You're thinking of the frequency domain, the actual waves travel in the time domain. You're looking at the different frequencies because somewhere in software somebody has used a transformation (probably a Fourier transform or FFT) to break the components down into their respective frequencies.
However, the microphone was moving back and forth as the sound waves hit it so if you were to measure the position of the mic, it would have been in one position at any given time, but all ways moving continuously as each wave hit it. The voltages would be singular for any moment of time, and the ADC in your sound card only records one sample per one instance of time. (for a sound card sampling happens more often than not at 24kHz, 48kHz or 96kHz)
One point can be represented as a frequency, but the only thing we know about it is it has zero frequency or DC. If you want to actually sample a sine wave, you need more than one point, or the sine wave couldn't be re-created, there would not be enough information:
The sampling theorem (often called "Shannons Sampling Theorem") states
that a continuous signal must be discretely sampled at least twice the
frequency of the highest frequency in the signal.
More precisely, a continuous function f(t) is completely defined by
samples every 1/fs (fs is the sample frequency) if the frequency
spectrum F(f) is zero for f > fs/2. fs/2 is called the Nyquist
frequency and places the limit on the minimum sampling frequency when
digitizing a continuous signal.
So to answer your question:
How is a single voltage reading at a single moment in time able to correlate to so many different frequencies occurring at so many different levels happening at that single moment in time?
It can't, the sound processing software is probably using a trick to cut the waveform off before you see it or filling in information to get frequencies at T=0.