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  • How to reduce noise in Skype?

    - by tkc
    According to Alsamixer I do have a HDA Intel soundcard with Nvidia MCP77/78 HDMI chip (Realtek sound card on MSI notebook). When I use Skype under Ubuntu 12.04 for video calls, the other side hears background noise such as in Windows when you have your microphone boosted. In fact they can hear everything, even the fans turning on. There is nothing tweaked on Ubuntu's fresh installation. Also tried this site: https://code.launchpad.net/~ubuntu-audio-dev/+archive/alsa-daily/+packages , but there are no *.deb files that I can test if any fix the problem. The question is if there is a way to add/tweak something to enable on software level the noise cancellation like the Windows sound drivers have that option. I use my build in mic.

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  • exchange live feed with pre-recorded video for wireless internet camera to router

    - by nate
    I wasn't sure if this should be asked in Web Applications, or Network Engineering, or what... Long story short, I have a video camera with mic that is wirelessly connected to a router (NETGEAR R6200), which can then be viewed through an online service. I would like to be able to somehow exchange the live feed with a pre-recorded video, or image, preferably with pre-recorded sound (the sound of silence would be easiest). Can I place this inbetween the camera and the router, do I need to redirect the camera feed to my laptop first, and then push out the fake video/audio onto the router, without the service knowing the difference? Thanks much and I hope this is well understood!

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  • Ubuntu Surround Sound Help?

    - by Adam
    I am running Ubuntu 12.10 on a Sony Vaio VGC-RB34G ( http://reviews.cnet.com/desktops/sony-vaio-vgc-rb34g/4505-3118_7-31289053.html ) with an integrated Realtek High Definition Audio sound card. On Windows, I can set up the sound card to provide sound to my surround sound system from the Line Out, Line In, and Mic ports, as in all of these are producing audio. I have tried to use alsa to achieve this result with no luck. Is there any possible way to do this on Ubuntu? Thanks! Here is a Picture of the manager I have on Windows and the result I want to achieve https://picasaweb.google.com/106733704390489891165/November102012#5809378929755566722

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  • microfone problem in ubuntu 13.04

    - by mikke
    It seems a little poor that nobody has a solution for this problem! because ubuntu 13.04 is great and i have the same probs with internal and external mic's i have never read a steatment from ubuntu developers (and i am searching for a few week's!!) there are some solution-suggestions but they do not work! i find it a little bit weak that cannonical doesn't have a solution (it seems that this problem stays since 10.xx!) if there is no solution in the next time i'll change to another distribution! greeez mike

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  • Microphone doesn't give any input, yet music does

    - by user52643
    I've tried a few guides but nothing works as of yet. I've tried two microphones, both of which work on my Vista setup but do nothing on Ubuntu. Nothing is on mute both on the Sound utility or Skype (which I'm trying to use my mic for). I've been informed that when my speakers are playing very load, this can be heard on the other end of a call, too. See link for cat /proc/asound/card0/codec#0 results which are apparently relevant... _' N.B. I am noob x

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  • Bad sound quality and headphones not working

    - by wifi
    Using Ubuntu 10.10, on a HP Pavilion t3019.es, which has a Realtek ALC880 soundcard. It has 6 rear jack outputs, plus digital audio input and output, plus 3 front jacks (mic, headphones and a blue one which i don't know what's for). The sound on my computer is very low, and when i raise the volume up to 50%, it starts sounding distorted, crackling. Also, the headphones don't work when i plug them (it just keeps on playing through the speakers). I tried to comment the "/etc/modprobe.d/alsa-base.conf" file according to the soundcard and jacks in my computer, but none of the lines added worked (naturally, didn't added them at once). I found out that adding "options snd-hda-intel model=generic" to it made the sound better, but it's not as good as in Windows yet. Any ideas? Other than setting the PCM value, didn't work for me. Thanks.

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  • Upgrade to Ubuntu 13.10 disabled track pad, brightness, sound on Lenovo Z570

    - by Vizir
    I just upgraded to Ubuntu 13.10 using Ubuntu's software update. It seemed to go all right, however after the system restarted and booted up Ubuntu 13.10, it began experiencing several problems. Some of them were due to expected conflicts with the updated OS, however right now I cannot figure out why certain hardware functions are now "broken" and how to fix them. As far as I can tell, these are the sound (permanently muted), mic (picks up no sound), brightness (set at maximum brightness regardless of using keyboard shortcuts or moving the screen brightness slider) and trackpad (mouse does not move, however plugged-in USB mouse does) This sounds to me like a driver issue, however I cannot figure out how to re-enable my drivers, re-install them, or whatever I have to do here. This did not happen durring my upgrades from 12.04 to 12.10, or 12.10 to 13.04, so I'm at a total loss as to why this happened this time around. My computer is a Lenovo Z570, dual-booting Ubuntu and Windows 7 from GRUB 2. Windows is working fine as far as I can tell.

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  • Intel annonce une puce pour serveurs composée de 32 coeurs, faisant partie de sa future gamme de pro

    Mise à jour du 02.06.2010 par Katleen Intel annonce une puce pour serveurs composée de 32 coeurs, faisant partie de sa future gamme de produits Knights Intel vient d'annoncer une puce pour serveurs composée de 32 coeurs, cadencés à 1.2 GHz, élaborée sur une architecture mêlant des coeurs x86 ainsi que d'autres spécialisés pour répondre aux besoins spécifiques des serveurs à haute performance. Répondant au nom de Knights Ferry, ce processeur est "le plus rapide pouvant traiter plus de 500 Gigaflops de données", d'après son constructeur. Il marque les premiers pas d'une gamme destinée aux serveurs (Knights), qui repose sur une architecture MIC (Many Integrated Cores). Les proce...

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  • Ubuntu and surround sound

    - by keith
    I'm trying to connect my Logitech x.530 to my Dell studio 1558 and get 5.1 sound. I have 3 jacks and in Windows I automatically could switch between line in adn mic. Which means I could get 5.1 surround over these 3 jacks. Ubuntu does not realise that one of the three jacks should be switched to give sound to the center and LFE speaker. I tried all the tips in the forums, but nothing works. I can only select 4.0 sound in pulse audio. As the dell and the IDT 92hd89e chip for the sound are probably famous laptops I thought there must be someone who has similar issues in the 5.1 sound. I would like to stay at Linux but without this hardware support it's almost impossible.

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  • Silverlight on Windows Phone

    This was a great announcement last week at MIX10: the programming model for the upcoming Windows Phone 7 is Silverlight!   For now it is Silverlight 3, with the possibility to use phone specific features: orientation location & map control (GPS) mic push notifications motion detection accelerometer compass light proximity contacts So we have the same programming model we already know, develop in Visual Studio, test with the built-in emulator or deploy...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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  • Audio recording and playback in Silverlight

    - by Ramesh
    I have a Silverlight 4 application that records user's voice through the mic. Now, as soon as the recording is completed, I need to play the recorded voice back to the user before posting it to the server. Is it at all possible to play it back to the user without getting into format conversions etc? Any ideas are welcome. Thanks!

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  • Android - Getting audio to play through earpiece

    - by Donal Rafferty
    I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class. My problem is that when I play it out it plays vis the speaker phone. I want it to play out via the ear piece on the device. Here is my code: public class LoopProg extends Activity { boolean isRecording; //currently not used AudioManager am; int count = 0; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); am = (AudioManager) getSystemService(Context.AUDIO_SERVICE); am.setMicrophoneMute(true); while(count <= 1000000){ Record record = new Record(); record.run(); count ++; Log.d("COUNT", "Count is : " + count); } } public class Record extends Thread { static final int bufferSize = 200000; final short[] buffer = new short[bufferSize]; short[] readBuffer = new short[bufferSize]; public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM); am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC); int ok = am.getRouting(AudioManager.ROUTE_EARPIECE); Log.d("ROUTING", "getRouting = " + ok); setVolumeControlStream(AudioManager.STREAM_VOICE_CALL); //am.setSpeakerphoneOn(true); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); am.setSpeakerphoneOn(false); Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn()); atrack.setPlaybackRate(11025); byte[] buffer = new byte[buffersize]; arec.startRecording(); atrack.play(); while(isRecording) { arec.read(buffer, 0, buffersize); atrack.write(buffer, 0, buffer.length); } arec.stop(); atrack.stop(); isRecording = false; } } } As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeatPoneOn method seems to have no effect at all, neither does the routing method. Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

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  • iPhone htaccess redirect redirecting the iPad

    - by Luis Armando
    how can I avoid this from happening? I currently have: RewriteEngine On RewriteCond %{HTTP_USER_AGENT} "smartphone|rover|ipaq|au-mic,|alcatel|ericy|vodafone\/|wap1\.|wap2\.|iPhone|android"[NC] RewriteCond %{REQUEST_URI} !^/m/ RewriteRule ^.*$ http://www.mydomain.com/m/$0 [R=301,L] but when someone logs in with an iPad he/she gets redirected =/ to the mobile version as well and I'd like them to see the normal site (without the /m/)

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  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

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  • github - Adding a file to branch

    - by Fiona
    Hi there, So I'm following thiese instructions:http://mark-kirby.co.uk/tag/osx/ and so far i've cloned the project I want to work on and created a branch. Now I wish to add files that exist in another folder on my machine... but I keep getting the following: fatal: pathspec 'Users/mic/OnePageCRMVC/MKTsite25-05/index.html' did not match any files However, the file definitely does exist... Am I trying to do something that is not allowed and the error message is throwing me off? Regards, Fiona

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  • Permissions error when connecting to EC2 via SSH on Mac OSx

    - by resonantmedia
    I am new to EC2. I created my security credentials from this site: http://paulstamatiou.com/how-to-getting-started-with-amazon-ec2 It worked great, I rebooted and now when I try to connect I get a login/password prompt. (Which I never set up.) After several attempts I get this error: Permission denied (publickey,gssapi-with-mic). What am I doing wrong? Thanks, Josh

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  • Audio/Voice Visualization

    - by Neurofluxation
    Hey you Objective-C bods. Does anyone know how I would go about changing (transforming) an image based on the input from the Microphone on the iPhone? i.e. When a user speaks into the Mic, the image will pulse or skew. Thanking you!!

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  • Error #2032 displayed in html page

    - by Rajeev
    I have a app which is used to testmicrophone.But when include in html page it displays Error #2032.How to resolve this. This is the HTML code <div style="display: inline;float:center;"> <object width="100" height="100"> <embed src="mic.swf" width="250" height="250"> </embed> </object> </div>

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  • How to make VLC play .vlm config file in "With no interface mode"?

    - by Ole Jak
    How to make VLC play .vlm config file in "With no interface mode" on windows? So I have .vlm config file that should stream audio from mic to localhost so no vlc ui needed. If I say to windows "play .vlm file with vlc" it plays correctly starts server where I need and streams data. but how to do such thing manulay from cmd (so we suppouse we can call vlc.exe by vlc and we are now in folder with vlc.exe and vlcConfig.vlm file)

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  • How could I stop ssh offering a wrong key?

    - by Alvaro Maceda
    (This is a problem with ssh, not gitolite) I've configured gitolite on my home server (ubuntu 12.04 server, open-ssh). I want an special identityfile to administer the repositories, so I need to access throught ssh to my own host ussing two different identity keys. This is the content of my .ssh/config file: Host gitadmin.gammu.com User git IdentityFile /home/alvaro/.ssh/id_gitolite_mantra Host git.gammu.com User git IdentityFile /home/alvaro/.ssh/id_alvaro_mantra This is the content of my hosts file: # Git 127.0.0.1 gitadmin.gammu.com 127.0.0.1 git.gammu.com So I should be able to communicate with gitolite this way to access with the "normal" account: $ssh git.gammu.com and this way to access with the administrative account: $ssh gitadmin.gammu.com When I try to access with the normal account, all is ok: alvaro@mantra:~/.ssh$ ssh git.gammu.com PTY allocation request failed on channel 0 hello alvaro, this is gitolite 2.2-1 (Debian) running on git 1.7.9.5 the gitolite config gives you the following access: @R_ @W_ testing Connection to git.gammu.com closed. When I do the same with the administrative account: alvaro@mantra:~$ ssh gitadmin.gammu.com PTY allocation request failed on channel 0 hello alvaro, this is gitolite 2.2-1 (Debian) running on git 1.7.9.5 the gitolite config gives you the following access: @R_ @W_ testing Connection to gitadmin.gammu.com closed. It should show the administrative repository. If I launch ssh with verbose option: ssh -vvv gitadmin.gammu.com ... debug1: SSH2_MSG_SERVICE_REQUEST sent debug2: service_accept: ssh-userauth debug1: SSH2_MSG_SERVICE_ACCEPT received debug2: key: /home/alvaro/.ssh/id_alvaro_mantra (0x7f7cb6c0fbc0) debug2: key: /home/alvaro/.ssh/id_gitolite_mantra (0x7f7cb6c044d0) debug1: Authentications that can continue: publickey,password debug3: start over, passed a different list publickey,password debug3: preferred gssapi-keyex,gssapi-with-mic,publickey,keyboard-interactive,password debug3: authmethod_lookup publickey debug3: remaining preferred: keyboard-interactive,password debug3: authmethod_is_enabled publickey debug1: Next authentication method: publickey debug1: Offering RSA public key: /home/alvaro/.ssh/id_alvaro_mantra debug3: send_pubkey_test debug2: we sent a publickey packet, wait for reply debug1: Server accepts key: pkalg ssh-rsa blen 279 ... It's offering the key id_alvaro_mantra, and it should'nt!! The same happens when I specify the key with the -i option: ssh -i /home/alvaro/.ssh/id_gitolite_mantra -vvv gitadmin.gammu.com ... debug1: SSH2_MSG_SERVICE_REQUEST sent debug2: service_accept: ssh-userauth debug1: SSH2_MSG_SERVICE_ACCEPT received debug2: key: /home/alvaro/.ssh/id_alvaro_mantra (0x7fa365237f90) debug2: key: /home/alvaro/.ssh/id_gitolite_mantra (0x7fa365230550) debug2: key: /home/alvaro/.ssh/id_gitolite_mantra (0x7fa365231050) debug1: Authentications that can continue: publickey,password debug3: start over, passed a different list publickey,password debug3: preferred gssapi-keyex,gssapi-with-mic,publickey,keyboard-interactive,password debug3: authmethod_lookup publickey debug3: remaining preferred: keyboard-interactive,password debug3: authmethod_is_enabled publickey debug1: Next authentication method: publickey debug1: Offering RSA public key: /home/alvaro/.ssh/id_alvaro_mantra debug3: send_pubkey_test debug2: we sent a publickey packet, wait for reply debug1: Server accepts key: pkalg ssh-rsa blen 279 debug2: input_userauth_pk_ok: fp 36:b1:43:36:af:4f:00:e5:e1:39:50:7e:07:80:14:26 debug3: sign_and_send_pubkey: RSA 36:b1:43:36:af:4f:00:e5:e1:39:50:7e:07:80:14:26 debug1: Authentication succeeded (publickey). ... What the hell is happening??? I'm missing something, but I can't find what. These are the contents of my home dir: -rw-rw-r-- 1 alvaro alvaro 395 nov 14 18:00 authorized_keys -rw-rw-r-- 1 alvaro alvaro 326 nov 21 10:21 config -rw------- 1 alvaro alvaro 137 nov 20 20:26 environment -rw------- 1 alvaro alvaro 1766 nov 20 21:41 id_alvaromaceda.es -rw-r--r-- 1 alvaro alvaro 404 nov 20 21:41 id_alvaromaceda.es.pub -rw------- 1 alvaro alvaro 1766 nov 14 17:59 id_alvaro_mantra -rw-r--r-- 1 alvaro alvaro 395 nov 14 17:59 id_alvaro_mantra.pub -rw------- 1 alvaro alvaro 771 nov 14 18:03 id_developer_mantra -rw------- 1 alvaro alvaro 1679 nov 20 12:37 id_dos_pruebasgit -rw-r--r-- 1 alvaro alvaro 395 nov 20 12:37 id_dos_pruebasgit.pub -rw------- 1 alvaro alvaro 1679 nov 20 12:46 id_gitolite_mantra -rw-r--r-- 1 alvaro alvaro 397 nov 20 12:46 id_gitolite_mantra.pub -rw------- 1 alvaro alvaro 1675 nov 20 21:44 id_gitpruebas.es -rw-r--r-- 1 alvaro alvaro 408 nov 20 21:44 id_gitpruebas.es.pub -rw------- 1 alvaro alvaro 1679 nov 20 12:34 id_uno_pruebasgit -rw-r--r-- 1 alvaro alvaro 395 nov 20 12:34 id_uno_pruebasgit.pub -rw-r--r-- 1 alvaro alvaro 2434 nov 21 10:11 known_hosts There are a bunch of other keys which aren't offered... why id_alvaro_mantra is offered and not the other keys? I can't understand. I need some help, don't know where to look....

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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  • vagrant fails to bring up additional adapter for centos vm using virtual box provider

    - by Anadi Misra
    this is in continuation of the question asked here about host only adapter on dhcp I upgraded to vagrant 1.6.3 and the updated Vagrantfile to following setting for multiple adapters # add additional adapter for inter machine networking dev.vm.network :private_network, :type => "dhcp", :adapter => "2", :netmask => "255.255.255.0" it goes through creating adapters but then fails bringing up the mic on vm Anadis-MacBook-Pro:full-stack-env anadi$ vagrant up Bringing machine 'full-stack-env' up with 'virtualbox' provider... ==> full-stack-env: Clearing any previously set forwarded ports... ==> full-stack-env: Clearing any previously set network interfaces... ==> full-stack-env: Preparing network interfaces based on configuration... full-stack-env: Adapter 1: nat full-stack-env: Adapter 2: hostonly ==> full-stack-env: Forwarding ports... full-stack-env: 22 => 4223 (adapter 1) full-stack-env: 8080 => 8090 (adapter 1) ==> full-stack-env: Running 'pre-boot' VM customizations... ==> full-stack-env: Booting VM... ==> full-stack-env: Waiting for machine to boot. This may take a few minutes... full-stack-env: SSH address: 127.0.0.1:4223 full-stack-env: SSH username: vagrant full-stack-env: SSH auth method: private key full-stack-env: Warning: Connection timeout. Retrying... full-stack-env: Warning: Connection timeout. Retrying... full-stack-env: Warning: Remote connection disconnect. Retrying... ==> full-stack-env: Machine booted and ready! ==> full-stack-env: Checking for guest additions in VM... ==> full-stack-env: Setting hostname... ==> full-stack-env: Configuring and enabling network interfaces... The following SSH command responded with a non-zero exit status. Vagrant assumes that this means the command failed! ARPCHECK=no /sbin/ifup eth 2> /dev/null Stdout from the command: Device eth does not seem to be present, delaying initialization. Stderr from the command: how ever when I log in to the environment I see two network interfaces as expected Anadis-MacBook-Pro:full-stack-env anadi$ vagrant ssh Last login: Wed Jun 4 12:54:47 2014 from 10.0.2.2 [vagrant@full-stack-env ~]$ ifconfig eth0 Link encap:Ethernet HWaddr 08:00:27:BD:39:57 inet addr:10.0.2.15 Bcast:10.0.2.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:febd:3957/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:511 errors:0 dropped:0 overruns:0 frame:0 TX packets:360 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:54574 (53.2 KiB) TX bytes:46675 (45.5 KiB) eth1 Link encap:Ethernet HWaddr 08:00:27:A3:86:C9 inet addr:172.28.128.3 Bcast:172.28.128.255 Mask:255.255.255.0 inet6 addr: fe80::a00:27ff:fea3:86c9/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:5 errors:0 dropped:0 overruns:0 frame:0 TX packets:9 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:1360 (1.3 KiB) TX bytes:894 (894.0 b) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) I am bit confused here on why it is trying to add another mic (eth2)? In the VM I used for creating this vagrant box, I had added two NICs already.

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