similar to: Asterisk Call Monitor and soxmix error

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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)

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.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

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

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:

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

Hi, How can I merge 2 gsm files into a single file? I have tried to use soxmix as below but failed. soxmix 1.gsm 2.gsm 1-2.gsm

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the

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

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 have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Roth Sent: Thursday, April 13, 2006 1:20 PM To: Asterisk Users Mailing List - Non-Commercial

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

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this

hello Newbie, i try examples to understand "asterisk". I have a pb with your macro macro-record-cleanup. the progress of the macro stops if the macro is execute on a hangup. I try many other configure with exchange of rules but it seems me that there is no execute after the first (or second) instruction after an hangup. i use two "sip" phones "x-lite", one with a

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

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