search for: dropout

Dear all I have a function in which I would like to change some values to NA according to some condition. dropout<-function(y, nahr=FALSE,...) { <some stuff for computing an index> if (nahr) y[index]<<-NA invisible(index) } in case y is a vector all works OK but if it is a part of data frame by calling dropout(df$y) or dropout(df[,number]) no change is done. Please can you help me what is...

...g. I know from the R help page that there is some controversy about how exactly to calculate the test statistic, but that's not what is causing the problem, as I can get the same W value. Am I calculating the test statistic incorrectly? Thanks, sample code below Sam Stewart #Ordinal example dropouts = c(rep(0,25),rep(3,10),rep(2,9),rep(1,13),rep(4,6)) remain = c(rep(0,31),rep(3,2),rep(2,6),rep(1,21),rep(4,3)) tab2 = rbind(table(dropouts),table(remain)) ordTest = wilcox.test(x=dropouts,y=remain,correct=FALSE,exact=FALSE) cumsum(colSums(tab2)) W = max(c(sum(rank(cbind(dropouts,remain))[1:length...

Dear R users,   Please assist me with the following problem. I have a dataset that looks like the following:   dat<-data.frame(   'id'=rep(c(1,2,3),each=3),   'time'=rep(c(1,2,3),3),   'y'= c(2,2,NA,2,NA,NA,2,5,7) )   I wish to create a variable for dropout time in dataframe 'dat' such that the dropout time is the time to drop out by the subject as follows:     dat<-data.frame(   'id'=rep(c(1,2,3),each=3),   'time'=rep(c(1,2,3),3),   'y'= c(2,2,NA,2,NA,NA,2,5,7),   'dropout time'=c(2,2,2,1,1,1,3,3,3) ) Any...

Hello, I'm having a problem with significant audio dropouts occurring in voicemail messages left via an ISDN-BRI trunk. Dropout durations are as short as 15ms and as long as 200-300ms. The audio that is recorded, appears to be otherwise complete, just with frequent holes punched in it. The same trunk has no problems with audio files played toward it...

...ave about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a highly- specced HP xeon server. This in turn connects to an Epygi gateway ( http://www.epygi.com/quadro-gateway/70/#isdn ), bringing in 4 ISDN BRI lines as a SIP trunk. The issues: Dropouts - by far the most serious issue we've encountered. On most calls (normally anything longer than 1 or 2 minutes), suddenly one end of the call will go silent and not be able to hear the other person. After a few seconds of "I can't hear you!" the audio returns and continue...

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 trouble?...

hi all I keep getting 'bzzt'-sounding dropouts, but I've only heard them when from behind NAT. I don't think it's related to bandwidth, as I'm on a 2400/640 ADSL line. I'm currently using ALAW. Any ideas how to debug this? SIP DEBUG doesn't show me anything when dropouts happen, although I'd doubted it'd do...

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): http://personal.inet.fi/musiikki/nebularia/test/t2title.ogg Decoded back from Ogg Vorbis (~6.5 MB): http://personal.inet.fi/musiikki/ne...

Hi, I need some help again. :-) I have "Too much recursion error" prototype.js line 1288, I heve this error when I try to use Effect.dropout (script.aculo.us) if I remove table ID, there is no error, and the effect works fine (remove the tr from table) But I need this ID assign to this table, as table ID is used by other effect (Builder.node) Please help .... Thanks YUAN Here is the table : <table width="75%" border=...

...oncepts.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 data dropouts at various scales (a phenomenon not present at data rates of 160 Hz and below): 1. Approximately every 20ms, a few samples are dropped (believed to be due to internal buffer wrapping). 2. Approximately every 200ms, about 50 samples are dropped (believed to be due to flash write times). 3. At se...

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f", "http://stream.lrz-muenchen.de:31337/m945-lq.mp3")...

...et 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 using a TDM400P interface). Cheers, Brian

hi I keep getting sound dropouts during SIP re-registration, and I can't find a remedy for it. I use SIP friends from MySQL. Below is SIP debug output for the re-registration Thanks in advance roy ------- *CLI> sip debug ip 80.202.161.221 SIP Debugging Enabled for IP: 80.202.161.221 *CLI> Sip read: REGISTER sip...

Is it possible to run a packet capture to see what happens over the wire when this occurs? That would give some concrete information. If you have the disk space, just rotate out a capture file every hour. Once you catch it, filter out a pcap file with what happens when the sources are disconnected, and send us a link. Brad Isbell brad at musatcha.com http://www.musatcha.com On Tue, Nov 4, 2014

We are having some problems with the audio during a call dropping out for a second or so and then returning. This is not at the beginning of the call and seems to happen randomly. We use primarily Cisco 7960's and our * is a very current version of CVS head. The server itself is a Compaq dual PIII 800 w 1GB of RAM, it has a single T100P card with a T1 to our voice provider. We have very light

> In my case, I have good incoming quality and terrible quality going out. > That is, I can hear people perfectly well but they complain that my > voice drops out and is garbled regardless of who places the call. This suggests to me that you may have congestion problems in your "upstream" traffic flow. Setting QoS on the packets may not help, if whatever router you are using

Hello, ? I've searched and searched for a solution to this problem but continue to come up empty. ? I have a very generic setup.? CentOS Linux 7.3.1611 running on AWS.? ? I installed the QMail toaster from here: http://www.qmailtoaster.com/ ?( I had been using Bill Shupp's Qmail Toaster for years and years, but he stopped supporting it.) ? As such, I did have to convert my

...let's say I'm writing a total of 80 MB/sec from 4 Windows clients and reading 220 MB/sec from 12 other Windows clients -- every half hour or so, all I/O seems to grind to a halt for 1 or 2 seconds. You can see this graphically using a program such as "gkrellm". You just see a dropout of all reading and writing from/to the disks. SMB traffic continues to come in from the network, but SMB traffic stops going out TO the network. This same pattern has been observed on multiple servers, so the problem isn't caused by some bad RAID card or other piece of defective hardware....

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP