{"id":1875,"date":"2026-10-12T09:13:00","date_gmt":"2026-10-12T13:13:00","guid":{"rendered":"https:\/\/www.xconvert.com\/blog\/?p=1875"},"modified":"2026-08-31T23:05:44","modified_gmt":"2026-09-01T03:05:44","slug":"convert-audio-for-phone-system-ivr","status":"publish","type":"post","link":"https:\/\/www.xconvert.com\/blog\/convert-audio-for-phone-system-ivr","title":{"rendered":"Convert Audio for a Phone System: 8 kHz Mono WAV &#038; mu-law"},"content":{"rendered":"\n<p class=\"wp-block-paragraph\">You recorded a clean greeting in Audacity, exported it as a WAV, uploaded it to FreePBX \u2014 and got <em>\u201cIncorrect audio format.\u201d<\/em> Or it uploaded fine and callers hear static. The problem is almost never the recording: phone systems are built around a 1988 telephony standard that expects <strong>8 kHz, mono, one specific byte layout<\/strong>, and your editor exported 44.1 kHz stereo. Here is what each platform actually parses, and why \u201cWAV\u201d alone is not an answer.<\/p>\n\n\n\n<blockquote class=\"wp-block-quote is-layout-flow wp-block-quote-is-layout-flow\"><p><strong>Quick answer:<\/strong> For <strong>Asterisk, FreePBX and FusionPBX<\/strong>, convert to <strong>PCM 16-bit Little Endian, 8000 Hz, Mono<\/strong> \u2014 Asterisk\u2019s <code>.wav<\/code> reader is signed-linear, not \u00b5-law. For <strong>any system whose documentation asks for \u201cG.711\u201d<\/strong> \u2014 most Cisco, Avaya and SIP hardware \u2014 use <strong>PCM mu-law<\/strong> (or <strong>PCM A-law<\/strong> outside North America and Japan) at <strong>8000 Hz, Mono<\/strong>. <strong>3CX<\/strong> wants PCM, 8 kHz, 16-bit, mono. Three settings fix nearly every rejection: <strong>codec, sample rate, channel count<\/strong>. Do all three on <a href=\"https:\/\/www.xconvert.com\/convert-audio-to-wav\">xconvert\u2019s audio-to-WAV converter<\/a>.<\/p><\/blockquote>\n\n\n\n<h2 class=\"wp-block-heading\">Jump to a section<\/h2>\n\n\n\n<ul class=\"wp-block-list\"><li><a href=\"#why-8khz\">Why phone systems are still stuck at 8 kHz<\/a><\/li><li><a href=\"#codecs\">mu-law, A-law and 16-bit PCM: what your PBX actually parses<\/a><\/li><li><a href=\"#platforms\">What each phone system wants<\/a><\/li><li><a href=\"#mistakes\">The three settings behind almost every rejection<\/a><\/li><li><a href=\"#amr\">Where AMR fits (and where it doesn\u2019t)<\/a><\/li><li><a href=\"#tool\">Convert audio for your phone system on xconvert<\/a><\/li><li><a href=\"#faq\">FAQ<\/a><\/li><\/ul>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"why-8khz\">Why phone systems are still stuck at 8 kHz<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\">The reference point is <strong>ITU-T Recommendation G.711<\/strong>, \u201cPulse code modulation (PCM) of voice frequencies,\u201d published in November 1988. Its \u00a72 gives the sampling rate as <strong>8000 samples per second<\/strong> and its \u00a73.1 states that <strong>eight binary digits per sample<\/strong> should be used for international circuits; multiply the two and you get the <strong>64 kbit\/s<\/strong> figure the telecom world quotes, which the Recommendation itself never spells out. It also specifies <strong>two<\/strong> companding laws, <strong>A-law<\/strong> and <strong>\u00b5-law<\/strong>, rather than one.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">That single sentence explains most of the friction:<\/p>\n\n\n\n<ul class=\"wp-block-list\"><li><strong>8 kHz sampling<\/strong> caps the reproducible audio at roughly 4 kHz. Voice survives; music and sibilance do not. This is why on-hold music always sounds thin \u2014 it is the channel, not your file.<\/li><li><strong>8 bits per sample, companded.<\/strong> G.711 does not store linear 16-bit samples; it squeezes a wider dynamic range into 8 bits using a logarithmic curve. That is why a \u201cG.711 WAV\u201d is exactly half the size of a 16-bit WAV at the same sample rate.<\/li><li><strong>Two laws, not one.<\/strong> A \u00b5-law file played by an A-law decoder is not silent \u2014 it is <em>audible but distorted<\/em>, the worst kind of failure, because it looks like a bad recording.<\/li><\/ul>\n\n\n\n<p class=\"wp-block-paragraph\">Modern systems do support wideband codecs (G.722 at 16 kHz, Opus), but <strong>prompt files are still overwhelmingly stored at 8 kHz mono<\/strong> \u2014 the lowest common denominator every trunk and handset in the path carries without a transcode.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"codecs\">mu-law, A-law and 16-bit PCM: what your PBX actually parses<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\">Here is the distinction that trips people up: <strong>\u201cWAV\u201d is a container, not a codec.<\/strong> A <code>.wav<\/code> file is a RIFF wrapper with a <code>fmt <\/code> chunk that declares which encoding is inside. Three of those declarations matter for telephony:<\/p>\n\n\n\n<figure class=\"wp-block-table\"><table><thead><tr><th>Inside the WAV<\/th><th>Format tag<\/th><th>Bits\/sample<\/th><th>Bytes per minute (8 kHz mono)<\/th><th>Who asks for it<\/th><\/tr><\/thead><tbody><tr><td><strong>16-bit signed linear PCM<\/strong><\/td><td><code>0x0001<\/code><\/td><td>16<\/td><td>~960 KB<\/td><td>Asterisk \/ FreePBX <code>.wav<\/code>, 3CX, FusionPBX<\/td><\/tr><tr><td><strong>G.711 \u00b5-law<\/strong><\/td><td><code>0x0007<\/code><\/td><td>8 (companded)<\/td><td>~480 KB<\/td><td>Any system whose manual asks for \u201cG.711 u-law\u201d; North America and Japan<\/td><\/tr><tr><td><strong>G.711 A-law<\/strong><\/td><td><code>0x0006<\/code><\/td><td>8 (companded)<\/td><td>~480 KB<\/td><td>European \/ international G.711 deployments<\/td><\/tr><\/tbody><\/table><\/figure>\n\n\n\n<p class=\"wp-block-paragraph\"><em>(Bytes per minute is arithmetic from sample rate \u00d7 sample width: 8,000 \u00d7 2 \u00d7 60 for 16-bit, 8,000 \u00d7 1 \u00d7 60 for companded 8-bit, plus a few hundred bytes of header.)<\/em><\/p>\n\n\n\n<p class=\"wp-block-paragraph\">The trap: <strong>a \u00b5-law WAV and a 16-bit PCM WAV both end in <code>.wav<\/code>, and a system that reads one may reject the other outright.<\/strong> Asterisk is the clearest example. Its own documentation lists the supported file formats as <code>wav (SLIN)<\/code> and <code>WAV (GSM)<\/code> \u2014 lowercase <code>.wav<\/code> means <strong>signed linear<\/strong> (16-bit PCM), uppercase <code>.WAV<\/code> means GSM 6.10, and \u00b5-law is supported as a <strong>raw, headerless<\/strong> <code>.ulaw<\/code> file rather than as a WAV. So if your FreePBX greeting keeps failing and you carefully set it to \u201cG.711 mu-law\u201d because that is what the trunk negotiates, you have picked the one flavour Asterisk\u2019s WAV reader does not want.<\/p>\n\n\n\n<figure class=\"wp-block-image size-large\"><img loading=\"lazy\" decoding=\"async\" width=\"2400\" height=\"1464\" src=\"https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision.png\" alt=\"Decision chart: pick PCM 16-bit Little Endian at 8000 Hz mono for Asterisk, FreePBX and 3CX; PCM mu-law for Cisco, Avaya and North American G.711; PCM A-law for international G.711; AMR narrow band only for mobile voice archives\" class=\"wp-image-1873\" srcset=\"https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision.png 2400w, https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision-300x183.png 300w, https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision-1024x625.png 1024w, https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision-768x468.png 768w, https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision-1536x937.png 1536w, https:\/\/www.xconvert.com\/blog\/wp-content\/uploads\/2026\/08\/ivr-codec-decision-2048x1249.png 2048w\" sizes=\"auto, (max-width: 2400px) 100vw, 2400px\" \/><\/figure>\n\n\n\n<p class=\"wp-block-paragraph\">xconvert\u2019s <strong>PCM mu-law<\/strong> and <strong>PCM A-law<\/strong> options encode G.711 into a RIFF\/WAVE container (format tags <code>0x0007<\/code> and <code>0x0006<\/code>); <strong>PCM 16-bit Little Endian<\/strong> produces the classic <code>0x0001<\/code> signed-linear WAV. Pick from that menu using the platform table below, not by what your SIP trunk negotiates \u2014 those are two different questions.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"platforms\">What each phone system wants<\/h2>\n\n\n\n<figure class=\"wp-block-table\"><table><thead><tr><th>Platform<\/th><th>Required prompt \/ greeting format<\/th><th>Notes<\/th><\/tr><\/thead><tbody><tr><td><strong>Asterisk \/ FreePBX \/ FusionPBX<\/strong> (<code>.wav<\/code>)<\/td><td>16-bit signed linear PCM, 8000 Hz, mono<\/td><td>Asterisk docs list <code>wav (SLIN)<\/code>; the FreePBX community thread on greeting formats asks for \u201cWAV Microsoft Signed 16-bit PCM 8000hz mono\u201d<\/td><\/tr><tr><td><strong>Asterisk<\/strong> (<code>.ulaw<\/code> \/ <code>.alaw<\/code>)<\/td><td>Raw headerless G.711<\/td><td>Not a WAV \u2014 no RIFF header at all<\/td><\/tr><tr><td><strong>Asterisk<\/strong> (<code>.WAV<\/code>, uppercase)<\/td><td>GSM 6.10 in a WAV container<\/td><td><strong>xconvert does not encode GSM 6.10<\/strong> \u2014 use the SLIN route instead<\/td><\/tr><tr><td><strong>3CX<\/strong> prompts &amp; music on hold<\/td><td>WAV, PCM, 8 kHz, 16 bit, mono<\/td><td>3CX\u2019s own docs state PCM \/ mono \/ 8 kHz \/ 16-bit<\/td><\/tr><tr><td><strong>Cisco Unity Connection \/ Unity Express<\/strong><\/td><td>G.711 \u00b5-law WAV, 8 kHz, 8-bit, mono<\/td><td>We have no vendor page we can point you at for this one \u2014 check your own release notes before you batch-convert. 16-bit PCM 8 kHz mono is also widely reported to import<\/td><\/tr><tr><td><strong>Twilio<\/strong> <code>&lt;Play&gt;<\/code><\/td><td>MP3, WAV, AIFF, GSM or \u00b5-law<\/td><td>Twilio transcodes for you; files over <strong>40 minutes can drop the call<\/strong><\/td><\/tr><tr><td><strong>Generic SIP handsets \/ door stations<\/strong><\/td><td>Usually 8 kHz mono; check whether the manual says \u201cG.711\u201d or \u201cPCM\u201d<\/td><td>\u201cG.711\u201d in a manual almost always means \u00b5-law or A-law, not 16-bit<\/td><\/tr><\/tbody><\/table><\/figure>\n\n\n\n<p class=\"wp-block-paragraph\">Twilio is the useful outlier: it accepts almost anything and transcodes server-side, and its documentation notes that <em>\u201cHigh bitrate, lossy encoded files, such as 128kbps mp3 files, will take longer to transcode and potentially sound worse than files that are in lossless 8kbps formats.\u201d<\/em> Even on a platform that accepts your 44.1 kHz MP3, handing it a properly prepared 8 kHz file produces a better result \u2014 the downsample happens either way, and doing it once cleanly beats doing it inside a real-time transcoder.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"mistakes\">The three settings behind almost every rejection<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\"><strong>1. Sample rate.<\/strong> Editors default to 44.1 kHz or 48 kHz. A 44.1 kHz file handed to a system expecting 8 kHz either fails validation or plays at the wrong speed. Set <strong>8000 Hz<\/strong> explicitly.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\"><strong>2. Channel count.<\/strong> Every telephony format above is <strong>mono<\/strong>, and uploading stereo is the second most common rejection. In a desktop editor, <em>deleting<\/em> one channel of a stereo track is not the same as mixing to mono \u2014 you lose everything panned to the discarded side. A converter that downmixes properly avoids this entirely.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\"><strong>3. Codec inside the container.<\/strong> Covered above \u2014 the one people get wrong <em>after<\/em> fixing the first two, because \u201cWAV\u201d feels like it should be enough.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">Two smaller ones, both our own advice rather than a vendor rule: <strong>trim the dead air<\/strong> off the head and tail of a prompt, but leave the cut a hair short of the first and last syllable, because an abrupt cut on a non-zero sample produces an audible click. And keep levels conservative \u2014 an 8-bit companded channel has far less headroom than a 16-bit master, so a prompt normalised for headphones often sounds harsh over a handset.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"amr\">Where AMR fits (and where it doesn\u2019t)<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\"><strong>AMR is not an IVR format.<\/strong> Adaptive Multi-Rate is the codec cellular networks used for the voice channel itself and that older handsets used for voice memos \u2014 it is what you find inside a <code>.amr<\/code> file recovered from a feature phone. No mainstream PBX takes an AMR file as a prompt.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">It matters in the other direction: if you are digitising old voice recordings, AMR is what you convert <em>from<\/em>. xconvert\u2019s <a href=\"https:\/\/www.xconvert.com\/convert-wav-to-amr\">WAV to AMR converter<\/a> offers <strong>AMR Narrow Band<\/strong> (8000 Hz) and <strong>AMR Wide Band<\/strong> (16000 Hz), mono only in both cases \u2014 the format has no stereo mode. At AMR-NB\u2019s top rate of 12.2 kbps a minute of speech is roughly 92 KB, about a fifth of a G.711 WAV, which is why it survived on storage-starved handsets. To play those files on anything modern, <a href=\"https:\/\/www.xconvert.com\/blog\/convert-amr-to-mp3\/\">converting AMR to MP3<\/a> is the simpler route.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">One honest limitation: <strong>xconvert does not encode GSM 6.10 or G.729.<\/strong> If your system will only take one of those, prepare a 16-bit PCM 8 kHz mono WAV here and run the final step in your PBX. For GSM that is straightforward \u2014 Asterisk\u2019s own codec table marks <code>codec_gsm<\/code> as distributed with Asterisk and needing no commercial licence. G.729 is the harder one: the same table marks <code>codec_g729a<\/code> as <strong>Codec(NO) Format(YES)<\/strong> \u2014 Asterisk ships the file-format module but not the transcoder \u2014 and flags a commercial licence as required, so you need a licensed codec module to encode it at all.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"tool\">Convert audio for your phone system on xconvert<\/h2>\n\n\n\n<ol class=\"wp-block-list\"><li>Open <a href=\"https:\/\/www.xconvert.com\/convert-audio-to-wav\">xconvert.com\/convert-audio-to-wav<\/a> and click <strong>Upload<\/strong> to add your MP3, M4A, WAV or other recording.<\/li><li>Open <strong>Advanced Options<\/strong>, then click <strong>Show All Options<\/strong>. <strong>Audio Codec<\/strong> is an advanced control \u2014 it is not on the default Advanced Options panel, which is why so many people never find it.<\/li><li>Under <strong>Audio Codec<\/strong>, choose <strong>PCM 16-bit Little Endian<\/strong> for Asterisk \/ FreePBX \/ 3CX, or <strong>PCM mu-law<\/strong> (<strong>PCM A-law<\/strong> outside North America and Japan) for any system whose manual asks for G.711.<\/li><li>Under <strong>Audio Sample Rate<\/strong>, select <strong>8000 HZ<\/strong>. Under <strong>Audio Channel<\/strong>, select <strong>MONO<\/strong>. Both are required \u2014 leaving either on the original value is the usual cause of a rejected upload.<\/li><li>Optionally use <strong>Trim<\/strong> with a <strong>Start time<\/strong> and <strong>Duration<\/strong> to cut dead air off the head and tail of the prompt.<\/li><li>Click <strong>Convert<\/strong> and download the WAV, then upload it to your PBX\u2019s recordings module.<\/li><\/ol>\n\n\n\n<p class=\"wp-block-paragraph\">Your file uploads over an encrypted connection, is processed on our servers, and is automatically deleted a few hours later.<\/p>\n\n\n\n<p class=\"wp-block-paragraph\">Related: <a href=\"https:\/\/www.xconvert.com\/blog\/convert-audio-to-16-bit-44-khz-wav\/\">Convert audio to 16-bit \/ 44.1 kHz WAV<\/a> covers the CD-quality end of the same three controls, and <a href=\"https:\/\/www.xconvert.com\/blog\/best-audio-format-voice-recordings\/\">Best audio format for voice recordings<\/a> covers the size-versus-clarity tradeoff when the destination <em>isn\u2019t<\/em> a phone system.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\" id=\"faq\">FAQ<\/h2>\n\n\n\n<h5 class=\"wp-block-heading\">What audio format does FreePBX need for system recordings?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">A WAV containing <strong>16-bit signed linear PCM at 8000 Hz, mono<\/strong>. FreePBX sits on Asterisk, and Asterisk\u2019s documentation lists its lowercase <code>.wav<\/code> support as <code>wav (SLIN)<\/code> \u2014 signed linear. The 2017 FreePBX community thread on greeting formats says the same thing in the users\u2019 own words \u2014 the format wanted is \u201cWAV Microsoft Signed 16-bit PCM 8000hz mono\u201d, and a FreePBX developer walks through \u201cwhat a 16bit 8KHz mono audio file looks like\u201d in Audacity. Choosing \u00b5-law here is the most common wrong turn, because \u201cG.711 \u00b5-law\u201d is what your <em>trunk<\/em> uses, not what the <em>file reader<\/em> wants.<\/p>\n\n\n\n<h5 class=\"wp-block-heading\">Should I use mu-law or A-law?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">Geography decides it. <strong>\u00b5-law<\/strong> is the deployed convention in North America and Japan; <strong>A-law<\/strong> is used across most of the rest of the world, and G.711 designates A-law for international circuits between countries using different laws. If your platform\u2019s docs just say \u201cG.711\u201d with no qualifier and you are in the US or Canada, it means \u00b5-law. Getting this backwards produces audio that plays but sounds distorted rather than failing outright \u2014 so if a prompt is audible but harsh and gritty, swap the law before you re-record.<\/p>\n\n\n\n<h5 class=\"wp-block-heading\">Why does my IVR prompt sound muffled or thin?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">Because it is band-limited to about 4 kHz by the 8 kHz sample rate \u2014 inherent to the channel, not to your file. Uploading a 48 kHz master will not fix it; the system downsamples anyway. What you <em>can<\/em> control: record close to the mic, keep the delivery clear rather than breathy, and avoid background music, which loses more to the 4 kHz ceiling than speech does.<\/p>\n\n\n\n<h5 class=\"wp-block-heading\">Can I upload an MP3 to my phone system instead?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">Rarely the right answer. Twilio explicitly accepts MP3 for <code>&lt;Play&gt;<\/code> and transcodes it, but most on-premises PBXes do not \u2014 Asterisk has no MP3 playback format in its standard build, and 3CX asks for WAV. Even where MP3 works, Twilio\u2019s own docs warn that high-bitrate lossy files take longer to transcode and can sound worse than a lossless 8 kHz file.<\/p>\n\n\n\n<h5 class=\"wp-block-heading\">How big should an IVR prompt file be?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">Small, by design. At 8 kHz mono, <strong>16-bit PCM is about 960 KB per minute and G.711 \u00b5-law\/A-law about 480 KB per minute<\/strong> \u2014 so a 20-second greeting lands near 320 KB and 160 KB respectively. If your export is several megabytes for a 20-second prompt, you are still at 44.1 kHz or still in stereo.<\/p>\n\n\n\n<h5 class=\"wp-block-heading\">Does xconvert support GSM or G.729 for Asterisk?<\/h5>\n\n\n\n<p class=\"wp-block-paragraph\">No \u2014 neither GSM 6.10 nor G.729 is in our encoder list. What we cover is the input side of both: convert to <strong>PCM 16-bit Little Endian, 8000 Hz, mono<\/strong> here, then let Asterisk or FreePBX handle the last step. GSM is easy \u2014 <code>codec_gsm<\/code> ships with Asterisk. G.729 is not: Asterisk\u2019s codec table lists <code>codec_g729a<\/code> as <strong>Codec(NO) Format(YES)<\/strong>, meaning Asterisk can play a <code>.g729<\/code> file but cannot transcode into one without a separately licensed codec module. That licensing is also why G.729 stays out of most general-purpose converters.<\/p>\n\n\n\n<h2 class=\"wp-block-heading\">Sources<\/h2>\n\n\n\n<p class=\"wp-block-paragraph\"><em>Last verified 2026-08-31.<\/em><\/p>\n\n\n\n<ul class=\"wp-block-list\"><li><a href=\"https:\/\/www.itu.int\/ITU-T\/recommendations\/rec.aspx?rec=G.711\">ITU-T Recommendation G.711<\/a> \u2014 the ITU\u2019s own record for the Recommendation: title \u201cPulse code modulation (PCM) of voice frequencies\u201d, approval date 25 November 1988, status \u201cIn force\u201d. Note that this record page carries no technical figures. The numbers above come from the Recommendation text itself: \u00a72 (\u201cThe nominal value recommended for the sampling rate is 8000 samples per second\u201d), \u00a73.1 (\u201cEight binary digits per sample should be used for international circuits\u201d), \u00a73.2 (the A-law and \u00b5-law encoding laws) and \u00a73.4 (paths \u201cbetween countries which have adopted different encoding laws should carry signals encoded in accordance with the A-law\u201d). The familiar 64 kbit\/s figure is \u00a72 and \u00a73.1 multiplied, not a line anywhere in the text.<\/li><li><a href=\"https:\/\/docs.asterisk.org\/Operation\/Asterisk-Audio-and-Video-Capabilities\/\">Asterisk Documentation \u2014 Asterisk Audio and Video Capabilities<\/a> \u2014 lists supported file formats including <code>wav (SLIN)<\/code>, <code>WAV (GSM)<\/code>, <code>.ulaw<\/code>, <code>.alaw<\/code>, <code>.gsm<\/code>, <code>.g722<\/code> and the <code>.sln<\/code> sample-rate family; and the codec table whose \u201cDistributed w\/ Asterisk?\u201d column reads YES for GSM (<code>codec_gsm<\/code>, commercial licence NO) and <strong>Codec(NO) Format(YES)<\/strong> for G.729A (<code>codec_g729a<\/code>, commercial licence YES).<\/li><li><a href=\"https:\/\/www.3cx.com\/docs\/system-prompts\/\">3CX \u2014 How to create 3CX supported System Prompts Sets<\/a> \u2014 the supported System Prompts format stated as WAV \/ PCM \/ Mono \/ 8 kHz \/ 16 bit \/ 128 kbps.<\/li><li><a href=\"https:\/\/www.twilio.com\/docs\/voice\/twiml\/play\">Twilio Docs \u2014 TwiML <code>&lt;Play&gt;<\/code><\/a> \u2014 accepted MIME types, the transcoding-quality note, and the 40-minute playback caveat.<\/li><li><a href=\"https:\/\/community.freepbx.org\/t\/correct-audio-formats-for-greeting-and-system-recordings\/42304\">FreePBX Community \u2014 Correct audio formats for greetings and system recordings<\/a> \u2014 a single 12-post thread from June\u2013July 2017 in which the requested format is stated as \u201cWAV Microsoft Signed 16-bit PCM 8000hz mono\u201d and a FreePBX developer demonstrates \u201cwhat a 16bit 8KHz mono audio file looks like\u201d.<\/li><\/ul>\n","protected":false},"excerpt":{"rendered":"<p>Phone systems need 8 kHz mono WAV \u2014 but which codec? PCM 16-bit for FreePBX and 3CX, mu-law for Cisco. Convert audio for your IVR in three settings.<\/p>\n","protected":false},"author":3,"featured_media":1872,"comment_status":"closed","ping_status":"open","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[5,14],"tags":[],"class_list":["post-1875","post","type-post","status-publish","format-standard","has-post-thumbnail","hentry","category-how-to-guides","category-tools"],"yoast_head":"<!-- This site is optimized with the Yoast SEO plugin v27.9 - https:\/\/yoast.com\/product\/yoast-seo-wordpress\/ -->\n<title>Convert Audio for a Phone System: 8 kHz Mono WAV &amp; mu-law<\/title>\n<meta name=\"description\" content=\"Phone systems need 8 kHz mono WAV \u2014 but which codec? PCM 16-bit for FreePBX and 3CX, mu-law for Cisco. 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