similar to: Creating a Double Bar Graph With Provided DataSet

Dear Sir / Madam I had been using Glusterfs of both nodes, and it is under in distribute mode. But I cannot use all of the gluster commands at one of the node recently, any gluster command not respond (E.g volume info or pool list), and I got the error in cli.log . Does anyone can help me, many thanks. Here is the log error: [cli.c:759:main] 0-cli: Started running gluster with version

Hearing no opinions to the contrary, let's say Banger's 6pm (tomorrow). Address and web site below. I submitted online for a reservation early this morning, and I'll try to get an updated head count on Sunday morning. -Hal Sent from my Verizon Wireless 4G LTE DROID On Nov 14, 2015 11:21 AM, Chandler Carruth <chandlerc at google.com> wrote: Did this ever get settled? On Fri,

----- Original Message ----- > From: "Sebastian Pop" <spop at codeaurora.org> > To: "James H Cownie" <james.h.cownie at intel.com> > Cc: "Hal Finkel" <hfinkel at anl.gov>, "John Leidel (jleidel)" <jleidel at micron.com>, "LLVM Dev" > <llvm-dev at lists.llvm.org>, "cfe-dev" <cfe-dev at

Hi all, I''m a software developer from New Zealand. I''ve built a couple of rails sites and have some ajax experience. I''d like to develop on a rails site with the aim of making a business - is anyone else looking to do the same? The two rails sites I built recently are foopad.com and friendr.com. Regards, Ben

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Winbind appears to lose connections every once in a while. Is there a way to tell which Domain Controller winbind is making connections to? I have 3 DCs currently and trying to track down the problem. Thanks, Toan

Appologies for cross-posting Dear R users, I am using R2WinBUGS to call WinBUGS from R. After loading data, model, and initial values I call this command res <- bugs(data = dfile, inits = list(ifile), parameters.to.save = c("beta"), model.file = mfile, working.directory = tdir, n.thin = nthin, n.chains = 1, n.iter = niterations*nthin, n.burnin = 0, DIC = F, debug = T,

Hey All, AT&T is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have

Hi, On Wed, Nov 11, 2015 at 8:01 AM, Cownie, James H via cfe-dev <cfe-dev at lists.llvm.org> wrote: > It would be much more convenient for those of us who are in town for SC and staying downtown if we could find somewhere in walking range. (Which in my mind is 1 to 3/2 of a mile; others may have different criteria, but I find a 30 min walk refreshing after taking beer :-). > Agreed.

Hi there, I am reading the 2004 paper "Smoothing with mixed model software" in Journal of Statistical Software, by Ngo and Wand. I tried to run their first example in Section 2.1 using R but I had some problems. Here is the code: library(nlme) fossil <- read.table("fossil.dat",header=T) x <- fossil$age y <- 100000*fossil$strontium.ratio knots <-

Hi everyone, There seems to be a good level of interest in this; so we should settle on a time and location. It has been pointed out that previous socials were held at http://bbrovers.com/ - we seem to have a preference for the availability of alcoholic beverages, and otherwise no strong opinions have been expressed. Logistically, we should pick someplace likely to have flexibility regarding

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in

Hello again, Because the LLVM in HPC workshop will be in Austin on Nov. 15th (http://llvm-hpc2-workshop.github.io/), we'll have an anomalously-high density of LLVM developers in Austin that day. I think it would be a great evening to have an LLVM social! I'm not familair with the Austin area, but I've cc'd some folks who are (or at least were) in Austin, so hopefully we can get

Another day, another game. I mounted the image of a game called "Memento Mori", a german adventure game. The installer was not exactly working perfect ("please enter (...) in .ini or something" where the name of the game, the installation folder etc. normally stands), but it worked. Even the game seemed to start at first, but after a few seconds it crashes. The error message

Tried your fixed_generic.h change but that didn't help. Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Sent: Wednesday, December 12, 2007 4:13:35 PM Subject: Re: [Speex-dev] 4kbps sounds robotic on TMS320C64 Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in

Hi, sorry for the repost again, but does anyone have a code snippet example of how to use the jitter buffer? Regards, Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: speex-dev@xiph.org Sent: Wednesday, January 10, 2007 8:09:10 PM Subject: [Speex-dev] Sample snippet of how to use jitter buffer Hi, I searched around in the Speex manual and API but

Dear Speex Developers, I'm trying to use Speex on a development board that has the Samsung 2140 ARM Processor (model S3C2410X01, ARM 920T Core). I enabled the arm assembly, i.e. build the speex library using the following configure command: ./configure --enable-fixed-point --enable-arm-asm --enable-fixed-point-debug --host=arm-linux After linking the library with my application, I ran

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time