similar to: Help with interpolation

Dear All, I am facing problem with NA treatment in my financial time series data. # data reading aluminum = read.csv(file="alu.csv", header=T, sep=",") fut = aluminum [,2] spt = aluminum [,3] # Missing Value Treatment (Linear Interpolation) spt = interpNA(spt, method = c("linear")) fut = interpNA(fut, method = c("linear")) fut=fut[,1] spt =spt[,1]

Hi, I read many articles about how to increase Xen file-space but I need to decrease the file size. Any suggestion? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos-virt/attachments/20100405/225b0ac9/attachment-0006.html>

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

But is there a better way of getting the loops in a function than using LoopInfo? For some tasks it is necessary that we don't process the same loop twice. Doing that with LoopInfo requires the use of a flag. On Fri, Apr 23, 2010 at 12:41 PM, Devang Patel <devang.patel at gmail.com>wrote: > On Thu, Apr 22, 2010 at 11:50 PM, Arjun <arjunsingri at gmail.com> wrote: > >>

Hey all What VoIP networking monitoring, asterisk monitoring tools would you recommend? I started working with an IT company that insists on using DSL with a Sonicwall router. The problem is that the clients are having sound problems. The owner is convinced that it's the Asterisk box. In the 4 yrs I have been doing this I have not had this bad a sound problem and it always came down to a bad

All, I've updated the open-vm-tools packages that have been floating around to the latest release - 4/25/2010 - for those that are interested. There is quite a bit of new additions/changes over the past year such as the addition of a fuse driver for vmblock devices, vmxnet3, pvscsi, etc. SRPMS and .spec's available here: http://www.davehollis.com/packages/open-vm-tools-kmod.spec

Here are some notes/observations I've done on my setup that works for me. Hopefully it will be added to some r-forge documentation/wiki. I use: Windows Vista Business SP2 32bit TortoiseSVN 1.6.7 (Build 18415 - 32 Bit , 2010/01/22 17:55:06) PuTTY v0.60 (with Pageant v0.60) I have a deprecated r-forge project ('r-oo') that I will use through out. My r-forge username is

Hello, Chris > 2010/3/20 Chris Lattner <clattner at apple.com> > > >> On Mar 19, 2010, at 2:40 PM, Louis Gerbarg wrote: >> >> >>> What do you mean by "variable sized pointers"? What does: >>> >>> struct S {void *X; }; >>> >>> return for sizeof(struct S); ? >>> >> >> It doesn't, at

Gents, first of all thanks for tinc, and thanks in advance for your advice. At the risk of revealing my stupidity and opening myself to ridicule.... I am trying to connect a new remote lan into an existing tinc vpn with the central tinc vpn server located at 10.57.132.1 on the 10.57.132.0/24 subnet. I set up this central tinc vpn server myself along with several other remote lans linked to

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

On Fri, Mar 19, 2010 at 3:47 PM, Chris Lattner <clattner at apple.com> wrote: > > On Mar 19, 2010, at 11:08 AM, 琬菁楊 wrote: > > > Hello Tristan and all, > > I have already know that if I want to do this feature(c -> EFI Byte code) > for GCC > I should further modify the GCC front end(parser) to solve the problem (the > size > of pointer is determined at

Hi, I'm trying to create a GEP (inbounds) instruction . I was looking at the createinboundsGEP method . I didnt get the inputiterator part. My question how do i create a GEP instruction and what do i pass for the input iterator arguments. Thanks, Rohith. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Apr 3, 2010, at 5:36 AM, Russell Wallace wrote: > On Fri, Apr 2, 2010 at 6:17 PM, Chris Lattner <clattner at apple.com> wrote: >> No, please don't. This is something we specifically do not want to support. The issue is not the parser, the issue is that struct field offsets are no longer constant in this model. > > What about declaring that pointers are always 64

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

On Sun, Apr 25, 2010 at 5:15 PM, Rohith Goparaju <rgoparaj at umail.iu.edu> wrote: >  Hi, > >    I'm trying to create a GEP (inbounds) instruction . I was looking at the > createinboundsGEP method . I didnt get the inputiterator part. My question > how do i create a GEP instruction and what do i pass for the input iterator > arguments. Generic answer to "how do I

Hi all, Following the example from http://llvm.org/docs/GoldPlugin.html#example1 gives the following error on Ubuntu Karmic: /usr/bin/ld: error: a.a: no archive symbol table (run ranlib) /usr/bin/ld: /usr/lib/crt1.o:../sysdeps/i386/elf/start.S:115: error: undefined reference to 'main' /usr/bin/ld: b.o: in function foo1:b.c(.text+0x4): error: undefined reference to 'foo2'

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Hey, There's one relevant patch in the F13 DDX package that hasn't been pushed into git still. The patch simply switches off the bicubic filtering if we're scaling the image by less than 200% in either direction in order to prevent some excessive blurriness in the image (very noticable with the gstreamer xvideo test). The patch has been in the Fedora package since F11 as a user