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My Homemade PBX


I've been fascinated with telecommunications from an early age. When I was twelve or so
I had an intercom system from which I could talk to most of the house from my corner of
the kids' room. It was made out of the amplifier from a record player, and miscellaneous
parts from old TV sets. When I was fourteen someone gave me a WW2 vintage
shortwave receiver with a blown power supply. I was able to make a new power supply
for it, out of TV set parts, and then I spent many hours scanning the airwaves for radio
signals in my native tongue. I was always making intercom systems, lamp signaling
apparati, even a homemade morse clicker although I never learned the code. What I
wanted to build most of all was a dial telephone system but I had no phones and no idea
how the switching apparatus could be designed.

My decision to go for a co-op and then fulltime job at Bell-Northern Research was
influenced by the closeness to telephone technology this would entail. What did I know; I
ended up designing computer hardware which, while it does run telephone central offices,
has no visible relation to telephones.

I did gain access to a lot of junked electronic components that would allow me to build
complex projects. Also I had learned about microcontrollers which allow the construction
of smart electronic projects. And lastly, I had begun to frequent suburban garage sales,
where telephones of every description were plentiful and cheap. The stage was set to
finally build my own dial telephone system. I spent several months of evenings and
weekends on this, in the 1992-93 timeframe. I didn't draw a schematic for it, but I will
give here what information I have in my notes or can remember.

This is intended for educational and/or entertainment purposes. In no way is it sufficient

information to duplicate the circuit. Perhaps it will satisfy the next person who asks about
it after reading the brag reference I inserted into this old Usenet posting.


• Eight telephone extensions with roughly telco spec voltages and currents (48V
onhook, 90VRMS 20Hz sinusoidal ringing, about 25mA loop current offhook.)
Ring trip is sub-spec but can handle at least 3 "500" type rotary dial telephone sets
in parallel without false tripping. Lines are not balanced, nor is one side ground.
• One central office line capable of inbound calls (ring detector) and outbound calls
with DTMF and pulse dialing (selectable, independent of the type of extension
phones. That is, the PBX can convert tone-pulse and pulse-tone.)
• Three internal voice buses, meaning up to three calls in progress simultaneously.
• Telco standard call progress tones (dial, busy, fast busy, audible ringing.)

The following features are available on the extension lines.
Flash Put current call on temporary hold, await command
00 Ring all stations
Set ring ring pattern:
0 --- ---
1 -- -- --
2 - ----
3 ---- -
01x 4 ----
9 ------
Ring again -- redial last extension, if busy ring me back when it becomes
Speed dial x (0-9)
Park/unpark call against extension x (put on hold so you can hang up /
retrieve from hold.)
05 Park/unpark call against own extension
06x Forward calls to extension x, or to outside line with 069...
060 Cancel call forward
081x... Set speed dial x (0-9) with digits that follow
082 Disable/enable ringing for outside calls
09 Join incoming outside call in progress
9 Get outside line. If outside line already on hold, send a flash on it
# Redial

Example Usage

Call arrives on outside line. All extensions that currently have outside ringing enabled
(toggled with 082) ring with the same cadence as the outside line.

Extension 5 picks up the call. All phones stop ringing.

Extension 2 picks up too, gets dial tone. Wishes to join the incoming call, so dials 09.
Connected to extension 5 and the outside line.
Extension 1 dials 9. Gets busy signal. Dials 02, gets stutter dialtone confirmation, hangs

Extensions 2 and 5 both hang up, terminating the outside call. Extension 1 gets a single
short ring. Picks up the phone, is automatically connected to the outside line.

Extension 1 makes a pulse-dial call but the outside line is configured for DTMF so the
PBX converts the digits.

Extension 1 parks the call by flashing, then dialing 05.

Extension 7 forwards all calls to an outside number by dialing 0697451576. Extension 3

retrieves the parked outside call by picking up and dialing 041.

Extension 8 dials 01800 to ring all available stations with the four-short-bursts ringing
pattern. Prearranged signal for a certain person to pick up, whatever extension they are
nearest. Extension 5 picks up, call is completed.

Extension 3 wishes to access a special feature on the outside line, transmits a flash by
flashing to enter command mode (confirmed with stutter dial tone), dialing 9.

And so on...

Hardware Design Philosophy

This is not a low-cost design intended for publication and exact reproduction by others. It
was completely tailored to what components I already had and what was cheap to buy. It
is also not terribly efficient; if in doubt I insert an extra op-amp buffering stage or more
clamping diodes just to be sure. The low-tech relay switching matrix is because I had tons
of relays but no CMOS switches, and didn't know how to use the latter at the time

The massively oversampled digital tone generator is because I am a digital weenie.

Similarly offhook detection in the line circuits could probably be done with solid-state
circuitry but I actually like the chatter of the relays as phones are dialed.

Analog Hardware

Let's start with the line circuit. There is one of these for each extension.
The voice paths 1-3 of all the line circuits are connected together. So if any given pair of
line circuits select the same voice paths with relays K3 and K4, they can talk with each
other. Any line that is not currently connected to a voice path is terminated via a 600 ohm
resistor so that it doesn't sound funny.

When the phone goes offhook, current is drawn through the 200 and 300 ohm resistors
and the line relay K2, which closes. Via the debounce circuit a clean OFFHOOK- signal
goes to the control complex. Pulse dialing and switchhook flash are detected by timing
the OFFHOOK- signal in software.

The large voltage changes caused by this are kept out of the voice circuitry by capacitor
C2, which at 4uF is much larger than it needs to be.

To ring the line, ring voltage is generated and relay K1 is activated. This places capacitor
C1 effectively in parallel with the relay coil, shunting the potentially strong ring current
around it so that it doesn't chatter (which would result in "false trip", i.e. a false offhook
indication.) However if the phone goes offhook, the capacitor is discharged by the DC
current and the line relay closes within one or two cycles of the ring waveform. The
control complex then immediately cuts off ringing by opening relay K1.

That's all there is to it, except for very careful sequencing of the relays. For example, the
control signal to K1 is synchronized in hardware to the zero crossings of the ring
waveform, and the voice path is connected via K3/K4 only when C2 is in steady state, so
that no click results. For example, if the line has just picked up after being rung, a brief
delay occurs before the call is connected.
Connected to each voice path is the circuitry to detect DTMF tones and generate audible
call progress tones. I don't have a schematic for it, but I do have a wiring diagram which
is just as good:
The top device is a Crystal Semiconductor 8870 DTMF receiver. It listens to the voice
bus via a super conservative arrangement of resistors, capacitors and diodes to ensure it
can't get damaged by voltage transients. Due to its high input impedance it can always

The bottom device is a DAC-08 digital-to-analog converter for generating the tones. A
dual op amp (LM358) is used to convert the current output of the DAC-08 to voltage,
then rebuffer it after lowpass filtering. The output impedance is 600 ohms. Again, diode
clamping is used to protect the electronics from harm. This circuit is only connected to
the voice bus when needed via the voice bus relay (not shown) as the 600 ohm
termination impedance is not wanted during an actual call.

Next, here is the circuit that generates the ring waveform:

Not shown is the DAC-08 that generates the 20Hz sinewave; it's hooked up the same way
as in the voice path circuit. An extra op amp is used as an adjustable gain stage to drive
the output transistors. Not shown is the push-pull darlington pair of output transistors
running on +/- 20V and the transformer that transforms the ring voltage up to the final
level. There is no provision to eliminate crossover distortion; a "nearly sinusoidal" ring
waveform is quite good enough. The transformer needs to work at 20Hz; normal AC line
transformers don't, so I had to salvage an audio output transformer out of a junked hi-fi
tube amplifier. The ring voltage is gated on and off by controlling the bits going into the
DAC; when no phone needs to be rung it is kept off for safety.

Not shown is the power supply which generates the following voltages:

• +/- 20V unregulated for the ring driver

• +/- 15V regulated for the analog electronics
• +10V unregulated for the relay drive (12V relays but close enough)
• +5V regulated for the digital electronics
• -48V unregulated but very well filtered for the line circuits

Not shown is the circuit for the outside line. In a nutshell it has a relay tree like K3/K4 in
the line circuits to be able to connect to any of the voice paths. Instead of the 600 ohm
resistor in the default tree brach, it has a DTMF dialer block consisting of a "5089"
DTMF generator, a 74LS139 to control it, and an op-amp driver stage to buffer the
output. More diode clamping, and coupling to the world via an air-gap type 1:1 audio
transformer (without the air gap it would saturate due to the DC current flowing on the
outside line side.) A ring detector similar to that found in a modem detects outside line
ringing, and a simple relay connects and disconnects the transformer to facilitate going
offhook, pulse dialing and flashing.

The reason I don't show these things is not that they are more obvious than the other stuff,
but that I don't have neat diagrams I can scan in.

Digital Tone Generation

The DACs take unsigned 8-bit linear audio samples. The following waveforms need to be

1. Silence.
2. 20Hz sinusoid for the ring voltage.
3. Sum of 350Hz and 440Hz sinusoid for dial tone.
4. Sum of 440Hz and 480Hz sinusoid for audible ringing.
5. Sum of 480Hz and 620Hz sinusoid for busy signal.

The samples all stored in an EPROM. Because of the way the clock frequencies worked
out and because I did not wish to bother with much analog filtering, I decided to generate
tones at 86.4K samples/second, and since the tone waveforms all repeat themselves
within 1/10 second I generated 1/10 second's worth. This required a 27512 type EPROM,
which by that time was a cheap part so the waste of resources was insignificant. The
following AmigaBASIC program was used to generate the samples (in assembler input
file format.)
ns = 8640
pi = 3.14159265#
OPEN "tones.asm" FOR OUTPUT AS #1
PRINT #1,"; Digitized ring, dial, audible ring, and busy tones"
PRINT #1," .org 0"
FOR i = 0 TO ns+3
PRINT #1, " .byte ";
PRINT #1, INT(SIN(2*i*2*pi/ns)*127.99+128);
t1 = SIN(35*i*2*pi/ns)
t2 = SIN(44*i*2*pi/ns)
t3 = SIN(48*i*2*pi/ns)
t4 = SIN(62*i*2*pi/ns)
PRINT #1, ","; INT(t1*63.99+t2*63.99+128);
PRINT #1, ","; INT(t2*63.99+t3*63.99+128);
PRINT #1, ","; INT(t3*63.99+t4*63.99+128)
The microprocessor controls all this with just seven control signals -- three 2-bit tone
selects for the voice paths and a 1-bit ring voltage enable. Digital sequencing
automatically gates the tones on and off at the 1/10 second points where they cross zero,
so pulsed tones don't have clicks where they are gated on and off. Likewise the ring
voltage is gated at zero crossing points, plus a synchronization signal is generated
somewhat ahead of the zero crossing point to allow the ring relays to be switched close to
it. All this makes the system "sound" cleaner and also reduces wear & tear on the ring
relay contacts.

The ABEL files for the two PALs in the digital tone section can be found here and here.

Control Complex I/O Summary

Quite a few control signals have accumulated in the previous description. These are
mapped into the 8031 CPU's address space as a bunch of read-only input locations and
write-only output locations.
Read Write
4000 H8 H7 H6 H5 H4 H3 H2H1 . . TN1 TN0 . D2 D1 D0
4200 M8 M7 M6 M5 M4 M3 M2 M1
4400 L8 L7 L6 L5 L4 L3 L2 L1
4600 R8 R7 R6 R5 R4 R3 R2 R1
4800 . RE _TS2__ _TS1__ _TS0__
6000 SW V2 V1 V0 __DTMF rec 0__ . . . . _DTMF sender__
6400 __DTMF rec 1__
6800 __DTMF rec 2__

Where, for line x, Hx is the OFFHOOK input, Mx and Lx are the controls for relays K3
and K4 respectively, and Rx is the control for K1 (ring relay.) For voice channel y, Dy is
controls the relay that connects the tone generator to the channel, "DTMF rec y" is the 4-
bit DTMF receiver listening to the channel, Vy is DTMF tone valid for the channel,
"TSy" is the call progress tone select for the channel (silence, dial, audible ring, busy.) RE
is the ring voltage enable, "DTMF sender" is the control bits for the DTMF generator.
TN1 and TN0 control the relays that connect the outside ("trunk") line to the
one of the voice paths. SW is the DIP switch that selects pulse or tone dial
for the central office line.

Not shown is the offhook control for the outside line, which is connected
directly to P1.1 of the 8031, and the ring detect from the outside line, which
is connected to P1.0.

The TSy and RE outputs are retimed in by the tone sequencing circuitry.

Control Complex Hardware

The control complex is an 8031 microcontroller operating at 11.0592MHz. A 2764

EPROM occupies locations 0-1FFF hex of its address space, and an 8Kx8 SRAM
occupies 2000-3FFF. The serial port is brought out to a standard RS232 connector vial a
MAX232 device. The addressable I/O locations described earlier are implemented as a
bunch of address decoding and 74x374 octal flip-flops/tristate drivers. There is no
schematic, but there is one PAL file.

ULN2003 drivers are used for the relays.

Control Complex Firmware

The firmware is fairly well documented in the source code. The assembler used was as31,
a free 8031 assembler from the net.

The firmware was debugged by keeping a simple debug monitor and download program
in ROM, and downloading the firmware to RAM for testing. No other debug tools were
used. When the firmware was finished, it was burned into ROM. The serial port is not
used with the final firmware.

Last words

It works! From a telephone, it sounds indistinguishable from the real telephone network.
When a call to another extension connects through, there is no click, you merely hear the
dying oscillations of the bell as the handset is lifted away. Phones and answering
machines of every description have been tried with this and all work fine.

I wired a lot of LEDs into the circuit to show the status of all relays, the tone selects, the
ring voltage, the external ring detect. By picking up three phones and dialing different
extensions with different ring cadences, quite a neat rhythm of clicks and flashing lights
can be generated.

Despite the unbalanced lines, there don't seem to be hum pickup problems. I have had a
telephone separated from the switch by about 100m of twisted-pair telephone wire and it
worked fine.
The only value this thing has is entertainment value. It's great when there are guests with
bored kids, and once set up it gets plenty of attention from the adults too, if they are
engineers that is.

I don't actually use it for anything.

It was worth building because it finally scratched that childhood itch.