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• Custom HTTP Status Codes (a la Twitter 420: Enhance Your Calm) [migrated]

  - by Max Bucknell

  I'm currently implementing an HTTP API, my first ever. I've been spending a lot of time looking at the Wikipedia page for HTTP status codes, because I'm determined to implement the right codes for the right situations. Listed on that page is a code with number 420, which is a custom code that Twitter used to use for rate limiting. There is already a code for rate limiting, though. It's 429. This led me to wonder why they would set a custom one, when there is already a use case. Is that just being cute? And if so, then which circumstances would make it acceptable to return a different status code, and what, if any problems may clients have with it? I read somewhere that Mozilla doesn't implement the joke 418: I’m a teapot response, which makes me think that clients choose which status codes they implement. If that's true, then I can imagine Twitter's funny little enhance your calm code being problematic. Unless I'm mistaken, and we can appropriate any code number to mean whatever we like, and that only convention dictates that 404 means not found, and 429 means take it easy.

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• Slow internet using Arch Linux

  - by GZaidman

  after a week or so of using Arch Linux I cant access the internet - it takes around 5 mins to load google (most of the other websites just give me a timeout), pacman's downloading speed range between 5-2Kbs, and pinging google takes around 9Kms. I'm connected using wireless network (wifi card is Intel Ultimate 6300 and router is Edimax 6524n). Every other Windows machine that's connected to the network (and even the T410 running Windows) is fine, so the problem lies in Linux. So far, i checked the resolv.conf file (my router ip address is listed), and the hosts file (pretty much default), and I disabled the ipv6 module. None of that helped. PS: i'm using NetworkManager (but the problem still occurs when connecting using wicd) running on Gnome3. Thanks in advance for any help you can provide! EDIT: something really strange happens whenever I ping google: i get an unknown host 'google.com', but the bit rate from the card jumps at the exact second I ping google (so far, the bit rate jumped to 54Mb/s from 1Mb/s over the course of 4 pings).

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• Mysterious winamp ratings that come from nowhere

  - by Phenom

  Winamp gives the option to rate songs. This is not a new feature. Everytime I would reinstall Windows and reinstall Winamp my ratings would all go away as expected, and I would have to re-rate everything. However, this time when I reinstalled Winamp, after using it for a few days, songs would appear in my top rated list as being rated. Some of these I remember rating a long time ago before I reinstalled Windows, and some of these I don't remember ever rating. Where are these ratings coming from?

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• Probability of failure with larger number of network elements

  - by MikeKulls

  I'm having a discussion with a work colleague. I'm saying that a network with 100 elements will have pretty much 10 times as many failures as a network with 10 elements, ie a tech will need to replace faulty hardware 10 times more often. He suggests that the failure rate doesn't go up in a linear fashion and the failure rate will be significantly less than 10x, in fact only slightly more failures. This is not the probability of an outage etc, we are just talking in relation to the amount of parts that a tech would need to swap out in a given time frame.

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• Is to possible to achieve the SATA II's 3Gbps (375MBps) between home network storage and home computers?

  - by techaddict

  Gigabit ethernet only has 1Gbps (125MBps), whereas SATA II has up to three times that rate. Is it possible to achieve the rate of three times Gigabit ethernet connection which is SATA II speeds, between home network storage and the laptops and desktops with SATA II hard drives? If so, how? Or, is the limitation of a gigabit ethernet port on the laptops the limitng factor, making 1000Gbps the fastest practical transfer speed possible? (practical, meaning that without taking apart the laptop and doing physical modifications like branching a SATA II transfer cable, etc.) -- I just realized -- wouldn't a USB 3.0 cable do the trick? Since USB 3.0 can reach up to 675MBps?

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• How to diagnose very slow pagefile

  - by svick

  Quite often, one of the applications I use freezes (“does not respond”) for a while, in extreme cases for few minutes. This happens especially when when switching apps. During this time, the HDD light flashes constantly and perfmon show that HDD is used 100% of the time (OTOH, CPU isn't) and that pagefile is being read (which is to be expected when switching apps), but at a very slow rate. When I sort the disk table in perfmon by read or write, the file read and wrote the most is the pagefile, but it's still quite low rate (I don't remember the numbers). How can I diagnose what's causing this? I use Windows Vista, and the computer is quite ordinary two years old laptop.

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• Cisco IOS ACL types

  - by cjavapro

  The built in command help list displays access list types based on which range. router1(config)#access-list ? <1-99> IP standard access list <100-199> IP extended access list <1100-1199> Extended 48-bit MAC address access list <1300-1999> IP standard access list (expanded range) <200-299> Protocol type-code access list <2000-2699> IP extended access list (expanded range) <700-799> 48-bit MAC address access list dynamic-extended Extend the dynamic ACL absolute timer rate-limit Simple rate-limit specific access list router1(config)# What are each of the types? Can multiple types of ACLs be applied to a given interface?

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• Determining Performance Limits

  - by JeffV

  I have a number of windows processes that pass messages between them hat a high rate using tcp to local host. Aside from testing on actual hardware how can I assess what my hardware limit will be. These applications are not doing CPU intensive work, mostly decomposing and combining messages, scanning over them for special flag in the data etc.. The message size is typically 3k and the rate is typically ~10k messages per second. ~30MB per second between processing stages. There may be 10 or more stages depending. For this type of application, what should I look to for assessing performance? What do I look for in a server performance wise? I am currently running an XEON L5408 with 32 GB ram. But I am assuming cache is more important than actual ram size as I am barely touching the ram.

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• 802.11 Capture Frame

  - by ALi

  I am using wireshark in monitor mode to capture all the frame. I buffered all the QosData in order to calculate the biterate of each station on the network. I calculate the biterate but it is not an accurate value. what i need to know is when the station use the network alone and when it uses it with another station? i know if two station with different wifi card for exemple ( st1 802.11g 54 Mb/s, st2 802.11n 300 Mb/s) if st1 uses alone the line the biterate used is 28 MB/s and if st2 uses alone the network the bit rate increase to approximately 150Mb/s but if the two station use the network at the same time the st1 lost about 80% of its bit rate because of the st1 wifi card thx in advance

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• ffmpeg add two audio streams to video

  - by Tossin Hausen

  I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

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• php-fpm or nginx: bad gateway

  - by John Tate

  I'm getting a bad gateway error all the sudden for a site. I didn't change the configuration for the site, I just added a new server config where I put them under /etc/nginx/servers and it stopped working. The new server works, and there is no conflict between the php-fpm listen addresses. server { listen 80; server_name obfuscated.onion; location = / { root /var/www/sites/obfuse; index index.php; } location / { root /var/www/sites/obfuse; index index.php; if (!-f $request_filename) { rewrite ^(.*)$ /index.php?q=$1 last; break; } if (!-d $request_filename) { rewrite ^(.*)$ /index.php?q=$1 last; break; } } error_page 404 /index.php; location ~* ^.+.(jpg|jpeg|gif|css|png|js|ico)$ { root /var/www/sites/obfuse; access_log off; expires 30d; } location ~ \.php$ { fastcgi_pass 127.0.0.1:9000; fastcgi_index index.php; fastcgi_param SCRIPT_FILENAME /var/www/sites/obfuse$fastcgi_script_name; fastcgi_param QUERY_STRING $query_string; fastcgi_param REQUEST_METHOD $request_method; fastcgi_param CONTENT_TYPE $content_type; fastcgi_param CONTENT_LENGTH $content_length; fastcgi_param PATH_INFO $fastcgi_script_name; include fastcgi_params; } } There is nothing unusual in php-fpm's log even when I raised the level to debug. [24-Jun-2013 09:10:37.357943] DEBUG: pid 6756, fpm_scoreboard_init_main(), line 40: got clock tick '100' [24-Jun-2013 09:10:37.358950] DEBUG: pid 6756, fpm_event_init_main(), line 333: event module is kqueue and 1 fds have been reserved [24-Jun-2013 09:10:37.358978] NOTICE: pid 6756, fpm_init(), line 83: fpm is running, pid 6756 [24-Jun-2013 09:10:37.359009] DEBUG: pid 6756, main(), line 1832: Sending "1" (OK) to parent via fd=5 [24-Jun-2013 09:10:37.389215] DEBUG: pid 6756, fpm_children_make(), line 421: [pool cyruserv] child 22288 started [24-Jun-2013 09:10:37.391343] DEBUG: pid 6756, fpm_children_make(), line 421: [pool cyruserv] child 21911 started [24-Jun-2013 09:10:37.391914] DEBUG: pid 6756, fpm_event_loop(), line 362: 5776 bytes have been reserved in SHM [24-Jun-2013 09:10:37.391941] NOTICE: pid 6756, fpm_event_loop(), line 363: ready to handle connections [24-Jun-2013 09:10:38.393048] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 [24-Jun-2013 09:10:39.403032] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 [24-Jun-2013 09:10:40.413070] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 I don't know why this has started happening, but the logs are not telling me anything. Please ask for more information than this, you'll probably need it.

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• how to record mic input and pipe the output to another program

  - by acrs

  Hi everyone Im trying to follow a tutorial on generating truly random bits How To Generate Truly Random Bits This is the command from the tutorial but it does not work rec -c 1 -d /dev/dsp -r 8000 -t wav -s w - | ./noise-filter >bits I know i can record my mic input using rec -c 1 no.wav this is the command i tried using rec -c 1 -r 8000 -t wav -s noise.wav | ./noise-filter >bits but i get root@xxc:~/cc# rec -c 1 -r 8000 -t wav -s noise.wav - | ./noise-filter >bits rec WARN formats: can't set sample rate 8000; using 48000 rec FAIL sox: Input files must have the same sample-rate I have complied noise-filter noise-filter I think the tutorial is using an older version of SOX and REC I'm using sox: SoX v14.3.2 on Ubuntu 12.04 server Can someone please help me ?

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• 1080 HD video display - Windows 7

  - by blasteralfred

  I am running Windows 7 in my Dell Studio 1555 Laptop. I have ATI Mobility Radeon HD 4570 graphics card inside. When I try to watch a video, I can see only some disturbed texture. I can hear the audio and no prob with that. The I can see the video frames peeking sometimes but is pretty slower. I have DirectX (directx_feb2010_redist) and Win7 codecs installed. Here are some screenshots; The video details are; Name: video.mp4 Frame width: 1920 Frame height: 1080 Frame rate: 29 fps data Rate: 7958 kbps Total bitrate: 8052 kbps Just comment if you want more info.

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• how to limit upload bandwidth per user in linux?

  - by Gihan Lasita

  Can anyone provide the tc command to limit upload bandwidth per user in Debian Lenny? I found that to mark packets per user with iptables I can use the following command iptables -t mangle -A OUTPUT -p tcp -m owner --uid-owner testuser -j MARK --set-mark 500 but I have no idea how to use tc update by running following commands, i managed to limit testuser upload bandwidth to 10Mbit iptables -t mangle -N HTB_OUT iptables -t mangle -I POSTROUTING -j HTB_OUT iptables -t mangle -A HTB_OUT -j MARK --set-mark 30 iptables -t mangle -A HTB_OUT -m owner --uid-owner testuser -j MARK --set-mark 10 tc qdisc replace dev eth0 root handle 1: htb default 30 tc class replace dev eth0 parent 1: classid 1:1 htb rate 10Mbit burst 5k tc class replace dev eth0 parent 1:1 classid 1:10 htb rate 10Mbit ceil 10Mbit tc qdisc replace dev eth0 parent 1:10 handle 10: sfq perturb 10 tc filter add dev eth0 parent 1:0 prio 0 protocol ip handle 10 fw flowid 1:10 now the problem is, i do not want to limit testuser's FTP bandwidth but by running above commands FTP speed also limited to 10Mbit. Regards

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• Is this hard disk dead?

  - by Korjavin Ivan

  Not sure, is this right site for this Q, but let me try Last time i have problem with hard disk. Sometimes its do strange sound, and i get it from logs: $dmesg | grep ata4 [29409.945516] ata4.00: exception Emask 0x10 SAct 0xf SErr 0x90202 action 0xe frozen [29409.945529] ata4.00: irq_stat 0x00400000, PHY RDY changed [29409.945538] ata4: SError: { RecovComm Persist PHYRdyChg 10B8B } [29409.945546] ata4.00: failed command: READ FPDMA QUEUED [29409.945562] ata4.00: cmd 60/30:00:56:22:5f/00:00:00:00:00/40 tag 0 ncq 24576 in [29409.945573] ata4.00: status: { DRDY } [29409.945580] ata4.00: failed command: READ FPDMA QUEUED [29409.945594] ata4.00: cmd 60/18:08:8e:22:5f/00:00:00:00:00/40 tag 1 ncq 12288 in [29409.945605] ata4.00: status: { DRDY } [29409.945611] ata4.00: failed command: READ FPDMA QUEUED [29409.945625] ata4.00: cmd 60/08:10:46:02:66/00:00:00:00:00/40 tag 2 ncq 4096 in [29409.945635] ata4.00: status: { DRDY } [29409.945641] ata4.00: failed command: READ FPDMA QUEUED [29409.945656] ata4.00: cmd 60/80:18:ee:04:66/00:00:00:00:00/40 tag 3 ncq 65536 in [29409.945666] ata4.00: status: { DRDY } [29409.945679] ata4: hard resetting link [29413.976083] ata4: softreset failed (device not ready) [29413.976097] ata4: applying SB600 PMP SRST workaround and retrying [29414.148070] ata4: SATA link up 3.0 Gbps (SStatus 123 SControl 300) [29414.184986] ata4.00: SB600 AHCI: limiting to 255 sectors per cmd [29414.243280] ata4.00: SB600 AHCI: limiting to 255 sectors per cmd [29414.243292] ata4.00: configured for UDMA/133 [29414.243324] ata4: EH complete [680674.804563] ata4: exception Emask 0x50 SAct 0x0 SErr 0x90a02 action 0xe frozen [680674.804575] ata4: irq_stat 0x00400000, PHY RDY changed [680674.804584] ata4: SError: { RecovComm Persist HostInt PHYRdyChg 10B8B } [680674.804603] ata4: hard resetting link [680678.840561] ata4: softreset failed (device not ready) Is this ata4 sata hard drive dead? Must i change it ASAP ? Need I specify more info?

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• different ways decrease cpu usage

  - by lego69

  I use nothing but my cpu usage has got 100%, how can I decrease this rate?

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• bytes per track

  - by Phenom

  If you know the transfer rate, how do you calculate the bytes per track?

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• netlogon errors

  - by rorr

  I have two instances of mssql 2005 and am using CA XOSoft replication. The master is a failover cluster and the replica is a standalone server. They are all running Server 2003 sp2 x64. Same patch levels on all servers. This setup has worked great for several months until we recently restricted the RPC ports on both nodes of the master(5000 - 6000 using rpccfg.exe). We have to implement egress filtering, thus the limiting of the ports. We began receiving login errors for sql windows authentication and NETLOGON Event ID: 5719: This computer was not able to set up a secure session with a domain controller in domain due to the following: Not enough storage is available to process this command. This may lead to authentication problems. Make sure that this computer is connected to the network. If the problem persists, please contact your domain administrator. We also see group policies failing to update and cluster file shares go offline at the same time. The RPC ports were set back to default when we started seeing these problems and the servers rebooted, but the problems persist. The domain controllers are not showing any errors. Running dcdiag and netdiag shows everything is fine. We have noticed that the XOSoft service ws_rep.exe is using a lot of handles(8 - 9k), about the same number that sqlserver is using. As soon as xosoft replication is stopped the login errors cease and everything functions correctly. I have opened a ticket with CA for XOSoft, but I'm not sure that the problem is actually xosoft, but that it is the one bringing the problem to light. I'm looking for tips on debugging RPC problems. Specifically on limiting the ports and then reverting the changes.

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• SoX on Windows 7 64-Bit Outfile Missing

  - by Christian

  I have come across the strangest problem when trying to run sox.exe on my Windows 7 installation. Whenever I try and record audio it works without any issues but it will not output an audio file. The crazy thing is that when I use the play command it successfully plays what I just recorded. Has anyone ever heard of this happening? Here are the commands (and output) that I'm using: C:\Program Files (x86)\Vox\sox-14-4-0>sox -d test.wav trim 0 00:05 Input File : 'default' (waveaudio) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:05.03 [00:00:00.00] Out:240k [ | ] Clip:0 Done. C:\Program Files (x86)\Vox\sox-14-4-0>play test.wav test.wav: File Size: 960k Bit Rate: 1.54M Encoding: Signed PCM Channels: 2 @ 16-bit Samplerate: 48000Hz Replaygain: off Duration: 00:00:05.00 In:100% 00:00:05.00 [00:00:00.00] Out:240k [ | ] Clip:0 Done. Am I losing my mind or is something up here?

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• On Windows 7, how can I tell if a recording is multi-channel without third party tools? [migrated]

  - by engineerchuan

  A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

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• Postfix additional transports - is it working?

  - by threecheeseopera

  I have enabled two additional transports in my postfix config to deal with recipient domains that demand connection limiting, per the instructions here at serverfault. However, I have no idea if this is working or not; in fact, I think it is not working, due to the send speeds I am seeing in the logs. How might I determine if my additional transports are working? If they aren't, do you have any tips on figuring out why? And, do you have any comments on my particular configuration? (am I a bucket of fail?) I have enabled the additional transports in master.cf: smtp inet n - - - - smtpd careful unix - - n - 10 smtp -o smtp_connect_timeout=5 -o smtp_helo_timeout=5 cautious unix - - n - - smtp -o smtp_connect_timeout=5 -o smtp_helo_timeout=5 I have set up the transport mapping file /etc/postfix/transport: hotmail.com cautious: yahoo.com careful: gmail.com cautious: earthlink.net cautious: msn.com cautious: live.com cautious: aol.com careful: I have set up the transport mapping and some connection-limiting settings in main.cf: transport_maps = hash:/etc/postfix/transport careful_initial_destination_concurrency = 5 careful_destination_concurrency_limit = 10 cautious_destination_concurrency_limit = 50 Finally, I have run converted the transport file to a db per the postfix docs: #> postmap /etc/postfix/transport And then restarted postfix. I do see my transport_maps setting when I run postconf, but I do not see any of the transport-specific settings ('careful_xxx_yyy_zzz'). Also the mail logs do not appear to be different in any way to what they were previously. Thanks!!!

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• Throttling apache downloads selectively

  - by Synchro

  I have a linux box running Debian Sarge (old I know) and apache 2.0.54. It serves two kinds of files - regular web pages and small images, and a lot of large podcast mp3s. The podcast downloads swamp the connection and make the rest of the site unresponsive, so I'm looking to throttle the data transfer rate (not the request rate) of just the podcasts. I've set up haproxy using this technique which does what it says it will, but solves a different problem - even only 5 simultaneous podcast downloads is enough to saturate the link. In a perfect world, haproxy would support per-connection throttling, but it doesn't. So far I've looked at mod_bw (won't compile for me, seems unsupported), mod_cband (unsupported, widely reported as problematic) and iptables using tc. The iptables approach would allow me to throttle things, but would not be at all selective, slowing down everything on the server, not just the podcasts, so would just move the bottleneck without changing overall behaviour. Ideas?

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• How to force 640*480@60Hz screen resolution on xubuntu 12.04

  - by c2h2

  It seems xubuntu won't be able to correctly set resolution at 640*480@60Hz at its Display settings. And I am unable to correctly my super small 6.4 inch Mitsubishi VGA panel via VGA cable. I have tried to hack both X11 conf /etc/X11/Xorg.conf and xfce4 conf, but all the document I can find is outdated. and conf files are changed into other location. Can someone give me a hand and I'll mark correct for other people to use? Thanks! EDIT: The board is an Intel Atom D2700, gpu is SGX545. I tried to use xrandr --output default --mode 640*480 It seems works fine, but refresh rate is 75Hz, but the screen only suports 60Hz So I used xrandr --output default --mode 640*480 --rate 60 but it give error: xrandr: Failed to get size of gamma for output default Can anyone pointing any directions?

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• Are there any command line utilities which can calculate and/or limit how fast a pipe is running?

  - by stsquad

  I'm doing some basic stress testing of a Linux kernel network IWF with netcat. The set-up is fairly simple. On the target side: nc -l -p 10000 > /dev/null And on my desktop I was running: cat /dev/urandom | nc 192.168.0.20 10000 I'm using urandom for some poor-mans fuzz testing. However I find that even at this rate I can break something quite quickly. EDIT So I've been playing with trickle to rate limit how fast I'm generating data: cat /dev/urandom | trickle -u 10 nc 192.168.0.20 10000 But it's hard to tell if this is working. What would be really useful is a the pv equivilent of trickle that can work with pipes.

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• Data transfer speed to USB storage connected to wifi router very slow

  - by RonakG

  Here is my setup. A Linksys Cisco E3200 wifi router. A MacbookPro running OS X Lion 10.7.4. A Seagate GoFlex 1TB hard drive connected to wifi router via the USB port. When I try to transfer data from my MBP to the HDD, the data transfer rate is very low. I'm getting around 3MB/s write speed. This is very slow compared to the speed I get when HDD is directly connected to the MBP. The HDD is NTFS formatted. And the router provides access to HDD using Samba share. So I connect to the HDD using smb://. What is the limiting factor here affecting the data transfer rate?

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