similar to: Audio level in compressed wav files

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?

There's a been a long standing issue with voicemail volume levels for files saved in WAV49 format as compared to WAV format. WAV49 is much smaller in emails and that's great, but it's also less than half the volume level than the exact same voicemail saved in WAV format. I've seen this mentioned by several others over the years in the mailing list -- has there been any

Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hello list, I have the following in my voicemail.conf : [general] format=wav|wav49 When I receive a WAV-file from my Asterisk-server, I am unable to play the file... There is no player on my Fedora that wants to play the file. When I make my Asterisk-server unreachable, the incoming call goes to the voicemail of my ITSP. Don't know what system they use there but the mailing I get with the

Hi All, When our users receive a voicemail we send it attached to an email. It used to work fine, encoded in wav49 and read by Windows media player. Recently the default player in the company has become VLC which is unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I can't get it working: In voicemail.conf: format = ogg The result is as follow: [Mar 11 09:42:17]

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

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

How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play them (hxplay, xine, mplayer, etc). I think I have all the normal codec packages installed. I can play regular wav files, but they're too big. - Mike

Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf)

Thanks for your quick replies - I had no idea this list was so responsive! One more issue for you - we are trying to build a custom voicemail application and so far everything works great by using AGI to play prompts, get input and record files. However one of our customers has asked for some pretty advanced functionality, specifically the ability to: * stop in the middle of recording (so far so

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

I know this has been moved to contrib, but is anybody using it successfully? We are looking at using Asterisk for the fine IVR features it has, tied in with another platform. Calls are getting routed to it, but the following is happening: (I've redacted phone#s and passwords) asterisk log: Sep 17 11:56:24 WARNING[17423]: No entry in voicemail config file for '+13609XX2000' Sep 17

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

Hello, When a voice message is saved and e-mailed as a wav, the total time of the voice mail does not show up in, e.g., windows media player, why is this? I have only used wav49/wav: ; Use wav49 format for all voicemail messages format=wav49|gsm|wav Justin.

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I am having problems with * voicemail emails. It will only mail out the gsm encoded messages. I would really like it to use the wav49 format to make it easier on my user's. I have tried specifying it with just wav49, and like wav49|gsm. It emails nothing, but if I set it to gsm|wav49 then sure enough it emails it out. According to the wiki it should send whatever is specified first. Anyone

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was