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  • Windows XP-64 loses audio sounds, drive letters... why?

    - by Ira Baxter
    Until sometime in early December, I had a wonderfully functioning XP-64 system. It was configured to auto download/install MS patches. I occassionally update the software on it, e.g. Open Office, Adobe Reader, Skype, but I don't fetch hundreds of tools or anything much beyond what I just mentioned. In December, suddenly my audio stopped, and drive letters assigned to various mount points on other machines quit being available. Apparantly, the services that support these (and some others) are now not starting up when I boot/login. There isn't anything obvious in the event log. If I manually restart the associated services, these facilities come back on line and work for awhile (a day) but pretty soon the problem reappears. I don't reboot very often, nor do I log out out much. Hints?

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  • No audio driver for ECS 7050-M v1.0a for Windows 7?

    - by NudeRaider
    I have an Elitegroup 7050-M v1.0a motherboard and I've used the Realtek HD audio driver it came with for years under Windows XP. There was a nice mixer panel, an equalizer and sound profiles that added effects. All of these menus are gone now that I've moved on to Windows 7 64 Bit. All I have is the basic settings under system control - sound, but they are not nearly as detailed. To clarify: Sound works, but the advanced settings I got when using older OS's are gone. Older drivers don't seem to work under Win7 and I can't find one for Win7. So how can I fully utilize the capabilities of my HD sound chip under Win7?

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  • No external microphone Acer AO722

    - by Leeghwater
    The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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  • How to remove HTML code from search result page content

    - by Jack Torris
    I have music website. There are 46 album pages and each page has different player and files. I just entered the one of album's URLs in a search engine. I found that Google is displaying player code in search result content. For example, enter this URL in Google and check the results. Each result displays a .mp3 file in content section. I see this: This page contains a demo of and documentation for the new jPlayer Playlist add-on, ... mp3:"http://www.jplayer.org/audio/mp3/Miaow-01-Tempered-song.mp3", ... I don't want Google to show the player code and mp3 files in search result. How can I hide audio files and player code from search engine? What would be the best solution for it?

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  • Steganography : Encoded audio and video file not being played, getting corrupted. What is the issue

    - by Shantanu Gupta
    I have made a steganography program to encrypt/Decrypt some text under image audio and video. I used image as bmp(54 byte header) file, audio as wav(44 byte header) file and video as avi(56 byte header) file formats. When I tries to encrypt text under all these file then it gets encrypted successfully and are also getting decrypted correctly. But it is creating a problem with audio and video i.e these files are not being played after encrypted result. What can be the problem. I am working on Turbo C++ compiler. I know it is super outdated compiler but I have to do it in this only. Here is my code to encrypt. int Binary_encode(char *txtSourceFileName, char *binarySourceFileName, char *binaryTargetFileName,const short headerSize) { long BinarySourceSize=0,TextSourceSize=0; char *Buffer; long BlockSize=10240, i=0; ifstream ReadTxt, ReadBinary; //reads ReadTxt.open(txtSourceFileName,ios::binary|ios::in);//file name, mode of open, here input mode i.e. read only if(!ReadTxt) { cprintf("\nFile can not be opened."); return 0; } ReadBinary.open(binarySourceFileName,ios::binary|ios::in);//file name, mode of open, here input mode i.e. read only if(!ReadBinary) { ReadTxt.close();//closing opened file cprintf("\nFile can not be opened."); return 0; } ReadBinary.seekg(0,ios::end);//setting pointer to a file at the end of file. ReadTxt.seekg(0,ios::end); BinarySourceSize=(long )ReadBinary.tellg(); //returns the position of pointer TextSourceSize=(long )ReadTxt.tellg(); //returns the position of pointer ReadBinary.seekg(0,ios::beg); //sets the pointer to the begining of file ReadTxt.seekg(0,ios::beg); //sets the pointer to the begining of file if(BinarySourceSize<TextSourceSize*50) //Minimum size of an image should be 50 times the size of file to be encrypted { cout<<"\n\n"; cprintf("Binary File size should be bigger than text file size."); ReadBinary.close(); ReadTxt.close(); return 0; } cout<<"\n"; cprintf("\n\nSize of Source Image/Audio File is : "); cout<<(float)BinarySourceSize/1024; cprintf("KB"); cout<<"\n"; cprintf("Size of Text File is "); cout<<TextSourceSize; cprintf(" Bytes"); cout<<"\n"; getch(); //write header to file without changing else file will not open //bmp image's header size is 53 bytes Buffer=new char[headerSize]; ofstream WriteBinary; // writes to file WriteBinary.open(binaryTargetFileName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists ReadBinary.read(Buffer,headerSize);//reads no of bytes and stores them into mem, size contains no of bytes in a file WriteBinary.write(Buffer,headerSize);//writes header to 2nd image delete[] Buffer;//deallocate memory /* Buffer = new char[sizeof(long)]; Buffer = (char *)(&TextSourceSize); cout<<Buffer; */ WriteBinary.write((char *)(&TextSourceSize),sizeof(long)); //writes no of byte to be written in image immediate after header ends //to decrypt file if(!(Buffer=new char[TextSourceSize])) { cprintf("Enough Memory could not be assigned."); return 0; } ReadTxt.read(Buffer,TextSourceSize);//read all data from text file ReadTxt.close();//file no more needed WriteBinary.write(Buffer,TextSourceSize);//writes all text file data into image delete[] Buffer;//deallocate memory //replace Tsize+1 below with Tsize and run the program to see the change //this is due to the reason that 50-54 byte no are of colors which we will be changing ReadBinary.seekg(TextSourceSize+1,ios::cur);//move pointer to the location-current loc i.e. 53+content of text file //write remaining image content to image file while(i<BinarySourceSize-headerSize-TextSourceSize+1) { i=i+BlockSize; Buffer=new char[BlockSize]; ReadBinary.read(Buffer,BlockSize);//reads no of bytes and stores them into mem, size contains no of bytes in a file WriteBinary.write(Buffer,BlockSize); delete[] Buffer; //clear memory, else program can fail giving correct output } ReadBinary.close(); WriteBinary.close(); //Encoding Completed return 0; } Code to decrypt int Binary_decode(char *binarySourceFileName, char *txtTargetFileName, const short headerSize) { long TextDestinationSize=0; char *Buffer; long BlockSize=10240; ifstream ReadBinary; ofstream WriteText; ReadBinary.open(binarySourceFileName,ios::binary|ios::in);//file will be appended if(!ReadBinary) { cprintf("File can not be opened"); return 0; } ReadBinary.seekg(headerSize,ios::beg); Buffer=new char[4]; ReadBinary.read(Buffer,4); TextDestinationSize=*((long *)Buffer); delete[] Buffer; cout<<"\n\n"; cprintf("Size of the File that will be created is : "); cout<<TextDestinationSize; cprintf(" Bytes"); cout<<"\n\n"; sleep(1); WriteText.open(txtTargetFileName,ios::binary|ios::out|ios::trunc);//file will be created if not exists else truncate its data while(TextDestinationSize>0) { if(TextDestinationSize<BlockSize) BlockSize=TextDestinationSize; Buffer= new char[BlockSize]; ReadBinary.read(Buffer,BlockSize); WriteText.write(Buffer,BlockSize); delete[] Buffer; TextDestinationSize=TextDestinationSize-BlockSize; } ReadBinary.close(); WriteText.close(); return 0; } int text_encode(char *SourcefileName, char *DestinationfileName) { ifstream fr; //reads ofstream fw; // writes to file char c; int random; clrscr(); fr.open(SourcefileName,ios::binary);//file name, mode of open, here input mode i.e. read only if(!fr) { cprintf("File can not be opened."); getch(); return 0; } fw.open(DestinationfileName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists while(fr) { int i; while(fr!=0) { fr.get(c); //reads a character from file and increments its pointer char ch; ch=c; ch=ch+1; fw<<ch; //appends character in c to a file } } fr.close(); fw.close(); return 0; } int text_decode(char *SourcefileName, char *DestinationName) { ifstream fr; //reads ofstream fw; // wrrites to file char c; int random; clrscr(); fr.open(SourcefileName,ios::binary);//file name, mode of open, here input mode i.e. read only if(!fr) { cprintf("File can not be opened."); return 0; } fw.open(DestinationName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists while(fr) { int i; while(fr!=0) { fr.get(c); //reads a character from file and increments its pointer char ch; ch=c; ch=ch-1; fw<<ch; //appends character in c to a file } } fr.close(); fw.close(); return 0; }

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  • xvidcap: Error accessing sound input from /dev/dsp

    - by stivlo
    I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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  • HTML5 media loading sometimes suspends or aborts: misconfigured Apache?

    - by Joan Botella
    Recently, some code that has been working fine for months started to run unexpectedly. That code is just a media files loading JavaScript function, that uses jQuery. It's pretty long, but in essence it is like this: var $audio=$('<audio>'); $audio.on('canplaythrough',function(e){ $audio[0].play(); }); $audio.attr('src','song.ogg'); Basically, the file only loads sometimes, and sometimes stops loading with a suspend or even an abort event. I have uploaded a little testing HTML to http://www.joanbotella.com/tests/loading , where you can see what's happening. You can download the test files from http://www.joanbotella.com/tests/loading/loadingTest.zip for local testing. I have just checked that opening the test index.html file directly into Firefox, and not through my localhost Apache server, makes the audio files perfectly playable. So, I assume, my hosting and I have the Apache server misconfigured for serving media files. My software versions are: Apache 2.2.22-1ubuntu1.7 , Mozilla Firefox 31.0 , Chromium 36.0.1985.125 and jQuery 1.11.0. Can you help me? Thanks in advance!

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  • Will proprietary software-based sound enhancements work with Ubuntu? (BeatsAudio, Dolby)

    - by LiveWireBT
    This question is targeted at mainstream or gamer-grade software-based audio/sound enhancements, found in highly integrated computing and entertainment systems like laptops, tablets and smartphones. These are mostly marketed with fancy badges of known audio-releated brands on the product or packaging, while being mostly uncertain about the actual implementation or components used and poorly differentiated from the general audio capabilities of the system or device. This question is not about actual hardware like speakers. If your headphones are not properly detected, your speakers are assigned wrong, work partially or not at all then your soundcard or chip is not properly detected and you should take a look at troubleshooting audio issues. This question is also not about enthusiast or recording-grade hardware like recording interfaces, amplifiers and DACs in a variety of formfactors. And this question is also not about audio encoding and playback of different audio formats like Dolby Digital, Dolby TrueHD and DTS. Most of these may be subject to patents and licensing, see restricted formats. If you are just searching for an equalizer, please take a look at this question: Is there any Sound enhancers/equalizer? Simply speaking: Every feature where you would flip a switch or check a box in a fancy looking interface in Windows that makes the sound change from neutral to fancy.

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  • I can't hear any sounds on ubuntu 11.10 on Dell inspiron N5010

    - by Ahmed
    I have a problem that I can't hear any sounds and I don't know where to start. I did the following : lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at fbf00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel -- 01:00.1 Audio device: ATI Technologies Inc Manhattan HDMI Audio [Mobility Radeon HD 5000 Series] Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at fbe40000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel And It seems that I have 2 soundcards. Is that normal ?? I also did this: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 Also on the sound setting GUI. I have 2 hardware profiles for sound cards but none of them works when I test the speakers. Where should I start searching ?

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  • Increase traffic to site [duplicate]

    - by Jack Trowbridge
    This question already has an answer here: How can I increase the traffic to my site? 5 answers I have made a social networking site and it's been on the web for over 6 months and it has over 8,000 members but I want it to grow bigger. What tools/methods can I use to grow its popularity? e.g CEO, PPC advertising Tools/methods requiring money and without and comparisons? Thanks in advance, Jack.

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  • Writing a Makefile.am to invoke googletest unit tests

    - by jmglov
    I am trying to add my first unit test to an existing Open Source project. Specifically, I added a new class, called audio_manager: src/audio/audio_manager.h src/audio/audio_manager.cc I created a src/test directory structure that mirrors the structure of the implementation files, and wrote my googletest unit tests: src/test/audio/audio_manager.cc Now, I am trying to set up my Makefile.am to compile and run the unit test: src/test/audio/Makefile.am I copied Makefile.am from: src/audio/Makefile.am Does anyone have a simple recipe for me, or is it to the cryptic automake documentation for me? :)

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  • How to remove music/videos DRM protection and convert to Mobile Devices such as iPod, iPhone, PSP, Z

    - by tonywesley
    The music/video files you purchased from online music stores like iTunes, Yahoo Music or Wal-Mart are under DRM protection. So you can't convert them to the formats supported by your own mobile devices such as Nokia phone, Creative Zen palyer, iPod, PSP, Walkman, Zune… You also can't share your purchased music/videos with your friends. The following step by step tutorial is dedicated to instructing music lovers to how to convert your DRM protected music/videos to mobile devices. Method 1: If you only want to remove DRM protection from your protected music, this method will not spend your money. Step 1: Burn your protected music files to CD-R/RW disc to make an audio CD Step 2: Find a free CD Ripper software to convert the audio CD track back to MP3, WAV, WMA, M4A, AAC, RA… Method 2: This guide will show you how to crack drm from protected wmv, wma, m4p, m4v, m4a, aac files and convert to unprotected WMV, MP4, MP3, WMA or any video and audio formats you like, such as AVI, MP4, Flv, MPEG, MOV, 3GP, m4a, aac, wmv, ogg, wav... I have been using Media Converter software, it is the quickest and easiest solution to remove drm from WMV, M4V, M4P, WMA, M4A, AAC, M4B, AA files by quick recording. It gets audio and video stream at the bottom of operating system, so the output quality is lossless and the conversion speed is fast . The process is as follows. Step 1: Download and install the software Step 2: Run the software and click "Add…" button to load WMA or M4A, M4B, AAC, WMV, M4P, M4V, ASF files Step 3: Choose output formats. If you want to convert protected audio files, please select "Convert audio to" list; If you want to convert protected video files, please select "Convert video to" list. Step 4: You can click "Settings" button to custom preference for output files. Click "Settings" button bellow "Convert audio to" list for protected audio files Click "Settings" button bellow "Convert video to" list for protected video files Step 5: Start remove DRM and convert your DRM protected music and videos by click on "Start" button. What is DRM? DRM, which is most commonly found in movies and music files, doesn't mean just basic copy-protection of video, audio and ebooks, but it basically means full protection for digital content, ranging from delivery to end user's ways to use the content. We can remove the Drm from video and audio files legally by quick recording.

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  • ALSA samples capture: cannot open device

    - by Randagio
    I'm quite new to Linux (Lubuntu 12.04 for sake of precision) and ALSA programming at all. I'm trying to write a C program to capture audio from internal PC microphone for processing it. So as first step I google a bit and I found this article for capturing audio samples A tutorial on using the ALSA Audio API but when I compile it and execute it with: ./capture "default" or ./capture "hw:0,0" and all the possible variants on theme it always raises the error: cannot open device hw:0,0 (no such file or directory). So the issue is: what is the name of the mic audio device to pass as parameter to record the audio from mic ? The mic is working ok because the Sound Recorder program records sounds perfectly and I can playback them. The output of the aplay -l is the following : **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 and this is the amixer output (cut) Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 4 [13%] [-40.50dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [-1.50dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] so for aplay it seems I have no recording device, but for amixer I've got the mic, a mic boost and mic stereo as well with all those gorgeous stuffs on their place !!. If so, how could my Sound Recorder record the audio without any problem at all ?!?! For sure I'm giving the wrong device name to the command line for capturing audio but I'm loosing the hope for finding the correct one ! Please help....before I tear my hair out !!!

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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • Mac OS Sound Problem.

    - by Lukas Šalkauskas
    Ok, this is strange problem, and I couldn't find any solution for it yet. This happened today then @work I plugged in external speakers jack, speakers was playing, everything was OK. But after I took jack off, I couldn't hear any sound from mac speakers. It seems like disabled. So now I only can listen music through headphones or external speakers but not mac speakers. Here is a few screen-shoots: Headphones jack in: Top Menu Icon: When I press increase sound button on the keyboard: Sound Settings: Headphones jack out: Top Menu Icon: When I press increase sound button on the keyboard: Sound Settings: If you had similar situation or you know how to solve this, please do not hesitate and help me ;)

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  • Using the Onboard VGA output with a PCIe video card. Both nVidia

    - by sebikul
    I have 2 video cards, one On board, a nVidia 6150SE nForce 430 and a PCIe nVidia GeForce GT 220 1GB DDR2 RAM I have already configured the PCIe card to use the dual monitor feature, using the VGA and HDMI ports, but now I want to add a third monitor, using the On board VGA port I have managed to enable the On board graphics processor, which is taking 400MB of ram, but I cant manage to use it, nvidia-settings does not detect it, like it's not usable (but is there) My questions are the following: How can I manage to get the On board VGA display to work together with the PCIe graphics card? If possible, how can I recover those 400 MB the on board card is taking (even without being used) or how can I get it to use the PCIe card available memory? System Details: Linux 2.6.35-28-generic i686 Ubuntu 10.10 (All updates installed) NVIDIA Driver Version: 260.19.06 (Official) If more info is needed please let me know. Here is the lspci output when the On board card is disabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) And this is when both are enabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) Output of lshw -class display: *-display description: VGA compatible controller product: GT216 [GeForce GT 220] vendor: nVidia Corporation physical id: 0 bus info: pci@0000:02:00.0 version: a2 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:18 memory:df000000-dfffffff memory:c0000000-cfffffff memory:da000000-dbffffff ioport:ef80(size=128) memory:def80000-deffffff *-display description: VGA compatible controller product: C61 [GeForce 6150SE nForce 430] vendor: nVidia Corporation physical id: d bus info: pci@0000:00:0d.0 version: a2 width: 64 bits clock: 66MHz capabilities: pm msi vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:22 memory:dd000000-ddffffff memory:b0000000-bfffffff memory:dc000000-dcffffff memory:deb40000-deb5ffff If what I'm looking for is not possible, please tell me, so I can disable the On board card and recover those 400MB of wasted RAM Thanks for your help!

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  • Why no multiple instances of Firefox on Linux as on Windows?

    - by Jack
    On Windows If I run Firefox as user jack, and then try to start another instance of firefox I will be unable to, as one is already running. If I choose to run firefox as administrator, then I can have two instances of firefox, separate from each other side by side, because they are under different user accounts. This does not seem to be true on Linux. As user jack if I start firefox, like on windows I am unable to start a new instance. If I open a terminal and change to root, set XAUTHORITY to jacks .Xauthority and try to start firefox as root....I get the error that firefox is already running. Why is this? Please don't spare any technical details in your answers....thankyou.

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  • How can I synchronize a text with audio/sound in XNA/XACT?

    - by Omkar
    Hello Geeks, I wanted to display the text while sound is playing at background. In short if there is sound/audio for "What is this", I want to display the text "What is this" in text box synchronously. Is this possible with XNA/XACT? and can I use this in standard C# based WPF or Silverlight applications? Appreciating your help.

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  • How to stream audio from ASP.NET MVC controller when it's still encoding?

    - by kyrisu
    Background I have wave files on my server that I want to stream. Because of the size I want to encode them to mp3. I've tried to use FileStreamResult - but it doesn't work because as soon as program leaves the controller stream is closed and I get - "Cannot access a closed stream" FileContentResult - but it's not a stream and the user would need to wait for encoding to finish Question Is there a way to stream audio from the controller while it's still encoding?

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  • convert decrypted .vobs to .avi with ffmpeg on ubuntu

    - by Arcath
    I have a .vob file that has bee ripped from a dvd, when I watch the .vob its very good quality video and 5.1 english audio but when I use ffmpeg it has rubbish video and mono french audio. That was using this command: ffmpeg -i /samba/ripping/vobs/12161840#2.vob -f avi /samba/ripping/avis/test.avi I've tried a few different variations on that but it never comes back with anything good just bigger files with bad video and incorrect sound. I know the videos good and the correct audio streams exist so how do I select a 5.1 track and get good video? ffmpeg gives the .vob details as: Input #0, mpeg, from '/samba/ripping/vobs/12161840#2.vob': Duration: 00:42:05.56, start: 0.287267, bitrate: 5738 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 8436 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, mono, s16, 192 kb/s Output #0, avi, to '/samba/ripping/avis/test.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.3 -> #0.1

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  • Is it possible to broadcast audio to shoutcast / icecast / other server? from flash player?

    - by Jeffrey
    I am trying to create a flash client that can stream audio to an online radio server. Theoretically a user could enter the server info / login, and then connect and start sending data to the server which could then be broadcasted and listened to by other clients. I don't think this would be very hard, but am unsure about what data formats to use and what is the best server for the job. I'd like to be able to use one of the most popular radio servers like shoutCast. Any ideas? Thanks in advance.

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  • Linux application that bundles multiple incoming audio and video streams into one container file?

    - by StackedCrooked
    I've been assigned to implement a video on-demand service for a local university. Different aspects of the lectures (video, audio, screen cast, white board) will be recorded. During a lecture all these data streams arrive at one Linux server. This server should transcode and bundle all these streams into one container (Matroska) file. My options seem to be: Write a GStreamer application do something with FFMPEG do something with VLC ...? Has anyone done something similar in the past? Can you recommend something? Edit For those interested, here are a few of my findings: Matroska is not a good format for streaming (it's possible, but it's not its primary intent) For Flash streaming you can use MPEG4 If you want to combine different videos into one video where each subvideo occupies a rectangular portion of the total screen, then this GStreamer script is useful (I found it on this blog post). Desktop capture works fine with VLC

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  • VST plugin : using FFT on audio input buffer with arbitrary size, how ?

    - by Led
    I'm getting interested in programming a VST plugin, and I have a basic knowledge of audio dsp's and FFT's. I'd like to use VST.Net, and I'm wondering how to implement an FFT-based effect. The process-code looks like public override void Process(VstAudioBuffer[] inChannels, VstAudioBuffer[] outChannels) If I'm correct, normally the FFT would be applied on the input, some processing would be done on the FFT'd data, and then an inverse-FFT would create the processed soundbuffer. But since the FFT works on a specified buffersize that will most probably be different then the (arbitrary) amount of input/output-samples, how would you handle this ?

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