similar to: Streaming 2 different sources with darkice

Dear R-community, I'm analysing some noise using the nlme-package. I'm writing in order to get my usage of lme verified. In practise, a number of samples have been processed by a machine measuring the same signal at four different channels. I want to model the noise. I have taken the noise (the signal is from position 1 to 3500, and after that there is only noise). My data looks like

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

Michael Smith wrote: > On Thu, 13 Jan 2005 13:19:08 -0500, Joel Ebel <jbebel@ncsu.edu> wrote: > >>I asked this questions before, but I think it got lost/ignored in all >>the traffic. So I'm asking it again. >> >>Why would my clients number always be 113 larger than my number of >>listeners? It was right earlier today, but when I restarted the

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

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 {

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

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")

We're running a darkice/icecast configuration to stream our LPFM radio station (and to provide a quick-n-dirty studio-transmitter-link). (See config data below.) We're getting intermittent "Buffer overrun!" errors out of darkice, and we associate these with interruptions in our audio stream that are quite noticeable. These occur anywhere from once to ten times per hour.

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.

On Thu, 13 Jan 2005 13:19:08 -0500, Joel Ebel <jbebel@ncsu.edu> wrote: > I asked this questions before, but I think it got lost/ignored in all > the traffic. So I'm asking it again. > > Why would my clients number always be 113 larger than my number of > listeners? It was right earlier today, but when I restarted the sources > it increased by 113, and it has stayed

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

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

On Fri, 5 Oct 2001, Samuel Hathaway wrote: > Hey, > > I've compiled DarkIce 0.6 dynamically linked to LAME 3.89. I'm running > Slackware 8 and using gcc 2.95.3. Running DarkIce yields the following > output: > > DarkIce 0.5 live audio streamer, http://darkice.sourceforge.net Actually, that's DarkIce 0.5's output. DarkIce 0.6 says: DarkIce 0.6 live

You helped me a few weeks ago setup DarkIce. Well I had nothing but problems doing it on RH62 so... here I am again but this time armed with RH71. I followed your instructions for installing lame and ogg vorbis and it's all fine. When I tr to install DarkIce though I get the following error. # make make all-recursive make[1]: Entering directory `/usr/src/download/darkice-0.6' Making

Samuel, I had the same problem. I think it's because the port in darkice does not match the port icecast or shoutcast is listening on. I thought it was 8000 but it was actually 8001. Check that. Matt -----Original Message----- From: owner-icecast@xiph.org [mailto:owner-icecast@xiph.org]On Behalf Of Samuel Hathaway Sent: Friday, October 05, 2001 2:43 PM To: icecast@xiph.org Cc:

On Fri, 5 Oct 2001, Matt & Stephanie Mencel wrote: > Using RH71. Got Lame, Vorbis, gcc3 and everything installed. When I try to > make DarkIce I get this error. Can anyone help? > > $ make > make all-recursive > make[1]: Entering directory `/usr/src/download/darkice-0.6' > Making all in src > make[2]: Entering directory `/usr/src/download/darkice-0.6/src'

Drew Lane wrote: > make all-recursive > make[1]: Entering directory `/usr/darkice/darkice' > Making all in src > make[2]: Entering directory `/usr/darkice/darkice/src' > c++ -DHAVE_CONFIG_H -I. -I. -I.. -O2 -pedantic -Wall -c > VorbisLibEncoder.cpp > VorbisLibEncoder.cpp: In method `bool VorbisLibEncoder::open()': > VorbisLibEncoder.cpp:157:

> > obviously you have some wierd version of vorbis on your machine. what > version do you have? I persume it's not 1.0 final... <p>Yeah, I think installing the final verison of vorbis solved a lot of problems. However, here's a new error I'm getting when trying to 'make' darkice: make all-recursive make[1]: Entering directory `/usr/darkice/darkice'

Yea, I figured it out...brain fart on my part. Got it working, but getting some sort of TCPSocket error now... still trying. :) [root@jabez etc]# darkice -c /usr/local/etc/darkice.cfg DarkIce 0.6 live audio streamer, http://darkice.sourceforge.net Copyright (c) 2000-2001, Tyrell Hungary, http://tyrell.hu Using config file: /usr/local/etc/darkice.cfg Using POSIX real-time scheduling, priority