similar to: AIX: Urgent!

Displaying 20 results from an estimated 3000 matches similar to: "AIX: Urgent!"

2013 Jan 22
4
Simple use of dcast (reshape2 package)
Suppose I have a small dataframe > aa Target Eaten ID 50 TPP 0 1 51 TPP 1 2 52 TPP 3 3 53 TPP 1 4 54 TPP 2 5 50.1 GPA 9 1 51.1 GPA 11 2 52.1 GPA 8 3 53.1 GPA 8 4 54.1 GPA 10 5 And I want to reshape it into ID TPP GPA 1 1 0 9 2 2 1 11 3 3 3 8 4 4 1 8 5 5 2 10 I realise that
2008 Aug 04
0
JACK / WaveInGetPosition: This function is not supported.
Hello, I'm running wine-1.1.2 on Ubuntu 8.04.1 with JACKD 0.109.2. I'm trying to run TPP, a small application that makes it possible to run Winamp DSP plugins without using Winamp. It takes the input from one sound device, does some processing on it and outputs it to another sound device. When i try to run this in Wine (with JACK), i get a popup that says: WaveInGetPosition: This
2015 Jan 27
2
[LLVMdev] Create a call to function malloc using LLVM API
Hi, I encountered an issue when attempting to create a call to function malloc. I just want to do a simple thing, suppose there is a variable p, if p is a pointer then allocate memory to p. Source code: int *p; p = (int *) malloc(sizeof(*p)); Try to generate LLVM IR for it: Type *tp = p->getType(); AllocaInst* arg_alloc = builder.CreateAlloca(tp);//builder is IRBuilder
2010 Aug 18
1
Plotting K-means clustering results on an MDS
Hello All, I'm having some trouble figuring out what the clearest way to plot my k-means clustering result on an my existing MDS. First I performed MDS on my distance matrix (note: I performed k-means on the MDS coordinates because applying a euclidean distance measure to my raw data would have been inappropriate) canto.MDS<-cmdscale(canto) I then figured out what would be my optimum
2017 Mar 12
2
WebRTC - Transport Issues.
Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username:
2006 Mar 14
1
groups issue with openssh (all versions since at least 3.8), AIX 5.3 and NIS
Hello We are have a massive performance issue in our environment since a while. SSH logins simply take 30 s to 1 minute to give a prompt, telnet are instantaneous. After doing a few tcpdump and comparisons between telnet and ssh connections, we noticed that in average a ssh connection is generating over 12000 nis sessions, scanning basically all the group.byname table a few times and we got a
2012 Jan 14
1
Error: unexpected '<' in "<" when modifying existing functions
Hi. I am trying to modify kmeans function. It seems that is failing something obvious with the workspace. I am a newbie and here is my code: myk = function (x, centers, iter.max = 10, nstart = 1, algorithm = c("Hartigan-Wong", + "Lloyd", "Forgy", "MacQueen")) + { + do_one <- function(nmeth) { + Z <- switch(nmeth, { + Z
2014 Jun 11
2
WSS over Asterisk
Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =
2015 Jan 14
1
WSS Socket Configuration
Hi Alexey, This is what works for me: [http.conf]: tlsenable=yes ; enable tls - default no. tlsbindaddr=144.x.y.z:8089 ; address and port to bind to - default is bindaddr and port 8089. tlscertfile=/etc/asterisk/keys/mycert.pem ; path to the certificate file (*.pem) only. tlsprivatekey=/etc/asterisk/keys/mycert.pem ; path to private key file (*.pem) only. Date: Tue, 13 Jan
2013 Sep 12
0
SIP over WSS connection : mask error
Hi, I use chrome and sipml5 to connect to asterisk webrtc interface using TLS. The wss connection seems ok and the SIP REGISTER sent from chrome to asterisk and the SIP response received. In the response, I get a "failed: A server must not mask any frames that it sends to the client" error msg and chrome terminates the ws connection. I've checked the asterisk debug logs, and the
2016 Feb 18
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from the wiki [1]. > So I'm using [2] live demo from
2008 Nov 23
2
Latin Hypercube with condition sum = 1
Hi I want to du a sensitivity analysis using Latin Hypercubes. But my parameters have to fulfill two conditions: 1) ranging from 0 to 1 2) have to sum up to 1 So far I am using the lhs package and am doing the following: library(lhs) ws <- improvedLHS(1000, 7) wsSums <- rowSums(ws) wss <- ws / wsSums but I think I can't do that, as after the normalization > min(wss) [1]
2015 Sep 15
3
Asterisk 13 WebRTC Status report
hi, i'm fighting with webrtc for 14 days reporting my experience to minimize number of crazy asterisk users i have working webrtc with simpl5 + asterisk 13 + pjproject 2.4.5 + chan_pjsip + secure websockets + secure audio + audio in both ways problems first, i needed run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from
2003 Apr 10
1
sshd and pam , conversation
I have setup openssh with hostbased authentication on linux (redhat). I want to allow/deny users based on a listfile, so i have a PAM module that does that, and it runs in the "account" section (oposed to pam_listfile.so, that uses the "auth" section - it wouldt work because with hostbased authentication openssh ignores the "auth" section). It's working
2009 Sep 17
4
Optimised ARM Ogg/Theora/Vorbis decoder
This is a note to announce the availability of "Ogg Theorarm", an optimised ARM implementation of decoding libraries for the Theora video code, and Vorbis audio codec from xiph.org. Full details of this code release can be found at <http://www.wss.co.uk/pinknoise/theorarm>, but highlights include: * Full speed playback of a 320x240x25fps clip with a 48kHz stereo audio track on
2012 Jun 27
1
Error: figure margins too large
Hello, I am running cluster analysis, and am attempting to create a graph of my clusters. I keep on getting an error that says that my figure margins are too large. d <- file.choose() d <- read.csv(d,header=TRUE) mydataS <- scale(d, center = TRUE, scale=TRUE) #Converts mydataS from a matrix to a data frame mydataS2 <- as.data.frame(mydataS) #removes "coden"
2016 Feb 18
2
Asterisk 13 and WebRTC. Is wiki page still valid ?
2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on my own private >> Asterisk server and access this demo
2006 Oct 24
1
RE: [Patch] Add hardware CR8 acceleration for TPRaccessing
Thanks for your advice. I will re-organize the patch. Thanks -- Dexuan -----Original Message----- From: Li, Xin B Sent: 2006年10月24日 18:08 To: Petersson, Mats; Cui, Dexuan; Betak, Travis Cc: xen-devel@lists.xensource.com Subject: RE: [Xen-devel] [Patch] Add hardware CR8 acceleration for TPRaccessing >> From: xen-devel-bounces@lists.xensource.com >>
2015 Jan 13
0
WSS Socket Configuration
Hi, I have a working WebRTC/SipJS+Asterisk(13.0.1) setup using ws sockets. Now I wanted to switch to wss to have encryption, but cannot find the required configuration parameters. Does Asterisk support wss sockets? How can I configure it? Thanks, Alexej -------------- next part -------------- An HTML attachment was scrubbed... URL:
2013 Aug 12
0
Asterisk WebRTC Support : WSS connection setup fails with error:00000000
Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate