on this site, I knew an organization where in the regional branch they took 100 numbers via E1 and already distributed them to branches via their VPN.