Uploaded System Recording

I uploaded a wav file (its possible its not formatted properly, red boxes started flying up in the right hand corner. Now I am staring at the lovely “Can not connect to asterisk screen”

I did not create a backup, was a newer installation and was JUST ABOUT to hit that snapshot button. Not sure where I should even start.

Some errors are suggesting that I … .“Error From Module Fax: 'Unable to locate the FreePBX BMO Class ‘Userman’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install userman 2) fwconsole ma enable userman’”

And… hoping I dont have to stay up all night to rebuild a freepbx installation.

Since I pretty much had to reload, I would like to confirm the correct wav format to upload system records (and voicemail greetings).

I believe it was Microsoft PCM 8khz mono or something like that.

Also is there anyway to upload custom greetings to the individual voicemail greetings of each extension?

The only way I know how to upload the custom audio to the users voicemail greetings through UCP.

I’m shocked that simply uploading a file, not formatted properly or not, can bring down your whole system. This just doesn’t seem right, I mean users have the ability to upload audio files through UCP. This would mean that a user by way of uploading the wrong file can bring down the whole PBX???

I logged in, ran updates on modules, then went to upload some custom recordings that came via email.

I am still say I used the wrong encoding until someone confirms the correct one again. Maybe it was an update that broke it?

Shame on me for not making a backup first though. I hadnt kicked on the auto-backup yet as it was a new/quick installation.

@frankb thanks for the tip on the UCP. I assigned the receptionist access to all VM and can easily upload them all from them.

Here, for example http://audio.online-convert.com/convert-to-wav a free encoder with a basket full of of PCM options under advanced.

How does a man chose?

@tm1000 can correct me if I’m wrong, but the whole point of using the GUI to upload the sound file is for FreePBX to generate the sound files with the correct encoding, with the correct file extensions in the correct folders. If that is not working for you, perhaps your audio file is in a format the GUI converter doesn’t like.

@lgaetz you are correct

I am going out of my way to get the file format into something the GUI does like. And still failing.

For example changing a sample rate for the UCP greeting allowed me to finally upload the file (and quickly). But the greeting still doesnt play when calling that users voicemail.

The only thing I have ever known to work is to upload the audio file in a very specific encoding, format, bit rate, etc. That seems to be the case with all my installs, including the reload I just did last night.

Upload your not working files somewhere

https://drive.google.com/file/d/0BzcwG6f4o-ckaGNaYXRBTkdkN0E/view?usp=sharing

Here is one I recorded using TWIML API to call in and get a greeting off a 15 year old phone system that we had no way to access the recording files on.

I trimmed it down and also attempted to convert it to a Microsoft format PCM 16 8000hz mono file.

I save all of my recordings as wav files and upload them. The interface as @lgaetz stated performs the conversions and places the files correctly. Are you using the district version of FreePBX?

Not mine. Many settings. What app do you make them in?

Audacity

Not even google can play your audio file

file Downloads/main\ greeting.wav
Downloads/main greeting.wav: RIFF (little-endian) data, WAVE audio, Microsoft ADPCM, mono 22050 Hz

and

soxi Downloads/main\ greeting.wav

Input File : 'Downloads/main greeting.wav’
Channels : 1
Sample Rate : 22050
Precision : 14-bit
Duration : 00:00:41.12 = 906752 samples ~ 3084.19 CDDA sectors
File Size : 459k
Bit Rate : 89.3k
Sample Encoding: 4-bit MS ADPCM

(Not going to work not in acceptable asterisk .wav format needs to be

RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

soxi should see

Channels : 1
Sample Rate : 8000
Precision : 16-bit
Duration : 00:00:00.80 = 6400 samples ~ 60 CDDA sectors
File Size : 12.8k
Bit Rate : 128k
Sample Encoding: 16-bit Signed Integer PCM

)

I suggest you have sox convert it on the pbx itself

https://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk

Sox crashes outright

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

Okay, so I am looking at this is the acceptable format…

RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

And I am just trying to find a program that can convert to that, or maybe some settings. I downloaded Audicity (which is what the linked file was converted with) and apparently I am doing something wrong.

I had a really good app for this about a year ago that worked, now I cant find it. Its rare that I have to dub the old phone system messages to get the recordings in place.

Knowing that uploaded poorly formatted files will crash the system is helpful.

Well I used sox to convert your file

sox --version
sox: SoX v14.4.1

so

sox main\ greeting.wav -r ‘8000’ -b ‘16’ -c 1 main_greeting.wav

(resample at 8000 Hz, bitspersample 16 channels 1 )

works just fine.

soxi main_greeting.wav

Input File : 'main_greeting.wav’
Channels : 1
Sample Rate : 8000
Precision : 16-bit
Duration : 00:00:41.12 = 328980 samples ~ 3084.19 CDDA sectors
File Size : 658k
Bit Rate : 128k
Sample Encoding: 16-bit Signed Integer PCM