similar to: Legend frame error (PR#9236)

Hello, Does R have any facility to create a plot that looks similar to this: http://www.augustcouncil.com/~tgibson/barmatrix.jpg Thank you, -TAG Todd A. Gibson

Hello, I have a data manipulation problem that I can easily resolve by using perl or python to pre-process the data, but I would prefer to do it directly in R. Given, for example: month length ratio monthly1 monthly2 1 Jan 23 0.1 9 6 2 Jan 45 0.2 9 6 3 Jan 16 0.3 9 6 4 Feb 14 0.2 1 9 5 Mar 98 0.4 2 2 6 Mar 02 0.6

Hi, I''ve migrated my tc configuration from CBQ to HTB. One problem appeared. Htb seems to miscalculate the bandwidth for classes with greater rates. For rates below 2Mbit there is almost no difference between the configured and the measured rate. For large ones the problem starts. My root class has 10Mbit rate and when the interface has heavy trafic the measured bandwidth

Hello all, When I view or print the below plot on my Linux machine under R 2.0.1 I see a nice thick solid and dashed line with a legend. However, while the lines are distinguishable, the legend is not. That is, the short (solid) line next to "line1" and the short (dashed) line next to "line2" seem to have the exact same length. What I would like to do is to expand the legend

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

What are these dns queries for? I'd like to disable them but I cant find any obvious reference to them in the asterisk source. I'm running Asterisk 1.4.21.2 I call voicemail and immediately hang up: I called from a sip client called line1, but I have no idea where 08c5b9e0 is coming from... 14-Jun-2009 12:37:07.926 queries: info: client 127.0.0.1#41105: query:

Consider this: $ cat test.dat 1 a 2 b Now, we want to read the 2nd column as a factor and ignore the first (since it's just a sequential ID). We can't just put "factor" among the colClasses (would have been nice), so let's try this instead > setAs("character","factor",as.factor) Arguments in definition changed from (x) to (from) >

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

I'm having problems getting my aastra 480i to register with the asterisk server. I can inititate calls from the phone, but >sip show peers does not show any IP address registered for this phone. I am probably missing something stupidly simple. Anyone have an example config to share or corrections for my configuration? Asterisk 1.2 aastra 480i CT has the 1.4 firmware <sip.conf>

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =

I figured I''d post here before submitting a ticket, but I''m seeing some confusing stuff when dealing with YAML now. I was using it to freeze objects in my database, and so i had some data already around to mess with. I upgraded both Ruby (1.8.3 -> 1.8.4) and Rails ( -> 1.1RC1) and this junk started. I''ve outlined the problem in two pastes, which I''ll

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was relesed. now I am having troubles with my dialing plan and voice mail. As part of the upgrade I re-built the machine so there was a blank slate however after installing 0.7.1 I had no mail box creation script and could not figure out how to go about creating a mailbox, any suggestions would be usefull. I have looked at

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at

This appends a single line to a file, with some cleverness involving guessing the right line endings to use. Also adds a test. --- builder/test-virt-builder.sh | 57 +++++++++++++++++++++++++++++++++++- customize/Makefile.am | 2 ++ customize/append_line.ml | 70 ++++++++++++++++++++++++++++++++++++++++++++ customize/append_line.mli | 20 +++++++++++++ customize/customize_run.ml

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hello, I have a customized pairs () fonction as follows that displays correctely my data. ------------------------------------------------------------------------ panel.cor1 <- function (x, y, digits=2, prefix="") { usr <- par("usr"); on.exit(par(usr)) par(usr = c(0, 1, 0, 1)) r <- cor(x, y,use="pairwise.complete.obs",

I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech