similar to: Voicemail- Recorded Mesage Low Volume

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio

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

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

Hi, Would anyone have a clue on this issue? I'm running asterisk 1.4.13. Trying to get a WAV voicemail file attachment sent to my email address. Voicemail is working fine. Email notification of a new message works fine. However, when I set up voicemail.conf to have an attachment of the file sent to email I get an error message like this on the console: [Feb 17 13:33:37]

Hello, I need to play some musiconhold content starting at a random duration from the start. Thanks to mode=custom option and either madplay or mpg123 programs, I could successfully get what I was after on a Debian Wheezy system. Now I realized sox version on my target system (Debian Squeeze) cannot convert to MP3 format. So I'm looking after workarounds. 0. I've read many mpg123 or

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]

Does soxmix works with asterisk ver. 0.9? I have ver. sox-12.17.5 on Gentoo but the option "m" does not combine two WAV files (In and Out) into one file. I have two separate files in /monitor folder. exten => 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 711,2,Monitor(wav,${CALLFILENAME},m) exten => 711,3,Dial(${sales_support},20,r) exten =>

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

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

I am using the Monitor() application (with soxmix for combining the audios) and the voice connected to the phone network is recorded at a lower volume then the voice connected directory to the Zap analog phone card. How can I get both the audios to be at the same volume on recording? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL:

[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

Hi all: I am managing a radio station that has 7 streams. I am using icecast, ices and ezstream on the streams and overall it is working pretty well. One of the streams is an old time radio stream and I use ezstream to run the prerecorded shows. The streams are 128K mp3 streams. Also this station is running on a VPS using Ubuntu 14.04. My question: Is there a package or some sort of

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

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 =>

It seems, that oggenc-ing audiofile, and then flac-ing diffrences between original file and vorbis compressed file gives a little better compression than simply flac-ing. I've tested it on one file only: file.wav 55829468 bytes flac -8 file.wav file.flac 37924329 bytes (0.6793 of original) oggenc file.wav file.ogg 4784799 bytes oggdec -o ogg.wav file.ogg sox tmp.wav ogg-.wav vol

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 Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

When a caller hangs up (e.g. after leaving a voicemail), my British Telecom exchange sends a continuous tone for about 15s and then silence. I can't get asterisk to recognise this tone as a hangup indication. I have tried indications.conf with both country=uk and country=us. My zapata.conf has busydetect=yes, callprogress=yes and I've tried setting busycount from 1 through 7 I am using