similar to: <<< GSM Audio Files >>>

Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh "Stream File" or "Get Data" in my agi. It seems that there may be more than one GSM file type (with

Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The last part of the question is

I couldn't find any specific reference to this but maybe Im missing something completly... anyways, when trying to mix a few wav files together post-recording (the -in/-out files) using a pretty vanila soxmix line, I get the error: Done Mixing OUT115-20060215-150749-1139976460.7898-out.WAV..... Mixing OUT115-20060215-155022-1139979011.8787..... /usr/bin/soxmix: invalid wav gsm frame size: 1

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. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

Hi 1. What is the best way to convert wav (44000 Khz) to gsm format for asterisk ? I;ve tried sox command but the outcome is not satisfying...The built-in gsm files shipped with asterisk are simply superb ..How do i create gsm files of similar quality ? Can anyone help me out ? if sox is the only way can anyone tell me the exact command ? 2. Can Freepbx 2.5 installed above asterisk 1.6.0 or

On 13:05 Tue 27 Mar , Apollon Oikonomopoulos wrote: > On 11:31 Tue 27 Mar , Apollon Oikonomopoulos wrote: > It turns out there's a missing byte-inversion when loading the blocks > which should be addressed in getblock{32,64}. Murmurhash treats each > block as an integer expecting little-endian storage. Applying this > additional change fixes the build on s390x (and

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

Hi, On 12:55 Mon 26 Mar , Josef 'Jeff' Sipek wrote: > On Mon, Mar 26, 2018 at 15:57:01 +0300, Apollon Oikonomopoulos wrote: > ... > > I'd be happy to test the patch, thanks! > > Ok, try the attached patch. (It is a first pass at the issue, so it may not > be the final diff that'll end up getting committed. It'd be good to know if > it actually

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

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

I'm on Centos 5.5, and would like to use sox to strip out any periods of silence > 5 seconds from a batch mp3 audio files. Googling I found sox, but it does not seem to support mp3 files by default. The man page says: .mp3 MP3 Compressed Audio MP3 audio files come from the MPEG standards for audio and video compression. They are a lossy compression format that achieves

Hello, Does anyone know of an audio player for Linux that will play asterisk's GSM files? I know I can convert them using sox, but I'm hoping to play them natively so I can test out some voicemail settings. Thanks! -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net

I would like to convert a file to WAV49 for use with Asterisk using the linux command line. Specifically I would like to upload sounds to use for unavail.wav and busy.wav, but I'd like them to be compressed so that space is not wasted. I tried using SOX but havent found the correct command-line options to make it or Asterisk happy. Ideas?

Good day, a quick bug while encoding speexes in speexenc. While reading a .wav in function read_samples, size can get to negative values, as long as fread is able to go forward with reading the file. This can happen with Protools-generated wavs, where they put additional information after the "data" fourcc. I created a very quick hack by adding: int tentative_read =

I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about

Hi, The dovecot 2.3.0.1 Debian package currently fails to build on all big-endian architectures[1], due to murmurhash3 tests failing. The relevant output from e.g. s390x is: test-murmurhash3.c:22: Assert(#8) failed: memcmp(result, vectors[i].result, sizeof(result)) == 0 test-murmurhash3.c:22: Assert(#11) failed: memcmp(result, vectors[i].result, sizeof(result)) == 0 test-murmurhash3.c:22:

OpenSSL 1.1 features a cleanup function that is automatically run on shutdown using atexit(3). This function frees all OpenSSL-allocated resources. In dovecot, OpenSSL is loaded indirectly using dlopen(3) against the relevant dovecot crypto module and is finally unloaded using dlclose(3). Until OpenSSL 1.0.1c this worked fine, however OpenSSL 1.0.1c makes sure[1] that the library stays loaded

On 5 June 2016 at 21:34, Frank Esposito <fpesposito at gmail.com> wrote: > Hello -- > > I have a flac file encode 92/24bit -- what tools are available to > convert these files to wave 44.1/16 bit so then I can convert to > mp3? Thanks > > -- > Frank Esposito Don't you mean 96/24? Also, which platform (*nix, Mac, Windows)? Off the top of my head: Audacity,

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