3 IVR Audio Converters, Compared Honestly

You record a greeting, upload it to the phone system, and it is rejected. Or worse, it is accepted and plays back as a distorted mess that makes the company sound like it is calling from inside a tin.

Telephony audio has not moved on much: 8 kHz sample rate, mono, and usually G.711 encoding as u-law in North America and Japan or a-law elsewhere. A 44.1 kHz stereo MP3 is wrong on every axis, and converting it properly is a specific job that general audio converters handle badly.

How to judge an IVR converter

Does it have presets for real systems? Asterisk, FreePBX, 3CX and Cisco CUCM all have their own expectations. A preset is faster and safer than assembling the settings yourself.

u-law or a-law? Getting this wrong produces audio that plays but sounds harsh. It is regional, and the tool should let you choose rather than assume.

Does it handle headroom? Telephony has little dynamic range. Audio that peaks near full scale clips through the codec, and a converter that normalises sensibly saves a re-record.

Does the file leave your machine? Prompts are usually harmless. Recorded customer messages are not.

The comparison

Facts checked August 2026; tools change. Table covers only the 3 alternatives — our tool gets its own section below.

Sazrika IVR Audio Converter

The most purpose-built of the three. It outputs WAV 8 kHz mono PCM16, G.711 u-law, G.711 a-law and raw .sln for Asterisk, with one-click presets for Asterisk, FreePBX, 3CX and Cisco CUCM plus support for wideband G.722 systems. Presets matter here more than in most converters, because the difference between a working prompt and a rejected one is often a single parameter.

Its privacy position is unusually strong: decoding, resampling and encoding all run in the browser through the Web Audio API, and the page states it has no upload endpoint at all. Free, no registration. It converts rather than edits, so trimming and levelling happen elsewhere.

Kernel Manic VoIP Converter

The most flexible in one pass. It outputs WAV, GSM, u-law, a-law, G.722 and SLN, all mono at 8 kHz, aimed at Cisco Call Manager and Asterisk for message-on-hold, IVR and pre-recorded prompts — and it accepts any input FFmpeg supports, which is effectively everything. It also adjusts volume up or down by percentage and changes playback speed before download.

Volume adjustment in the same step is genuinely useful given how much telephony audio needs levelling. The trade-off is that files are uploaded through a web form rather than processed locally. No registration, free.

Audacity

The desktop answer, and the right one when the audio needs work rather than conversion. It is a free, open-source multi-track editor for Windows, macOS and Linux, and it exports to WAV among many other formats — so you can trim, normalise, remove noise and then export at the required rate and encoding.

What it does not do is hold your hand. Nothing in Audacity knows what FreePBX expects, so you set the project rate to 8000 Hz, mix to mono and pick the encoding yourself. That control is exactly what you want for a prompt with background noise, and exactly what you do not want for a straightforward format conversion.

IVR Audio Converter

Ours sets up an on-hold or IVR prompt to phone-system specification — 8 kHz mono, the correct codec, headroom, and loop-safe start and end points — in one place. The last two are the parts other converters skip: headroom because telephony clips easily, and loop-safe boundaries because on-hold audio repeats, and a click at the seam plays to every caller waiting.

What it does not do: edit the audio, adjust speed, or offer named presets for every PBX vendor. If you need to clean up a noisy recording first, Audacity is the tool for that step; if you want a preset that says “3CX”, Sazrika has it. G.711 is worth understanding if you are unsure which of u-law or a-law your region uses — it is the difference between a clean prompt and a harsh one.

Which one to pick

• A named preset for your specific PBX — Sazrika.
• Audio that needs levelling or speed changes — Kernel Manic, or Audacity.
• A noisy recording needing real editing — Audacity.
• A prompt that also has to loop cleanly — the tool below.

How to do it with IVR Audio Converter

1. Open the IVR Audio Converter and load your recording.
2. Set 8 kHz mono and pick the codec your system expects — u-law or a-law by region.
3. Leave headroom rather than maximising level; telephony codecs clip harshly.
4. For on-hold music, check the loop points before exporting, since callers hear the seam repeatedly.

The walkthrough is in how to prepare on-hold audio for a phone system. Other audio tools are in the tools directory.

You might also need

If the on-hold audio needs to repeat seamlessly, the Audio Loop Maker sets the loop points before conversion.

For trimming a long recording down to the prompt itself, the Audio Trimmer is the quicker first step.

Frequently asked questions

What format does IVR audio need to be?

Almost always 8 kHz, mono, and G.711 — u-law in North America and Japan, a-law across most of Europe and elsewhere. Some systems accept plain 16-bit PCM at 8 kHz, and newer ones support wideband G.722. Check your PBX documentation before converting.

Why does my prompt sound distorted?

Usually clipping. Telephony codecs have limited dynamic range, so audio mastered loud for music distorts badly. Leave several decibels of headroom rather than normalising to peak.

Can I just upload an MP3?

Some systems convert it for you, and the result is rarely good — an MP3 at 44.1 kHz stereo has to be downsampled and re-encoded, and doing that yourself with control produces noticeably cleaner audio than letting the PBX guess.

Final thought

Convert deliberately rather than letting the phone system do it. Every stage of automatic conversion costs quality, and telephony audio has so little to spare that one careless step is audible to every caller.

Try the free IVR Audio Converter