similar to: IMAP traffic logging

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

Hi all, I am a biostatistician and I have developed my own ranking system for clinical data. I would like to test the efficiency of it w.r.t. to other ranking systems. I would like to simulate the data and after assigning ranks to my observed scores(after neglecting dropouts), observe the type I error. If I want to do a Kruuskal Wallis type of test, what test statistic should I use to test for a

Hey all. This morning I found that my audio playback is randomly sprinkled with sound skips and dropouts. I went to /var/log/yum.log and found this: Feb 09 20:18:22 Updated: lame-3.99.4-2.el6.rf.i686 I'm not saying that caused the problem but it's all I could find that changed. When I ls in /dev there is no dsp entry. That would explain why mplayer Can't open audio device

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it

I'm not sure if it's even intended to work in this particular case, but there's a problem when chroot is explicitely defined and rawlogging is used. Rawlog respects chroot point, without being actually chrooted. Consider configurtion as below, with two cases (dovecot.rawlog dir is of course created earlier, with appropriate uid/gid). 1) dovecot.userdb:

Hello, I am currently pricing out various T1 and PRI options for a client of mine. We need voice and data - we want T's. Whether it be two seperate T's, two superate fractional T's, or one combined fractional T, we need it done. We are getting pricing and one provider is telling us that they have quality issues with the "Integrated Access" product. From what they

I have time-series data from a pair of inexpensive self-logging 3-axis accelerometers (http://www.gcdataconcepts.com/xlr8r-1.html). Since I'm not sure of the vibration/shock spectrum I'm measuring, for my initial sensor characterization run the units were mounted together with the sample rate set to the maximum of 640 samples/sec. Unfortunately, at this sample rate there are significant

I am trying to monitor with Nagios that my phones are consistently checking into dovecot via IMAP. I used to be able to parse syslog for imap-login messages. I have since updated my phones, the phoness email client (k9-mail) and dovecot. Now the imap-login messages are few and far between. However, I can clearly see via tcpdump that some traffic is happening when I click refresh on k9. I enabled

Hi, this is my first post to the list after years of passive following every now and then. Dovecot: 2.0.13 (3d07ab746a67) prebuilt binaries from xi.rename-it.nl OS: i686 Debian 6.0 squeeze/sid I am about to run a setup where users are virtual and their mail is owned by vmail user. I bumped into minor client issue and needed to enable rawlogging according to

Howdy all... When making SIP calls through my X100P from X-Lite to the PSTN I'm getting 3-5 second dropouts in both directions. I've tried ulaw and GSM, but that doesn't seem to make a difference, and the * box is on my local net. Here's my hardware: Celeron 2.4GHz, 512MB, Slackware 9.1, 2 X100P, 1 T100P. Any ideas what could be happening, or pointers as to how to shoot this

First of all, *thanks* to all the developers for their hard work on Ogg Vorbis. It's greatly appreciated. I'm having a problem with encoding in Debian GNU/Linux 3.0. The oggs contain occasional dropouts and "blips" here and there. Example files: Original (~7 MB): http://personal.inet.fi/musiikki/nebularia/test/t2title.flac Ogg Vorbis file (~800 KB):

Hello everyone, Does anyone out there have actual experience with running * on a mini-itx board from VIA? They look good, but I have some reserves because of VIA's problems with PCI latency in recent years (audio dropouts, wierd things happening). I am looking at the EPIA CL-10000. For $270, I can get a CL-10000 (1ghz C3, dual ethernets, etc), 256mb RAM, and a nice small (12 x 2 x 11)

Hello, I need help in performing a Van_der_Waerden normal scores test in R. I have two arrays of scores(final on therapy scores from drug and placebo) and want to use the normal scores procdeure to test for significance. (observations are unequal in number - due to dropouts). Could you please help me out with the coding or let me know if there is a package that can be used (for example,

Hi Thanks for a very nice piece of software. I have a tiny suggestion for upslog.c. It adds a signal handler for the USR1 signal, which simply triggers it to abort the current sleep and log the requested now. I find it useful because my upssched-cmd can then killall -USR1 upslog to ensure that there is an entry in the log for brief power dropouts which would otherwise be unrecorded (because

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emikell at rimmkaufman.com --

Hi Andy, On Dec 8, 2014, at 7:07 PM, Andy Juniper <ajuniper at freeuk.com> wrote: > It adds a signal handler for the USR1 signal, which simply triggers it to abort the current sleep and log the requested now. I will admit that I'm not an expert on how signals interrupt system calls on all systems, but for what it's worth, the code looks good to me. Is it correct to assume that

Hi, I'm not sure whether this is possible but if it is, I'm sure someone on here might know ... Is it possible to use Monitor() to record a conversation[1], but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on for a long time? We already have this happening if the callers press a specific key sequence (which we've defined in features.conf)