similar to: nlme formula from model specification

Hi devs :-) At the Mozilla Festival in London recently, I took part in a Hack the DJ workshop, looking at ways to take digital DJ'ing to the next level: http://www.bbc.co.uk/blogs/researchanddevelopment/2011/11/setting-a-dj-challenge-at-the.shtml One of the ideas proposed was stem mixing, using multichannel files in DJ applications. A proprietary implementation of this idea is Fireplayer

Hello, I'm wondering how I can merge two featuresets into one. My dataset is two sets of microarray data and it looks like followings: > rawData $v1 TilingFeatureSet (storageMode: lockedEnvironment) assayData: 2197815 features, 59 samples element names: channel1, channel2 protocolData rowNames: LT290677RU_D1_2011-02-16 LT286300LU_D1_2010-07-24 ... LT003990RU_D1_2010-11-04 (59

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Everythings works great with asterisk exept one feature with redirect : it doesn't redirect when ringing ... BTW are their any plans to extend the manager API ?? Michael Devenijn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031212/1200012e/attachment.htm

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

what i did in this situation was dump the stats/status file to the local server through a lynx --dump call from php .. and then i also update a sql table to reflect the changes so i can poll that instead .. a. <p>On Thu, 26 Feb 2004, MacSym wrote: > > Hi everybody, > > I am wondering if it is possible to generate the XML/XSL files on a remote > server (status.xsl,

I am working on a script that takes numeric performance indicators and runs them against a series of regressors (dummy regressors, yes\no stuff via 0 and 1, e.g. Was is Christmas this week 0=no, 1=yes). The script is as follows (Written as a function): -- Begin Script -- doEnv <- function(HOUR,ENVNAME,REPORTNAME) { library(RODBC) library(forecast) library("geneplotter")

Hi all, what's your opinion about CDR-Event (like Hangup or Ring etc.) on AstManager ? Or, is something like this already implemented ? Regards, Thomas

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hi List, I used to access a Paradox database using RODBC, but since last week I am not able anymore to fetch any table which has a name longer than 11 or 12 characters. Here is the the pattern of my code, nothing spectacular: library(RODBC) channel2<-odbcDriverConnect('DSN=xxx') #table names with up to 11 characters still work sqlFetch(channel2, 'abcdefghijk') #table names

Hello all. I have two small questions in one post, for the sake of brevity. 1. I have some objects that I want to delete. I have the line: rm (c (channelheader, paste ("channel", 1:3, sep=""))) I have tried a few variations, including list=, but cannot figure it out. In SAS, I can use a ':' as a wildcard. Is there any equivalent in R? 2. Is there any possible was to

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

I imported a spreadsheet into a variable sh e.g. sh$aaaa, sh$bbbb, etc... doing the following: tsSource <- ts(paste("sh$",NAMEVARIABLE,sep="") ... ) fails. The paste isn't evaluating properly. What is the proper way to concatenate a data source with a member name such that they evaluate properly. actual code below: doEnv <-

Hi All, I'm using R 2.11.1 on 64 bit windows XP. The little function I wrote below I use often to import the first 1001 lines in an excel sheet to R. This works fine on the 32 bit version of R but fails on the 64 bit [both on the same machine, using the same function, importing the same .xls file]. The message from 64 bit R is: Error in sqlTables(channel1) : first argument is not an

Dear List, I have a problem with RODBC on a Paradox-DB, sqlQuery, and special characters in table names. Unfortunately, some of the latter include the underscore <_>. And I am not able to change them. That's not a problem, when I quote the table name: > sqlQuery(channel2, 'SELECT * FROM "anmeldung-alt"') Btw, if I swap ' and " it does not work: >

Hi all, I am very new to R and I'm trying to import data from Microsoft Access. So far, I've managed to do so successfully using the following code: testdb <- file.path("c:\Databse.accdb") channel2 <- odbcConnectAccess2007(testdb) data.table <- sqlFetch(channel2,"data") This successfully imports a table(?) called "data.table". But when I try to

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/9999-00000029 Channel2:

Hi all, I'm going to buy E1 digium110P ,any one knows how i can get faxtomail working for three different channels? I mean: channel1-->FaxtoMAil1@XXX.com channel2-->FaxtoMail2@xxx.com?? for 1 channel is not dificult with*@home and NVfaxdetect Using *@home 2.5 faxToPDFmail works, but i always get error opening pdf in outlook. i've solved this with a mimeconstruct update explained

I'm looking for ideas here. This probably will wind up being more of an alsa problem, but I thought the people on this list might have been in similar circumstances and have ideas for me. I work at a radio station with both a webcast and FM broadcast. We want to use icecast both for our webcast and as a backup studio to transmitter link. However, the webcast audio needs to be