similar to: Subject: Re: SoxMix - Fails to Execute

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

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 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

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,

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

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 couldn't find any specific reference to this but maybe Im missing something completly... anyways, when trying to mix a few wav files together post-recording (the -in/-out files) using a pretty vanila soxmix line, I get the error: Done Mixing OUT115-20060215-150749-1139976460.7898-out.WAV..... Mixing OUT115-20060215-155022-1139979011.8787..... /usr/bin/soxmix: invalid wav gsm frame size: 1

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= 1111,CS,cs@x.com,,delete=yes Thanks, Chris

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

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 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:

Thanks for the answer Matthew. > > > > 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. > > > What

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

Due to management concerns our asterisk system has been setup to record all phone calls for some time now (before the 1.0 release). Everything was working fine until we upgraded 1.0.5 where all calls are recorded except those that pass through a queue (we are not using the queue record functionality because there are some minor issues with using it in our scenario). Specifically, the