similar to: cmd Backgound problem with option m

** I am using the following way to get p-values from fiser exact test. However, I do need to print for each pair the values "n00, n01, n10, n11". How can I print that as a table and not a matrix as below along with the p-value? Any help will be greatly appreciated fish <- function(y, x) {n00 = sum((1-x)*(1-y)); n01 = sum((1-x)*y); n10 = sum(x*(1-y)); n11 = sum(x*y); a =

Hi, I want to set a make a contingency table which will look like this.. The problem is that I can't  set the table like the following.                      col1     col2       Total     row1              a         b         n10                      rp1       rp2        100                      cp1       cp2       rtp1   ---------------------------------------------     row2            

I couldn't find this is in the FAQs or in the R archives and I've poked around the lattice package manual, but... I've been trying to figure out how to generate graphics files with a transparent background with the lattice package. The code to generate the levelplot is fine and I can produce a simple png file with a transparent background just fine, but I'm not sure how to

Hi, I'm having trouble understanding how to construct a random number generator for a parametric bootstrap. My aim is to bootstrap a Likelihood Ratio statistic (under the null) for a linear model. The function at this point is given by boot.test.n01 <- function(data, indeces, maxit=20) { y1 <- fit1+se(e2)*rnorm(314) mod1 <- glm(y1 ~ X1-1, maxit=maxit) y2 <-

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Hello People, I'm newbie in * 1.0.1, running a Linux 2.6.7 in a Debian Sarge, and this is my situation: +------------+ +-------------+ | Sip Server |-------------|CISCO PSTN GW| +------------+ +-------------+ \ || \ || \ +----------+ || | Asterisk |=========

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

If I have a global variable in my dialplan and I change it, does that change immediately take affect for all calls that are active? Here is my situation. The company I work for has two office groups that share a building. The two offices are separate companies but support one another and want to be able to transfer calls as if they were all on the same phone system. Each company has 4

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hi , i worked with h324m gateway for 3g video calling .It? configured successfully . my code in extensions.conf is [from-zaptel] exten => _X.,1,h324m_gw(0 at mainmenu) exten=>_X.,n,Hangup [mainmenu] exten => 0,1,h324m_gw_answer() exten => 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error --

Before I open a bug I'll ask again if anyone else is having trouble with receiving MWI on SIP devices in 1.4. My configuration was working fine in 1.2 but as soon as I change to any build of 1.4 I don't get notification on any of several SIP devices. I can post my configuration but since it was working I can only assume it would break if something in voicemail.conf has changed or

Yes it sounds silly, but its true. I made a menu with some submenu with the simple menu system. After pressing 23 times return to get into the submenu or back to the mainmenu it freeze. I think this is a bug of syslinux. I tried it on 3 different PC/Servers always the same effect. Here is my default and one of my submenu files: default: #Projekt PXE-Boot von Franz Lang www.franz-lang.com

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the tones to be recognized during the background( ) the playback and background files play, but asterisk doesn't do anything when I start pushing keys - I've tried it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice]

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack