similar to: Monitor application inestability and high load

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 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,

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

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"

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 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

In 1.6.0, when creating a MDT, you could specify multiple --mgsnode options and it would failover between them. 1.6.3 only seems to take the last one and --mgsnode=192.168.1.252 at o2ib:192.168.1.253 at o2ib doesn''t seem to failover to the other node. Any ideas how to get around this? Robert Robert LeBlanc College of Life Sciences Computer Support Brigham Young University leblanc at

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

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

Mahmut, First of all, I'd like to reiterate what a great patch this has been. I'd also like to voice support for having an option to mix the files on the fly, and name them uniquely. While I was able to smoothly put the files together with soxmix, I see on the fly mixing as hugely beneficial to an automated solution to saving/delivering the messages without intervention. One feature,

I hope this is the right list. I have created an ext2 filesystem and removed the dir_index feature. I don't know if this kind of experimentation is going to help me learn something about filesystems or not. Well what is dir_index? Then I ran e2fsck -f -v -pD and the /dev file. Now what did I remove? Htree. I guess it can always be put back and it's on an experimental filesystem.

I decided that I would try out 2.5.44, and I noticed that htree was merged. If I don't do the tune2fs -O dir_index, and e2fsck -D, the (exisintg) fs won't use htree, right? Once I do the tune2fs and e2fsck, will I still be able to go back to a non-htree kernel if needed? (Will a htree-ized fs work on a non-htree kernel?) I'm guessing that it won't. I've seen a 2.4 htree

I've been thinking recently that we should re-enable DIR_INDEX in mke2fs by default. When it first came out, we had done this and were bitten by a few bugs in the code. However, this code has been in heavy use for several thousand filesystem years in Lustre, if not elsewhere, and I'm inclined to think it is pretty safe these days. Likewise, RHEL/FC have had RESIZE_INODE as a standard

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

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

Hi, I enabled dir_index on a filesystem and dumpe2fs -h also reports this. My understanding is that this will be used for newly created directories and that old directories can be indexed using "fsck.ext2 -D". Two questions: - Is there a way to tell is a given directory is indexed or not? - Is there a better way to index the root fs than to boot off a live CD? Best, Norbert