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• Converting An Older VMWare Image to ESXi

  - by Bart Silverstrim

  I have just installed VMWare ESXi and imported one physical machine without issue and one virtual machine, from an older version of VMWare Server on a Linux system, to it without issue. But I have another machine (Windows XP), virtual running on the VMWare Server, that I can't get the Converter to work with. Converter runs, says everything is fine, but when I go to the ESXi VSphere manager and hit the button to power up that vm, the console stays black with a blinking cursor and the processor for that VM spikes 100% and stays there. An event log item in VSphere warns something about activation issues with Windows. Has anyone else run into this issue before? Is it something with the disk controller? I copied the folder with the VM directly to the storage drive on ESXi hoping to create a new machine and point the data store to that folder or at least that drive image; nope, something about not using an IDE controller (must be something with the older format). In short, converter is doing something that particular machine doesn't like, and I can't find a way to simply open that hard disk image or convert it unless someone else has seen this. I try attaching a bootable CD image for disk repair to see if it can check the hard drive but I can't seem to get the console to boot from it...either too slow on the button or it just isn't able to easily boot from a virtual drive image (.iso). Any suggestions or help?

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• Typing Japanese on Windows Vista with Dvorak

  - by Ken

  I'm using Windows Vista, and I type English with the Dvorak keyboard layout, and I want to be able to type Japanese text that way, too. I've figured out how to set it up to let me type Japanese here, but it uses QWERTY. What I've got so far is: click the "EN" in the taskbar, and select "JP" if the letter that appears in the taskbar is "A", hit alt-~ to change it to "?" type as if I was typing Romaji on a QWERTY keyboard, (e.g., left pinky home row, right ring finger top row), and hiragana appear (??) press spacebar to convert to kanji (e.g., ?), and return to accept That all works great, but it assumes I'm on QWERTY, which isn't very comfortable for me. I want everything the same, but to be able to type kana with Dvorak (e.g., left pinky home row, left ring finger home row - ??). I can do this on Mac OS, so it's not an unheard-of feature. But it was kind of an obscure setting to find, so I figure on Windows it's probably a really obscure setting. :-) But I haven't been able to find it yet. Thanks!

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• Does the Lenovo t60p vga port support an s-video signal?

  - by Matthijs Wessels

  I just bought a new television. The problem is it turns out it doesn't have a VGA port. It does have: s-video, component, hdmi and scart. My Lenovo t60p only has vga. If have search frantically for a solution and even though it seems I have sooo many options they are all dead ends. Or I keep ending up having to buy a 100 euro box to convert the signal. However, I found that some video cards support s-video through the vga port. It says look it up in your video cards documentation. I have a Lenovo t60p laptop with a ATI MOBILITY FireGl v5250. But I can't seem to get my hands on any documentation where this is supposed to be documented. I found this website: http://forum.notebookreview.com/showthread.php?t=179529&highlight=s-video There this guy says he thinks it's in the t60 but dropped in the t61, but suggests to the guy with the t60 that it won't work. I can't really conclude anything from that. Furthermore, I am not looking for the best of the best quality. So when I found this: *http://www.amazon.com/VideoSecu-Computor-Presentation-Converter-VGA2TV/dp/B000X3FAJU/ref=pd_cp_e_3_img I woudl be quite happy with this. Except that I don't think I can order it because I don't live in the US. Can anybody give me a definite answer, to whether the vga port of my lenovo t60p ati firegl v5250 supports s-video? So that I can just by a vga to s-video cable to achieve my goal.

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• Add and remove letterhead in Word document

  - by Daniel Wolf

  Our company has letterheaded paper (pre-printed paper with our logo on it). Whenever we send something out by mail, we print it on that paper. However, when we send the same document via email, we convert it to a PDF file. Now the problem is: when converting a Word document to PDF, it should contain the letterhead. When printing the same document on paper, it should not (or else the letterhead would be printed twice). Currently, we are using two different Word document templates - one with letterhead, one without. So whenever we want to add or remove the letterhead, we have to create a new document with the other template and copy and paste everything over. Nasty solution. What I'm looking for is some simple way to switch the letterhead on and off. What I've tried so far: Switching the template: There does not seem to be a simple way to switch the template for an existing document. Using a picture watermark: Our letterhead goes all the way to the border of the page. (No printer supports this, of course, but it is fine for export to PDF.) Apparently depending on the current default printer, Word will not allow a borderless watermark, instead shifting the image around. Using the page header: When editing the page header, I can insert pictures at arbitrary positions, which is great. However, I could not find a way (short of macros) to enable/disable just the pictures in the header. (The text should remain there.)

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• Can not access SQLServer database

  - by btrey

  I'm trying to convert an Access database to use a SQLServer backend. I've upsized the database and everything works on the server, but I'm unable to access it remotely. I'm running SQLServer Express 2005 on Windows Server 2003. The server is not configured as a domain controller, nor connected to a domain. The computers I'm trying to access the server from are part of a domain, but there are no local domain controllers. I'm at a remote location and the computers are configured and connected to the domain at the home office, then shipped to us. We normally log in with cached credentials and VPN into the home office when we need to access the domain. I can use Remote Desktop Connection to access the 2k3 server which is running SQLServer. If I log into the server with my username, I can bring up the database, access it via the Trusted Connection, and the database works. If I try to run the database locally, however, I get the Server Login dialog box. I can not use a Trusted Connection because my local login is to the home office domain and is not recognized by the SQLServer machine. If I try to use the username/password that is local to the SQLServer, I get a login failed error. I've tried entering the username as "username", "workgroup/username" (where "workgroup" is the name of the workgroup on the SQLServer), "sqlservername/username" and "[email protected]" where "1.2.3.4" is the IP of the SQLServer. In all cases, I get a login failed error. As I said, I can login to the server via Remote Desktop Connection with the same username and password and use the database, so permissions for the username appear to be correct for both a remote connection and for database access. Not sure where to go from here and any assistance would be appreciated.

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• .ashx cannot find type error on IIS7 , no problems on webdev server

  - by Aivan Monceller

  I am trying to make AspNetComet.zip work on IIS7 (a simple comet chat implementation) Here is a portion of my web.config. <system.web> <httpHandlers> <add verb="POST" path="DefaultChannel.ashx" type="Server.Channels.DefaultChannelHandler, Server"/> </httpHandlers> </system.web> <system.webServer> <handlers> <add name="DefaultChannelHandler" verb="POST" path="DefaultChannel.ashx" type="Server.Channels.DefaultChannelHandler, Server"/> </handlers> </system.webServer> When I publish the website on my localhost IIS7 I receive an error: POST http://localhost/DefaultChannel.ashx 500 Internal Server Error Could not load type 'Server.Channels.DefaultChannelHandler The target framework of this project is .Net 2.0 I tried the Classic and Integrated Mode application pool for .Net 2.0 with no luck. I also tried converting the project to 4.0 and tried the Classic and Integrated Mode application pool for .Net 4.0 with no luck. I also tried adding the managed handler through IIS Manager's Handler Mappings. If you have time please download the source (184kb) to reproduce the problem on your own machine. The zip contains a VS2010 solution (.Net 2.0). You could also try to convert this to .Net 4.0 I am using Windows 7 anyway if that matters. If you need more details, please drop your comments below. This is working fine by the way on my webdev server.

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• Tracking down the cause of a web service fault running in IIS

  - by BG100

  I have built a web service in Visual Studio 2008, and deployed it on IIS 7 running on Windows Server 2008 R2. It has been extensively tested, handles all errors gracefully and logs any uncaught errors to a file using log4net. The system normally runs perfectly, but occasionally (2 or 3 times a day) a fault occurs and screws up the application which needs an iisreset to get it working again. When the fault occurs I get some random errors that are caught by my catch-all error log, such as: Value cannot be null. Could not convert from type 'System.Boolean' to type 'System.DateTime'. Sequence contains no elements These errors are raised on every request until I manually do an iisreset, but there is no sight of them when the system is running normally. The web service is a stateless request-response application. Nothing is stored in the session. It could be load related, but I doubt it as there are only around 30 or 40 requests per minute. This is proving very tricky to track down. I can't work out what is putting IIS into this bad state, and why it needs an iisreset to get it working again. Nothing is reported in the event log. Can anyone suggest how to enable more extensive logging that might catch this fault? Also, does anyone know what might be causing the problem?

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• Is ffmpeg incorrectly interpreting .aif files?

  - by marue

  Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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• Disk Partitioning problem with fdisk.

  - by MA1

  Currently i am using fdisk to create/resize windows partitions. Following is a sample input script to fdisk to create/resize windows partitions: fdisk /dev/sda < partInput the contents of partInput are as follows: d #delete the partition 3 #partition number to be deleted n #add a new partition p #primary: type of new partition 3 #new partition number 18804 #start cylinder of new partition 77433 #end cylinder of new partition t #change the type of partition 3 #partition number whose type(filesystem) is to be changed 7 #HPFS/NTFS: partition type(filesystem) n #add a new partition p #primary: type of partition 77434 #first cylinder of new partition 77825 #end cylinder new partition w #write all the above changes As you see in the above input we are using cylinders for start and end. Earlier i am using sectors as unit and everything is working fine but getting problems when partitioning a 1.5TB hard drive. Then i changed the unit to cylinders but it is working on some machines not all. On some machines fdisk failed to create the partition table correctly. So, i am thinking to move to parted if there is no way to do the above using fdisk. Please also tell me how to correctly convert sectors to cylinders? How to perform all the above steps using parted without losing the data OR how to use fdisk correctly?

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• Typing Japanese on Windows Vista with Dvorak

  - by Ken

  I'm using Windows Vista, and I type English with the Dvorak keyboard layout, and I want to be able to type Japanese text that way, too. I've figured out how to set it up to let me type Japanese here, but it uses QWERTY. What I've got so far is: click the "EN" in the taskbar, and select "JP" if the letter that appears in the taskbar is "A", hit alt-~ to change it to "?" type as if I was typing Romaji on a QWERTY keyboard, (e.g., left pinky home row, right ring finger top row), and hiragana appear (??) press spacebar to convert to kanji (e.g., ?), and return to accept That all works great, but it assumes I'm on QWERTY, which isn't very comfortable for me. I want everything the same, but to be able to type kana with Dvorak (e.g., left pinky home row, left ring finger home row - ??). I can do this on Mac OS, so it's not an unheard-of feature. But it was kind of an obscure setting to find, so I figure on Windows it's probably a really obscure setting. :-) But I haven't been able to find it yet. Thanks!

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• Copy a website and preserve the file & folder structure

  - by DrStalker

  I have an old web site running on an ancient version of Oracle Portal that we need to convert to a flat-html structure. Due to damage to the server we are not able to access the administrative interface, and even if we could there is no export functionality that can work with modern software versions. It would be enough to crawl the website and have all the pages & images saved to a folder, but the file structure needs to be preserved; that is, if a page is located at http://www.oldserver.com/foo/bar/baz/mypage.html then it needs to be saved to /foo/bar/baz/mypage.html so that the various Javascript bits will continue to function. None of the web crawlers I've found have been able to do this; they all want to rename the pages (page01.html, page02.html etc) and break the folder structure. Is there any crawler out there that will recreate the site structure as it appears to a user accessing the site? It doesn't need to redo any of teh content of the pages; once rehosted the pages will all have the same names they did originally so links will continue to work.

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• How can I avoid a few seconds of blank video when using -vcodec copy?

  - by arlomedia

  I'm processing user-uploaded videos on a CentOS web server with ffmpeg. I need to convert each video to a standard size and format, then extract a 30-second sample clip from each video. I want to use the "-vcodec copy" flag in the extraction command to avoid encoding a second time. This command works for my initial conversion: ffmpeg -i uploaded.mov -f mp4 -vcodec libx264 -vpre medium -acodec libfaac -r 15 -b 360k -ab 48k -ar 22050 -s 480x320 formatted.mp4 And this sometimes works for the extraction: ffmpeg -i formatted.mp4 -vcodec copy -acodec copy -ss 0 -t 30 formatted_sample.mp4 However, when I run the extraction command on some videos, the extracted sample clip starts with several seconds of blank video. The audio starts right away but the video doesn't start for 3-6 seconds. To demonstrate the problem, I've uploaded two video clips and run the above commands on them. I created the first clip in Final Cut Express and encoded it with Handbrake before uploading to the web server: 1a) uploaded clip 1b) converted with first command 1c) extracted with second command, missing first six seconds By comparison, this second clip comes from Apple's website and does not show the problem: 2a) uploaded clip 2b) converted with first command 2c) extracted with second command, no problem Can anyone see what's different about the two source clips? And if so, is there anything I can do in my conversion command so that when the extraction command runs, the clip is set up to avoid the missing video? By the way, I initially had the problem with ffmpeg 0.6.1 installed from yum, but I upgraded to the latest git version and the problem remains.

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• avconv gets killed if mkv has subtitles

  - by Lukas Knuth

  What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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• Photoshop CS4. How do I make sure my color stays the same in my different .psd files? (could be RGB

  - by Kris

  I asked 2 photoshop experts I know but they haven't got a clue because all my settings are exactly the same, in both files. (except the RGB type !! I'm not sure. Please read on) I use RGB color, 72DPI, 8 bits / channel. No adjustments (filters, like greyscale, etc ...) are selected / used. The layers are both normal, and opacity and fill are 100% (yes, in both files). I took two screenshots, and you can see the difference: http://www.flickr.com/photos/30465871@N05/4623864297/ http://www.flickr.com/photos/30465871@N05/4624469754/ Both colors are ff795d, but that doesn't matter, any color I use gives me the same problem: they both look different. Now, I know the CMYK settings (see screenshots) are different, but when the settings are the same the color changes. Why is this happening and how do I solve this problem? My guess is I'm working with different a type of RGB. It's sRGB IE61966 - 2.1 in the file I created (file info raw data) but I can't find that in the file that started with a screenshot. If that's the problem, how do I change / convert the RGB, once the file is already open? Thank you.

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• Making libmagic/file detect .docx files

  - by Jonatan Littke

  As seen elsewhere, docx, xlsx and pttx are ZIPs. When uploading them to my web application, file (via libmagic andpython-magic) detects them as being ZIP. I store the contents of the file as a blob in the database, but naturally I don't want to trust the user with what kind of file type this is. So I would like to trust file for and automatically generate a filename during download. I know one can modify /etc/magic but the format (magic(5)) is way too complicated for me. I found a bug report on the issue at Debian bugs but since it's from 2008 it doesn't seem to be fixed any time soon. I guess my only other alternative is to indeed trust the user (but still store the contents as a blob) and only check the file extension based on the file name. This way I can disallow some extensions and allow others. And when the user re-downloads his file, he can have it in whatever way he uploaded it. But this solution is insecure if the file is shared with others, since you can simply rename the file to allow uploading it. Any ideas? Lastly, I found a list of magic numbers for docx etc, but I'm unable to convert these into the magic(5) format.

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• AVConv increases song duration when converting MP3

  - by chauffch

  I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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• Barriers to IPv6 deployment: addressing

  - by sysadmin1138

  There are several things that are keeping IPv6 deployment from being a topic of active discussion here at my work. There are the usual technical issues, but one non-technical one appears to be a major stumbling block on the path to actually getting a deployment project going. Addresses, memorizing of. Specifically, IPv4 addresses are comprehensible, and IPv6 addresses just look like a big long string of hex. The human mind has real trouble memorizing lists of more than 7-8 items, and an IPv4 address (192.168.231.148) has four items in it which makes it easy for us to memorize. A fully populated IPv6 address has not only 8 sections, but each section has 4 hex digits in it. IPv6 addresses were not designed for memorization. To the technician who knows that the DNS server is at 192.168.42.42 (or more likely "42.42", since the company prefix is likely memorized), the idea of memorizing an IPv6 address fills them with dread. Which in turn makes them much less enthusiastic about participating in an IPv6 deployment project. Because of how our network works we're not fully dynamic in terms of v4 addressing. We have several to many subnets that are entirely statically assigned for a variety of reasons, chief among them being that the overhead of static DHCP assignments is perceived as being too great. Also, some devices still aren't smart enough to pull DNS addresses out of DHCP while also having a static assignment, and therefore require manually configured DNS settings. Therefore, some v6 address memorization will have to be done. We're not under any mandate to get v6 out the door, so we don't have pressure from the top. However, it is time to start prepping our infrastructure to handle IPv6 even if we don't convert wholesale. For those of you who have been in IPv6-land for a while, what short-cut methods do you use to discuss or keep track of subnets and specific/critical IP addresses? If I can help reduce some of the dread surrounding IPv6 we might get the project going.

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• ffmpeg - h264 to xvid creates large file

  - by fatnic

  I'm trying to use ffmpeg to convert a h264/aac video file to an xvid/mp3 file so I can play it in my ultra-cheap media player. At the moment the converted video file is TWICE the size of the original mp4. Is there any way to get a smaller file size without loosing too much quality? Even a drop to -qmin 1 is pretty awful! The command i'm using is ffmpeg -i input.mp4 -vcodec libxvid -sameq -acodec libmp3lame -ab 128k -ac 2 output.avi And the ffmpeg output is Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4' Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 Duration: 01:34:27.69, start: 0.000000, bitrate: 1520 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x304 [PAR 1:1 DAR 45:19], 1387 kb/s, 25 fps, 25 tbr, 25k tbn, 50 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s Output #0, avi, to 'output.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0(und): Video: mpeg4, yuv420p, 720x304 [PAR 1:1 DAR 45:19], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1(und): Audio: libmp3lame, 48000 Hz, stereo, s16, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1

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• Best video codec to store my own collection

  - by Jack

  Hello! I think this question has already been asked but with different flavours. My problem resised in the fact that my camera (Canon G9) creates video with almost raw codec (I think it's plain old MPEG) so a 10 minutes video is almost 900mb. I would like to convert them in a format that has a good trade-off between space and quality, but I would prefer having the quality as good as the original (of course this is not possible because of lossy compression) just saving as much space is possible with a minimal lose of quality. Which codec should I look for? H264? It seems to be the champion of the moment.. otherwise which other ones could I try? XviD? Which parameters should I use? I mean how many kbits/s is a fair good bitrate to keep high quality? And what about audio codec? video specs are 640x480 at 30fps or 1024x768 at 15fps.. thanks in advance!

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• Reconnecting OST with Exchange

  - by syrenity

  Hi. I have a quite big problem with customer's MS Exchange. The server got it's disk filled about 2 weeks ago, so it's currently offline. They plan to upgrade it, but not in hurry, as they use it mainly for OWA and back-up - the mails exchange is done via SMTP and POP3. Trying to diagnose some problem today, one of the users has (following the ISP instructions), removed the Exchange account from Outlook, which essentially left the OST orphaned. The user naturally didn't move the emails or any other data to the Archive / PST before, so these emails located on the OST only. So currently I'm trying to figure out how to restore them. There are 2 options: 1) Make the user buy some tool to convert them to PST, and import as archive / main Outlok file? 2) Reconnect the Outlook to Exchange (once it up), let it sync the old server content, then shutdown Outlook and replace the new OST with the old one, start Outlook again in offline mode and move these files to archive. 3) Any other method? Can someone advice what would be the best approach here? The used versions are Outlook 2007 and Exchange 2003. Thanks!

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• Did chkdsk make it harder to restore files?

  - by neyl

  My friend asked me to try and fix his loaded Sansa Clip + which wasn't playing. After opening it in MSC mode I discovered that the Music directory was empty and total of all files was only a few MB. However Disk properties showed me that it was 7Gb full. I then ran Tools - Error Checking and Windows dutifully informed me that disk was corrupt and I should run again Allowing Windows to Fix Errors. I did that and it told me everything was fixed and that all files were placed in FOUND.000 Dir. FOUND.000 was about 7.5 GB with FILE0000-1546 . CHK. (I am aware of methods like ChkBack to scan and convert to mp3 etc BUT Original filenames and structure needed!) Now I started getting worried that I made things worse! I have plenty of experience with Data Recovery Programs - Recuva, Restore My Files etc. and I was anyhow planning to use them to scan the drive. But NOW after CHKDSK "fixed" the drive maybe it modified critical FAT information vital for data recovery. So I run these programs and 0!!!. No trace of files! I tried a ton of Recovery Programs with same results TILL EaseUS Data Recovery Wizard found all files and I purchased program for $55! My Question In your opinion - did running CHKDSK with automatic fixing of errors make matters worse (i.e. many data recovery progs. didn't find a trace and they would have done if not for chkdsk) or was the filesystem too corrupt anyhow for regular File Recovery Progs.? If I would be a Professional - would I be responsible for running CHKDSK - automatic Fixing. Do you know of a better Data Recovery Program than EaseUs Data Recovery wizard - According to my experience I haven't found better!? Thanks

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• Configuring wsgi for a simple Python based site

  - by jbbarnes

  I have an Ubuntu 10.04 server that already has apache and wsgi working. I also have a python script that works just fine using the make_server command: if __name__ == '__main__': from wsgiref.simple_server import make_server srv = make_server('', 8080, display_status) srv.serve_forever() Now I would like to have the page always active without having to run the script manually. I looked at what Moin is doing. I found these lines in apache2.conf: WSGIScriptAlias /wiki /usr/local/share/moin/moin.wsgi WSGIDaemonProcess moin user=www-data group=www-data processes=5 threads=10 maximum-requests=1000 umask=0007 WSGIProcessGroup moin And moin.wsgi is as listed: import sys, os sys.path.insert(0, '/usr/local/share/moin') from MoinMoin.web.serving import make_application application = make_application(shared=True) QUESTION: Can I create a similar section in apache2.conf pointing to another wsgi file? Like this: WSGIScriptAlias /status /mypath/status.wsgi WSGIDaemonProcess status user=www-data group=www-data processes=5 threads=10 maximum-requests=1000 umask=0007 WSGIProcessGroup status And if so, what is required to convert my simple_server script into a daemonized process? Most of the information I find about wsgi is related to using it with frameworks like Django. I haven't found a simple howto detailing how to make this work. Thanks.

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• converting to MXF using ffmpeg

  - by Prakash

  I have been trying to use FFmpeg utility to convert a avi file using DNxHD to mxf format. I am using "FFmpeg" with params as following: ffmpeg -i ccvt_box.avi -vcodec dnxhd -video_size 1920x1080 -r 24 -b:v 115m ex.mxf The error it is giving : ffmpeg version N-43737-g76c3fff Copyright (c) 2000-2012 the FFmpeg developers built on Aug 20 2012 18:50:42 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: libavutil 51. 70.100 / 51. 70.100 libavcodec 54. 53.100 / 54. 53.100 libavformat 54. 25.104 / 54. 25.104 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 11.101 / 3. 11.101 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 Input #0, avi, from 'ccvt_box.avi': Duration: 00:00:10.00, start: 0.000000, bitrate: 691 kb/s Stream #0:0: Video: indeo5 (IV50 / 0x30355649), yuv410p, 340x344, 10 tbr, 10 tbn, 10 tbc Metadata: title : bob.avi [dnxhd @ 0x7fcd60818e00] video parameters incompatible with DNxHD Output #0, mxf, to 'ex.mxf': Stream #0:0: Video: dnxhd, yuv422p, 340x344, q=2-1024, 90k tbn, 24 tbc Metadata: title : bob.avi Stream mapping: Stream #0:0 -> #0:0 (indeo5 -> dnxhd) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

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• Auto-rotate rotated images with mogrify

  - by Frank Presencia Fandos

  Some of my images have been taken rotated but kept this data. The problem is that, when using mogrify to convert them from JPG to png, that data seems to dissapear. For showing this problem, I think the best is to show the script and an screenshot. Script with the code. Put it in a text file, give it execution permission, double click, run (from terminal if you wish) and wait a while. All the JPGs in that folder will be converted to png. #! /bin/bash echo "Converting JPG to png. Please don't close this window." mogrify -alpha on -format png *.JPG mogrify -alpha on -format -alpha on png *.jpg It works great and adds an alpha channel. This is personally useful when I edit them later, not to add the channel individually. Now the screenshot that illustrates the problem: As you can see, the original ones' (JPGs) preview is right, the modified preview is wrong, the Shotwell rendering is right and the GIMP edit is wrong and didn't even say the image was rotated, as it uses to do with other images. How can I edit my script to preserve the orientation?

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• Cut (smart edit) .mts (AVCHD Progressive) files un Ubuntu Lucid

  - by pts

  I have a bunch of .mts files containing AVCHD Progressive video recorded by a Panasonic camera, and I need software on Ubuntu Lucid with which I can remove the boring parts, and concatenate the interesting parts, all this without reencoding the video stream. It's OK for me to cut at keyframe boundary. If Avidemux was able to open the files, it would take about 60 hours of work for me to cut the files. (At least that was it last time I tried with similar videos, but of a file format supported by Avidemux.) So I need a fast, powerful and stable video editor, because I don't want that 60 hours of work go up to 240 or even 480 hours just because the tool is too slow or unstable or has a terrible UI. I've tried Avidemux 2.5.5 and 2.5.6, but they crash trying to open such a file, even if I convert the file to .avi first using mencoder -oac copy -ovc copy. mplayer can play the files. I've tried Avidemux 2.6.0, which can open the file, but it cannot jump to the previous or next keyframe etc. (if I make it jump to the next keyframe, and then to the previous keyframe, it doesn't end up at the original keyframe, sometimes displays an error etc.). Also I'm not sure if Avidemux 2.6.x would let me save the result without reencoding. I've tried Kdenlive 0.7.7.1, but playback is very choppy, and it cannot play audio at all (complaining that SDL cannot find the device; but many other programs on the system can play audio). It would be a pain to work with. I've tried converting the .mts file to .mkv using ffmpeg -i input.mts -vcodec copy -sameq -acodec copy -f matroska output.mkv, but that caused too much visible distortions in the video in both mplayer and Avidemux. I've tried converting the .mts file with TsRemux.exe, but Avidemux 2.5.x still can't open that file. Is there another program to cut and concatenate the files? Is there a preprocessor which would create a file (without reencoding the video) on which Avidemux wouldn't crash?

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