similar to: asterisk and mp3 on 1.4.43

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

this is not really a development question, but i didn't find another mailing list to ask it. any idea why size of wav file doubles when it is encoded to speex and back to wav: $ ls -ls testi.wav 40 -rw-r--r-- 1 foo foo 40674 May 4 14:38 testi.wav $ speexenc --denoise --agc --quality 10 testi.wav testi.spx Encoding 8000 Hz audio using narrowband mode (mono) $ ls -ls testi.spx 20

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

Dan Nelson (dnelson@allantgroup.com) wrote: > I guess the problem might be oggenc's option parser, then. Given a > stereo 22050hz input file, I can't seem to get oggenc to encode less > than 22kbits. The lowest bitrate it will allow on the commandline (for > 22050hz/2ch input) is -b 30, but if you also add -M 1, it will generate > a file with an average bitrate of 22. $

Hello, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

I have the following in extensions.conf: [global] MP3ROOT=/var/lib/asterisk/mohmp3 [default] exten => 1111,1,Answer ; Answer the line exten => 1111,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => 1111,3,MP3Player(${MP3ROOT}/sample-hold.mp3) The command that runs is: 14030 pts/0 S 0:00 /usr/bin/mpg123 -q -s -b 1024 --mono -r 8000

Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed back through * with no need for an mp3 decoder, I thought I'd post the result to the list to avoid wasting

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

Hi, Was wondering if any gurus could tell me if it is possible to stream Ogg on very low bandwidth links (for example satellite phone) where the expected bandwidth would almost certainly be less than 19Kbps, and probably more like 10Kbps. I am not after great audio at this bitrate, mono with an audio bandwidth less than 5KHz. I have done some testing (not streaming though) listening to 8Kbps MP3