similar to: WAV file best sound quality

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

thanks Billy. I replaced print "STREAM FILE $filename \"\"\n"; with print "EXEC PLAYBACK $filename \n"; and it worked fine. Interestingly when I did print "STREAM FILE beep \"\"\n"; within the script, it worked. If I wasnt a newbie to asterisk I wouldve thought this to be strange. >From: "William Piper"

I've been using it in a test environment with no problems. However, I haven't used it in production yet. I'm doing some voice broadcasting with a PRI and so far I'm content with the performance. -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Akpome Akpoguma Sent: Friday, February 03,

I downloaded recordPad and recorded a wav file and tried playback on asterisk got the same error as before -- WARNING [1225991360] Format.wav.c:132 check_header:unexpected header size 18-- when I recorded in gsm format on my laptop asterisk did playback well I used sox to resample the recorded wav file on the asterisk machine into wav again and asterisk playback worked well. The sound

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

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, I am trying to implement a web application that allows the user to playback WAV files inside a web browser. __________ Background __________ - The file server stores audio files as GSM 6.10 or TrueSpeech encoded WAV files. - The web server (most probably ASP.NET on IIS) offers the client the ability to search audio files on the file server (a different machine). - The client will be

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

A friend of mine made the following comment in a discussion I had with him that on a website we adminster we should offer a) WAV or maybe shorten files b) Ogg as a decent reference lossy encoded version He's been trying to convince me that we should offer MP3 (in lieu of WAV) and possibly Ogg. The audio files are primarily vocals I am not a physics guy but his statements don't

I''m building an app that allows users to sample .wav files. What is the best way to have RoR play sounds? I was just going to do something like this, but am wondering if there is a better way. Thanks! <script> function EvalSound(soundobj) { var thissound=document.getElementById(soundobj); thissound.Play(); } </script> <embed src="alf.wav" autostart=false

Hi RoR Folks, Have a MySQL database with a blob datytype with .wav files loaded into them. Would like to have a button on a form that toggles between play and pause when clicked and will play the .wav file. Any suggestions on the best way to implement this? Thanks! -- Posted via http://www.ruby-forum.com/.

I need to grab the voicemail WAV file once the voicemail command is done. Is there a hook to be notified that voicemail is done, and get the name of the recorded file? Thanks MD

Hello, Does anyone know of a Flash or Java player that can play WAV files created by Asterisk? Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101003/3b992637/attachment.htm

When I upload a pre recorded wav file using trixbox, it can't be played on the welcome message. But when I record using xlite, it works ok. trixbox required 8Khz PCM 16bit recording, I used it, but still no success. Any idea? Thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

Hi I have trouble converting a file in .wav format to .ogg format. Have any one of you have code for converting a file (.wav -> .ogg ) using Ogg-vorbis win32 SDK 1.0. If you have code please help me. Regards, Ravi. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

Hi, I am new to working with silk codec. I am doing silk codec file conversion to mp3. I am unable to decode its failing. Can you please let us know how to use and fix. The issue. Regards, Venkatarao B. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170125/a9bcc4a6/attachment-0001.html>

Hello, I'm having some problems recording calls. Here is the situation. Call come in Zap/1-1, rings through to SIP/101. Call is answered, talk talk talk, hangup Zap/1-1-in.wav is full of loud static Zap/1-1-out.wav sounds fine Call comes from SIP/101, rings to SIP/102 Call is aswered, talk talk talk, hangup SIP..in.wav sounds fine SIP..out.wav sounds fine. So, it's something about