similar to: Call recording, monitor & soxmix in Asterisk 1.0.9

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

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 =>

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

Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix

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:

Hi, I'm having high load, choppy sound and slow responsives with an asterisk server (version 1.2.12.1) that make a peak of 90 channels (around 60 phones calling at max, isn't necessary to reach this peak to get the problem). All the traffic is SIP, with recording for every call. The server has: Intel(R) Xeon(TM) CPU 3.20GHz (with HyperThreading disabled for inestability) 4G RAM 2 DD SCSI

Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together

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

Is the path to soxmix in the $PATH environment variable when asterisk starts. If you're running from an init script it may not have path set at that point. When you log in, you set the path variable. Have you tried putting explicit paths into the command in your extensions.conf? IE /usr/bin/soxmix instead of just soxmix. HTH Chris That sorted it, thanks

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Hi, i am looking for a tool to merge the two wav files of a monitored call into one. soxmix does that well but actually merges the two channels. I would prefer a solution that creates a stereo wav file of the two mono files so you have the called party on one (e.g. left) channel and the calling party on the other (e.g. right). I can do this interactivly using audacity but i am looking for a tool

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called "extensions-phones.d" and it contains a number of .conf files. In my extensions.conf file I have put a #include "extensions-phones.d/*.conf" in my [globals] context If we reload and restart *, and

Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,

[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