similar to: Monitor, append audio?

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

When I call inbound with a cell phone (via SIP PSTN trunk) some of my prompts the first word is cut off. I'm assuming the prompt is needing to be transcoded on the fly and it's not getting transcoded fast enough. I did a file convert to create gsm versions (currently they are referenced in my dial plan with no extension Seem to have same problem. How do I determine which file

I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call:

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone

sox works great. to splice out a 10-minute segment starting 12m34s into a .wav file: sox infile.wav outfile.wav trim 12:34 10:00 to splice two segments together, well: cat seg1.wav seg2.wav > joined.wav <p>samurai.fm wrote: > SOX might work? > > -----Original Message----- > From: owner-icecast@xiph.org [mailto:owner-icecast@xiph.org] On Behalf Of > Mailing List Receiver

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

When I use sox-12.17.5 recording and mixing works fine but removing the -in.wav and -out.wav file doesn't work. When I tried sox-12.17.6 recording doesn't work but removing the IN / OUT wav files is working. Anybody has a similar experience. The command didn't change but it seems to me there is a timing error: The creation time for the file is: Jan 29 15:30 18-20050129-152954-in.wav

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after

Hello All, I am using monitor with soxmix, however the quality seems somewhat low after sox converts to mp3. Does anyone know a way to get a higher quality file? Some of my lines are coming in on isdn. Regards, Greg

I''m trying to test some code that has validates each and I''ve got a very strange failure Mock ''Book_1027'' expected :store_with_privacy? with (#<Clip:0x1a99b96 @name="Clip_1025">) but received it with (#<Clip:0x1a99b96 @name="Clip_1025">) The Spec it "should check that a book can save a clip" do @user =

I'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

First, our current setup. I'm setting up a Samba 3.4 environment to replace our old Samba 3.0.x setup and make ready for Windows 7. There are several different campus locations, and each has it's own Samba PDC. Due to some (possibly poor) decisions made early in the development of the system, instead of giving each campus its own domain name, we used the algorithmic rid base parameter in

Hi, I have started using the call recording facilities in Asterisk 1.2 recently, and having worked out some of the foibles regarding call forwarding etc etc, I think I have a mostly working system. I do still seem to have a problem with recording volume though. It seems that all SIP call legs are recorded at "normal" volume, but all my Zap (ISDN) and IAX (via Provider -> ISDN) calls

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

Hi, I sent the message below a couple of months ago but I got no reply. Now that there's more activity perhaps someone could kindly provide me some guidance. Theora is great and the developers are doing a wonderful job, so I'd like to implement the codec in a way which does it justice... Original post: I wrote a plugin to enable LiVES (a video editor) to encode theora/vorbis/ogg files.

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

I'm trying to figure out the best way to record a conference. Many people suggest something like this: exten => 2060,1,Answer exten => 2060,2,Wait(1) exten => 2060,3,Monitor(wav,myfilename) exten => 2060,4,Meetme(1,ps) However, this creates two files for each user that connects to the meetme. (I know they can be mux'd together to make one with sox..I've done that too)

Hello group, I'm not sure this is the right place to ask a user-question instead of a dev-question - if not, a pointer in the proper direction would be appreciated. My issue is that I want to automatically (in a nightly batch job) convert a large number of voice recordings from A-Law WAV to Speex, in order to reduce storage and bandwidth demands for tranferring the files over our LAN. In my