similar to: selection columns by type, ie, numeric

> dt1[ vapply(dt1, FUN=is.numeric, FUN.VALUE=NA) ] a c 1 1 1.1 2 2 1.0 ... 10 10 0.2 Bill Dunlap TIBCO Software wdunlap tibco.com On Fri, Apr 29, 2016 at 9:19 AM, Carl Sutton via R-help < r-help at r-project.org> wrote: > Good morning RGuru's > I have a data frame of 575 columns. I want to extract only those columns > that are numeric(double) or integer to do

Good morning RGuru's I have a data frame of 575 columns.? I want to extract only those columns that are numeric(double) or integer to do some machine learning with.? I have searched the web for a couple of days (off and on) and have not found anything that shows how to do this.?? Lots of ways to extract rows, but not columns.? I have attempted to use "(x == y)" indices extraction

Since the aggregate S3 method for class formula already has got na.action = na.omit, ## S3 method for class 'formula' aggregate(formula, data, FUN, ..., subset, na.action = na.omit) I think that to deal with NA's, it is enough: aggregate(Value~ID, dta, max) Moreover, passing na.rm = FALSE/TRUE is "don't care": aggregate(Value~ID, dta, max, na.rm=FALSE)

Dear All, I like to simulate a physiologically based pharmacokinetics model using R but am having a problem with the daspk routine. The same problem has been implemented in Berkeley madonna and Winbugs so that I know that it is working. However, with daspk it is not, and the numbers are everywhere! Please see the following and let me know if I am missing something... Thanks a lot in advance,

I have a file server with two shares accessible to 2 different groups. After the last update ( from debian 2:3.2.5-4 to 2:3.3.2-1 ) i cannot any more access ONLY ONE of the two shares and I can't understand the reason! Can anyone hel me? I'm getting mad! Thanks Giorgio from smb.conf: [documenti_movi] path = /home/documenti_movi valid users = @staffmovi read

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Hi, We are a company developing a software mainly in C++. We want to integrate R inside our software in order to use mainly the engine (usual stats as mean, sigma, Pearson, outlier detection, CPA, multivariate, ...) and probably later the chart solution. Of course we don''t want to temporary write the data to some csv files and then do an ugly system() call :

Hi, We are a company developing a software mainly in C++. We want to integrate R inside our software in order to use mainly the engine (usual stats as mean, sigma, Pearson, outlier detection, CPA, multivariate, ...) and probably later the chart solution. Of course we don''t want to temporary write the data to some csv files and then do an ugly system() call :

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site),

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

Hi Giorgio, here is another description how to compile LLVM on Windows: http://wiki.dlang.org/Building_and_hacking_LDC_on_Windows_using_MSVC Maybe this is helpful. I created this for Windows 7 but I also repeated it successfully on Windows 8. Regards Kai On 22.07.2013 22:51, Giorgio Franceschetti wrote: > Hi all, > yes, I do not know python and I installed it only for being able to

Open the console with verbose turned up. Make a test call and see where it is hanging. That will isolate the problem. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo > Sent: Thursday, April 27, 2006 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi Christian, may be the issue is related to this other post: http://dovecot.org/pipermail/dovecot/2015-September/102094.html Anyway, what I can tell you is that for every? user account,? for every folders created after the last doveadm fts rescan for that user, dovecot does not invoke indexer. A new rescan fix the problem for all the folders created up to that moment. I was wondering if I was

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi, just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same zaptel/libpri/mISDN/add-ons. It crashes when transferring a call. Anybody tried it with success? Thank you Giorgio