similar to: detect volume level change on SIP channel

On Dec 17, 2009, at 11:02 PM, Brian Willoughby wrote: > > On Dec 17, 2009, at 19:00, Ron Decline wrote: >> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >>> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: >>> >>>> Is it possible to switch the Left / Right channel when encoding in FLAC? >>>> (I have

Hello all, This is my first attempt at posting a patch. So if I screw this all up, my apologies and please someone let me know without beating me up too bad. To use this patch. You need to have an extra line in /etc/asterisk/voicemail.conf that looks like volgain=10.0 The 10.0 gets passed to sox which you will need installed on your system. 10.0 is what works for me, anything over 1.0 will

Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at

On Dec 17, 2009, at 19:00, Ron Decline wrote: > On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: >> On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> >> wrote: >> >>> Is it possible to switch the Left / Right channel when encoding >>> in FLAC? >>> (I have some flac files with incorrect left/right channel

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 =>

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this?

Ok, Here's another bizarre one (no strange curve balls to throw this time :P). I have several mp3 files of some easy listening music that I pulled off some CDs we have. They sounds fine and are at a nice volume level. When I run this script I wrote: (I run it by doing ./script < filename) mpg123 -s --rate 44100 --mono $1.mp3 > $1.raw sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav

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

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

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hey guys, I'm trying to get music on hold working. I have a wav file. It plays fine on my windows laptop in all sorts of audio applications. If I put it on our asterisk 1.2.4 box and do something like: sox -V nov_2005.wav /var/lib/asterisk/mohmp3/nov_2005.raw sox: Detected file format type: wav sox: Chunk fmt sox: Chunk fact sox: Chunk data sox: Reading Wave file: Microsoft U-law format, 1

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

On Dec 18, 2009, at 12:12 AM, Brian Willoughby wrote: > > On Dec 17, 2009, at 20:42, Ron Decline wrote: >> Anyway, as you and dathead2 suggested, I can just flac -d, then swap the channels on the wav with SoX, and then flac -a, I can then use metaflac to transfer the tags. Easy enough to write a script to do the work. > > > flac -a will analyze, not encode. Apart from

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

/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

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

Hello, > 2. I don't have a good source of wav data for testing. I've noticed that > introducing bugs into speex (even gross ones like returning completely > incorrect codebook entries) tends to, sometimes subtly, degrade quality > instead of blowing up. Is there an existing set of source data - and > maybe even a test harness that will do binary comparison, so I can avoid

Can you explain? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Tiago Geada <tiago.geada at gmail.com> </div><div>Date:03/07/2014 9:04 PM (GMT+02:00) </div><div>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re: