similar to: Can't capture "-1" return on Dial command

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

Dear R group, I am trying to plot uisng a for loop. Here is the script: for (i in unique(cfn$ID)){ plot(cfn$TIME[cfn$ID==i],cfn$DV[cfn$ID==i],pch=16)} I could access only the last plot of the series and cannot go back to see all the plots. I appreciate your help in resolving this issue and also please suggest how I can export the plots as a pdf or other format files for publication purposes.

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

Dear all, I have written a little R program to convert images. See below. Within the loop over j (the filenames) memory consumption grows constantly. rm( ... ) inside the loop did not help. Memory does not grow if I remove the writeBin statements between the two #-------- marks. But obviously this is not solution I want... Thanks for any advice. Manfred Baumstark P.S. As I'm new to R:

Good afternoon, We are running callgrind and cg_annotate version 3.6.1 on Centos Linux Version 5.5 x86_32. One month ago Mr. Josef Weidenorfer issued a special patch that fixed callgrind on Centos Linux Version 5.5 x86_32. We can now profile complex C++ programs which use our own shared library libmdMatchup.so. However, when we use version 3.6.1 cg_annotate callgrind.out.22533 --auto = yes

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

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,

Hi all, I have experimented a segmentation fault error using RMySQL_0.4-6 library in R 1.5.1. Look the sequence: ------------------------------------------- > library(RMySQL) > m <- dbManager("MySQL") > m <MySQLManager:(1137)> > describe(m) <MySQLManager:(1137)> Driver name: MySQL Max connections: 10 Conn. processed: 0 Default records per

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)

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 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

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"

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

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

----- 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 >

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 =>

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 =>

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: