similar to: Voicemail volume wav vs. wav49

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?

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.

Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. -- ------------------------------------------------------------

Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=0000254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

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

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says "Don't Change the Format Unless You REALLY Know What

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

Can anyone point me in the direction of a utility which will let me determine the length (in seconds) of a wav49 file created by Asterisk? Many thanks, Malcolm

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

Hello all, I first want to thank everyone for all your contributions. I've building an asterisk system for a month or so now and without everyone in the online asterisk community I wouldn't have made it this far yet. Thanks! ...ok, mushiness out of the way.. :) I am looking for a failover and ultimately a load balancing asterisk solution. I've done a good bit of research and I

Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the "wav" format are about twice the level of those made using "WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed and about 10 times the size of the other formats. -brian

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

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail server. However, the voicemail server now has no knowledge of the location details of the phones,

The special wav49 ms hack doesn't produce correct WAV files on output. they cannot be played back on windows media player nor winamp. for example a typcial voicemail message with 9 seconds stops after 6 seconds playback with the error: "invalid fileformat (Error=8004022F)" attached is such a file. in the meantime I use the uncompressed wav format and postprocess this in