search for: mylee

I''m attempting to use caches_page to speed up the loading of certain pages on my site, and have run into a couple questions: 1) The site has ''states'' (as in U.S. states), so I have a State model and a States controller. Going to http://sitename/states/ routes you to the States controller, ''index'' action. The States controller looks like so: class

I am having trouble understanding how the 'by' function works. Using this bit of code: i <- data.frame(x=c(1,2,3), y=c(0,0,0), B=c("red","blue","blue")) j <- data.frame(x=c(1,2,3), y=c(1,1,1), B=c('red','blue','green')) plot(0, 0, type="n", xlim=c(0,4), ylim=c(0,1)) by(i, i$B, function(s){ points(s$x, s$y, col=s$B) })

I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far.

The answer was (thanks to Mark Leeds) to do with the use of a factor instead of a vector. on [2017-08-05] at 08:57 Myles English writes: > I am having trouble understanding how the 'by' function works. Using > this bit of code: > > i <- data.frame(x=c(1,2,3), y=c(0,0,0), B=c("red","blue","blue")) > j <- data.frame(x=c(1,2,3),

We have added Asterisk to a line of 'mission critical' servers at our business, and being in the web application development business one of the core things we do is to monitor web server availability. I'd like to add Asterisk to the servers that our monitoring systems are handling, and also that our SIP trunk provider has our Asterisk system correctly registered at all times.

I''m following the Depot tutorial in the Rails book, and have just hit a wall. As instructed by the tutorial, I added the following code to the Product model: protected def validate errors.add(:price, "should be positive") unless price > 0.0 end When I attempt to add a new product, I''m getting an error: undefined method `>'' for nil:NilClass

I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: "anonymous" "" So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was

I have a new Asterisk system going into production next week and I'm a bit stumped as to the best way to handle the Dialplans for it. The Asterisk system is replacing 4 separate PSTN lines with both SIP & PSTN inputs. The setting up of the dial plan is giving me some design headaches, which probably means I'm missing something obvious and doing this the hard way. I have separate

I am not finding anything relating to this on Google, so I thought I'd pose the question here... I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a custom built PHP5.2.10 install to interconnect with our Firebird SQL database, which I've done. But I noticed that the default install path for PHP5 on this box appears to be /usr/local/bin/php rather than the path

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

I am running Asterisk 1.4.2 and recently we changed the SIP provider of our main incoming DID number. The new provider prefixes all CallerID records with a +1 in front of the number, whereas the previous SIP provider did not. Consequently now all my blacklisted numbers aren't matching in my Dialplan, so I'm getting tele-spammed. Is there a way that I can work with the blacklist

It won't be long until big labels attempt to eliminate the digital ins and outs of equipment. The problem they face is most people that currently use a rack system featuring digital interconnects will NEVER revert to an analog only system. I know that I won't! Think back to VHS, BetaMax, SVHS and LaserDisc. VHS has incredible market share because its licensing is open. Betamax was

Where are the archives of r-help kept, please? Jonathan -- Dr. Jonathan Myles e-mail:jonathan.myles at mrc-bsu.cam.ac.uk MRC Biostatistics Unit Tel. 01223 330371 Institute of Public Health FAX 01223 330388 University Forvie Site Robinson Way CAMBRIDGE CB2 2SR -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list --

Is there any way of getting just the LaTex output which R CMD Rd2dvi generates on the way to produces a .dvi file, which could then be incoorporated into another document? Thanks, Jonathan -- Dr. Jonathan Myles e-mail:jonathan.myles at mrc-bsu.cam.ac.uk MRC Biostatistics Unit Tel. 01223 330372 Institute of Public Health FAX 01223 330388 University Forvie Site

I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct

I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming & FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its

I'm pretty new to this whole Asterisk system & VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home & home office. My wife & I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime

merijnv@sci.kun.nl pondered the following.... Subject: [OT] Re: [vorbis] Format comparison >Eh, just a (seemingly offtopic) question, but how do you create a 4 channel >lp ? With just one needle ? >-- Merijn In a nutshell. Normal LP has two halves in the groove - the L and R channel. A Quad channel LP still has to 'halves' in the groove. Each channel is actually BOTH the

Is there a way to "unplug" the windows graphics from R, so that it will work across a teletype telnet session or in a Linux box proving resistant to having windows added to it? Thanks Jonathan Myles -- Dr. Jonathan Myles e-mail:jonathan.myles at mrc-bsu.cam.ac.uk MRC Biostatistics Unit Tel. 01223 330371 Institute of Public Health FAX 01223 330388 University Forvie