Build four phones for you and your friends

1. The big picture

Each friend gets an old corded phone plugged into a small adapter. The adapter plugs into the home router with an Ethernet cable (no Wi-Fi). All four phones connect to one cloud server. The server is the phone book, the switchboard and the audio relay. You dial 101, 102, 103 or 104. Nothing else can be dialed, including 911.

Two ways to do the adapter

2. Safety first

3. Choosing old phones

Most old analog phones work. A thrift store or eBay is a good source.

Checks before you buy

• Is it a plain corded analog phone with a standard modular (RJ11) plug on the line cord? Four-prong plugs need an adapter.
• Is it push-button? If it has a tone/pulse switch, set it to tone.
• Does the seller say it was tested, and do they accept returns? Many vintage listings do not.
• Use one phone per adapter, so the ring has enough power.

The two eBay phones you asked about

Both should be electrically standard, but neither listing confirms it works. Test any phone on a bought adapter first.

4. The server

Which server to buy

Get a basic virtual server (VPS) with a fixed public IP address, running Debian 12. Four phones need very little, and your budget of under $100 a month leaves lots of room.

Pick: DigitalOcean, New York region, Debian 12, the 2 GB plan (about $12 a month). Prices change; check the pricing page (I have not opened it).

Setup, step by step

For four phones, Asterisk is the quickest to configure and is in the Debian repositories. You can switch to Kamailio plus rtpengine later, when you want circle rules or many more phones.

sudo apt update && sudo apt install -y asterisk fail2ban ufw
sudo ufw allow 22/tcp && sudo ufw allow 5060/udp && sudo ufw allow 10000:20000/udp
sudo ufw enable

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=YOUR_SERVER_PUBLIC_IP
external_signaling_address=YOUR_SERVER_PUBLIC_IP

[phone-endpoint](!)
type=endpoint
context=rotary
disallow=all
allow=ulaw
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

[phone-auth](!)
type=auth
auth_type=userpass

[phone-aor](!)
type=aor
max_contacts=1
remove_existing=yes
qualify_frequency=30

[101](phone-endpoint)
auth=101
aors=101
[101](phone-auth)
username=101
password=CHANGE_ME_1
[101](phone-aor)

[102](phone-endpoint)
auth=102
aors=102
[102](phone-auth)
username=102
password=CHANGE_ME_2
[102](phone-aor)

[103](phone-endpoint)
auth=103
aors=103
[103](phone-auth)
username=103
password=CHANGE_ME_3
[103](phone-aor)

[104](phone-endpoint)
auth=104
aors=104
[104](phone-auth)
username=104
password=CHANGE_ME_4
[104](phone-aor)

[rotary]
; Calls between the four Rotary phones only
exten => _10[1-4],1,NoOp(Rotary call to ${EXTEN})
 same => n,Dial(PJSIP/${EXTEN},45)
 same => n,Hangup()

; Everything else, including 911, plays a message and hangs up
exten => _X.,1,NoOp(Blocked: ${EXTEN})
 same => n,Playback(ss-noservice)
 same => n,Hangup()

sudo systemctl restart asterisk
sudo asterisk -rx "pjsip show endpoints"

How the server works

The phones sit behind home routers, so they cannot find each other directly. The server does four jobs: phone book (who is where), switchboard (connect the call), audio relay (pass the sound along) and rule keeper (only 101 to 104).

1. Register. When a phone powers up, its adapter sends the server its extension and password. The server records where it can be reached. The adapter repeats a small message about every 30 seconds so the router keeps the connection open.
2. Dial. The adapter plays dial tone itself, reads the keypad tones, and asks the server for the extension.
3. Route. The server checks the rule (101 to 104 only) and forwards the call to the registered address.
4. Ring and answer. The other adapter rings its phone. You hear ringback. When it is answered, the call connects.
5. Audio. Both phones send their audio to the server, which passes it to the other side. This is why calls work through any router.
6. Anything else dialed, including 911, matches the catch-all rule and plays a message.

Security basics: long random passwords, only the three port ranges open, fail2ban running, no anonymous SIP. Later, add SIP over TLS and SRTP.

5. Get four phones working (bought adapters)

This is the fast path, and it is what your friends will use. You need four phones, four Grandstream HT801 adapters, four Ethernet cables and the server from section 4.

6. Build your own adapter

The adapter does three jobs. A phone-line module powers the phone, rings it and senses when you lift the handset. An audio codec turns voice into digital samples and back. An ESP32-S3 runs the firmware: tones, keypad detection, and the phone protocol (SIP and RTP) over Ethernet.

Parts and the module

The phone-line module: Silvertel Ag1171

A single-supply (3.3 V to 5 V) phone-line interface in a 14-pin package, with a built-in DC-DC converter and ringing generator, 600 ohm impedance. It gives the phone a constant 26 mA at 3.3 V (30 mA at 5 V), rings it with up to about 65 Vrms (no load), and detects the hook with a logic output.

Module facts from the datasheet

Parts for one bench prototype

An AliExpress listing for the Ag1171 at $1 to $3 is far below DigiKey's price of about $8.50. It may be a different or fake part. This is the part that sends ring voltage to a phone, so buy genuine units for the first prototype (two, so you have a spare). You also need a multimeter.

The protection circuit (datasheet Figure 4)

Build the circuit from the datasheet's Figure 4 between the phone jack and pins 1 and 2. Do not improvise it. The parts it lists:

For in-home use the datasheet says only ESD protection is needed (not lightning or power-cross protection), and it strongly discourages varistors and gas discharge tubes.

Wiring

The GPIO numbers are my suggestion; check them against the pin map of your ESP32-S3 board and avoid pins used for flash, PSRAM and boot straps.

Ten build steps

Do these in order. Do not move on until each one works.

Firmware plan

Build on ESP-IDF with ESP-ADF audio helpers. Run these tasks: hook (reads SHK, debounces, sends events), ring (drives RM and F/R, handles ring trip), audio (20 ms frames to and from the codec, mixes in tones), DTMF (digit detection while no call is up), network (SIP and RTP) and a state machine tying them together.

For SIP and RTP, from most work to least: check whether ESP-ADF's SIP component covers REGISTER with digest auth, INVITE, CANCEL, BYE and G.711; write a minimal SIP client yourself (about eight message types); or first prove the whole design on a Linux board with a mature library such as baresip or PJSIP, then port it.

Echo control. Because the module leaks 12 to 21 dB of the earpiece audio back into the microphone path, plan an echo canceller that uses the audio you are playing as its reference. ESP-ADF and the Espressif speech libraries include an acoustic echo canceller; check whether it suits a short electrical echo like this one, or write a simple suppressor that lowers the transmit level while the far side is speaking.

Call states

#define PIN_RM  GPIO_NUM_4
#define PIN_FR  GPIO_NUM_5
#define PIN_SHK GPIO_NUM_6

// Returns true if the handset was lifted during the ring.
// Set CONFIG_FREERTOS_HZ=1000 so a 25 ms delay is accurate.
bool ring_once_with_trip(void) {
    gpio_set_level(PIN_RM, 1);                          // ringing mode on
    for (int i = 0; i < 20; i++) {                      // 20 cycles at 20 Hz = 1 s
        gpio_set_level(PIN_FR, 0); vTaskDelay(pdMS_TO_TICKS(25));
        gpio_set_level(PIN_FR, 1); vTaskDelay(pdMS_TO_TICKS(25));
        if (shk_high_for_ms(10)) {                      // debounced off-hook
            gpio_set_level(PIN_FR, 1);                  // forward polarity
            vTaskDelay(pdMS_TO_TICKS(20));              // more than 10 ms, less than 500 ms
            gpio_set_level(PIN_RM, 0);                  // end ringing
            return true;
        }
    }
    gpio_set_level(PIN_RM, 0);
    vTaskDelay(pdMS_TO_TICKS(3000));                    // 3 s silent (RM duty 25%)
    return false;
}

#include <math.h>
#include <stdint.h>

// Power of one frequency f in a block of N samples at sample rate fs
float goertzel_power(const int16_t *x, int N, float f, float fs) {
    float c = 2.0f * cosf(2.0f * (float)M_PI * f / fs);
    float s0, s1 = 0.0f, s2 = 0.0f;
    for (int i = 0; i < N; i++) {
        s0 = (float)x[i] + c * s1 - s2;
        s2 = s1;
        s1 = s0;
    }
    return s1 * s1 + s2 * s2 - c * s1 * s2;
}

// Use N = 205 samples at 8000 Hz. Check these seven frequencies:
// low group:  697, 770, 852, 941      high group: 1209, 1336, 1477

7. Test plan

Do all of this with the four phones in two or three locations, before handing them out.

• All four show as registered on the server
• Every phone can call every other phone (12 combinations)
• Ring is loud enough, and stops when the caller hangs up
• Both sides hear each other clearly, with no one-way audio and no echo
• Dialing 911 or any other number plays the blocked message
• Unplug and replug an adapter: it re-registers within a minute
• Restart the server: phones re-register by themselves
• A 30-minute call stays clear
• Two calls at the same time work

8. Trial with friends (2 weeks)

This is only for you and your friends: no investment, no customers, no pre-sale. Give a phone to each of three friends and keep one. Ask them to just use them.

Track in a simple table: date, who called whom, length, and planned or spontaneous. Check the server log with:

sudo asterisk -rx "cdr show status"

Ask each friend after two weeks

1. How many times did you call, and who started most calls?
2. What did you do differently with this phone than with your mobile?
3. What annoyed you? (setup, ring volume, sound quality, cord, size)
4. What would make you use it more?

9. Purchase list

Prices come from listings and search results seen in November 2026 and will change. Check each page, stock and shipping before ordering.

For the 4-phone trial (bought adapters)

For the DIY adapter (one bench prototype)

Suggested order

1. Message the two eBay sellers with the questions in section 3, and buy the phones that pass.
2. Buy the four HT801 adapters, cables and cords, and create the server.
3. Order two Ag1171 modules and the ESP32-S3 module while the trial runs.
4. Order the codec, protection parts and enclosure. Check the datasheet's later electrical characteristics before choosing level-matching resistors.

10. Common problems

11. What comes next (optional)

Nothing here needs money or customers. It is only for you and your friends. If you love it:

• Your own adapter: finish section 6 and swap it in for one friend's phone.
• Server v2: Kamailio as the registrar and proxy and rtpengine for audio relay, with rules for who may ring whom instead of fixed extensions.
• Share it: publish the server configs, firmware and board files so others can copy your setup.
• Later, only if you want to: decide together whether to build more phones for other people.