similar to: Resampling Audio for use with Asterisk

I have a resource ''users''. the index page has a ''link_to'' tag I created on the line item level for a specific view. The link_to that is frustrating me in particular says: <%= link_to ''View'', :controller => ''users'', :action => ''view'' %> What''s happening is that it''s going to

Hi, up until now I've always deployed VMs with their storage located directly on the host system but as the number of VMs grows and the hardware becomes more powerful and can handle more virtual machines I'm concerned about a failure of the host taking down too many VMs in one go. As a result I'm now looking at moving to an infrastructure that uses shared storage instead so I can

Hi, in the earlier version there was a sounds.txt with the transcript of the soundfiles. Does this still exist somewhere? Is there a plan to make speech synthesis available the same way as soundfiles, ie. instead of playing language/soundfile.wav, send the text to the speechengine and play the output...? Jay... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have an upcoming project that will involve a large text file. I want to 1. read the file into R one line at a time 2. do some string manipulations on the line 3. write the line to another text file. I can handle the last two parts. Scan and read.table seem to read the whole file in at once. Since this is a very large file (several hundred thousand lines), this is not

Sometime in the last 10 days the SVN build system (or was it the stable build system?) changed to dynamically download the soundfiles instead of including them with the downloaded code. I didn't get any prompt the the soundfiles were about to be downloaded, and to my great surprise on the next trip I made into voicemail, I now have an entirely new set of sounds on my system. I don't

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to be able to send a prerecorded message to the person I am calling. I know that you can use the application map to do this. Just to test I enabled the testfeature example that is in the features.conf file. When I hit #9 during a call the other user does not hear the monkeys, they only hear a series of beeps. I have tried with different soundfiles and they all give the same problem.

Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like "You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes" I've seen on

I have this problem: When playing a PC game via wine while listening to music with let's say Amarok, there's no sound in the PC game anymore. It's like the sound channel is occupied by the music player so wine cannot playback any soundfiles. Sounds are working if I play a Linux game like Nexuiz while running some music via a music player (ok, completely different, I know, but

a nice example about dns islanding. http://retrohack.com/a-word-or-two-about-dns-islanding/ and with only 2 dc's setup the resolv.confs like : DC01 Primary DNS 10.1.1.2 Secondary DNS 127.0.0.1 DC02 Primary DNS 10.1.1.1 Secondary DNS 127.0.0.1 http://technet.microsoft.com/en-us/library/ff807362%28v=ws.10%29.aspx says: If the loopback IP address is the first entry in the list of

In beta 3 (Windows, libvorbis-1.0beta3.zip, compiled using BCB), I get a runtime overflow on line 322 when encoding a simple pure tone (44100Hz, 96kbps, 128kbps, or 160kbps). Changing the float to a double avoids the problem. float val=decay[i]*decscale; float att=fabs(f[i]/val); /// floating point overflow ... Chris http://www.goldwave.com --- >8 ---- List archives:

I need to connect an older APS UPS unit to a machine running CentOS 7. Unfortunately the UPS only has a serial port whereas the computer does not. I am aware that there are USB-serial adapters but that the hardware or the drivers might fall short of expectations. Does anyone have positive experience with such an adapter? Or, conversely, would recommend avoid a particular adapter?

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

Hi all, I am relatively new to Linux and wine and I am trying to run an audio editing software called Goldwave 5.06 (version 4.26 works relatively well). 5.06 installs ok but when I go to run it I get the following error: "Driver does not support 16 bit colour depth or visual buffer could not be created. Visual disabled." And the meter displays do not work. I have tried adjusting

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

GoldWave v 5.08 does not work. It starts, but when try to load any .wav file after pressing "Play" button it plays file endlessly. After pressing "Stop" button, playing loops - it plays endless some final part until program will closed. I had done log with +relay,+seh,+text parameters. It is relative big, so I do not attach it. Instead, I pack it with BZip2 and place to

Here are some minor things I noticed when updating the vorbis module for GoldWave: Bitwise.c, line 175 and 207 Warning: Negative unsigned value ret=-1UL; Info.c, line 385 Warning: Unreachable code break; vorbisfile.c, line 1407 Warning: Call to function with no prototype int host_endian = host_is_big_endian(); fix: add 'void' to line 1339: static int host_is_big_endian( void )

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

hello there, I just started using ogg files. And want to thank you all for this new format. I have written an addon for mirc (chatclient for irc chat networks) to play and exchange soundfiles and I am almost done with implementing the ogg vorbis format into this application. The mirc-scriptinglanguage provides me a command to read a specified number of bytes starting at a specified location of

Hi, The topic of my PhD is finding structure in a musical piece, starting from audio. The idea is to use audio descriptors in order to find repetition and maybe also transformations. Currently I am using descriptors that can be described as a small subset of the FFT of the FFT, which gives me a not-too-many-dimensional vector for one second of sound, with a stepsize of 0.1 second. I am also