Correct Audio Formats For Greeting and System Recordings

I have used the correct Audio formats (I suppose)

WAV Microsoft Signed 16-bit PCM 8000hz mono (I am surprised searching the forums on this turns up no results)

But when I log into the UCP I can not upload a greeting. It just hangs (all night I have confirmed) regardless of browser. I also want to confirm the audio format for the IVR/System Records.

I am using http://www.audacityteam.org/download/ Audacity to create the files.

I dont know why this info isnt listed where you upload, and it offers half a dozen formats for you to “convert to”?. That would lead me to believe you could upload anything and convert it to a supported format. So that is confusing since everyone I have ever seen discuss this mentions a very specific audio format and bit rate, etc, etc need to be used.

https://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk has a pretty good discussion of converting from various file formats to “native” file formats for Asterisk.

It doesn’t solve your immediate problem (there’s probably one of the things that’s causing a problem) but it should get you closer to solving the native problem. If you can get the files onto the actual phone server, you can do a lot from the command line and see what kinds of errors “sox” will give you.

Is the formatting of Asterisk supported by FreePBX? Doesnt seem to be the case.

you can record your greeting direct from the PBX, just dial *97 and fallow the instruction, after that if want to edit the audio you can download the AUDIO from the PBX

It is but the file you are using is terrible. Google can’t play it and sox crashes outright

[root@freepbxdev1 ~]# /usr/bin/sox '/tmp/main_greeting.wav' -r '48000' -b '16' -c 1 '/var/spool/asterisk/tmp/temp.1499119387960.wav'
Segmentation fault

Probably from extensive attempts to convert it. What program should I record from the old phone system with, for example? iPhone recorder? Android?

The Audicity app seems to save it back to 32bit no matter what, which is what I uploaded.

At any rate, when I did upload a voicemail greeting to users in the UCP they convert succesfully but when you call that extension the greeting doesnt play. Does there need to be an initial voicemail setup performed by the user first?

No but you need to upload a file first that won’t crash underlying programs. You keep jumping around in this thread and all I can say is that we need to work on one issue at a time

No. The file you uploaded is a 16bit 22050 Hz mono audio file.

Here is what a 16bit 8KHz mono audio file looks like


I guess I have two threads, sorry. What did you use to convert?

I used Adobe Audition but I also used Audacity which created this file


Which also worked.

And just in case you missed it, any resultant recording needs to be “readable” by the asterisk user, no matter where or what it is.