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  • Events not sent to WPF based ActiveX control (COM interop) when using Reg-Free-COM

    - by embnut
    I have a WPF based ActiveX control (COM interop). I am able to use it correctly by registering the control. When I tried to Reg-Free-COM (using manifest files) the control seems to be activated, but the events (such as mouse click, RequestBringIntoView etc) dont respond. Interestingly, Double click and tab key works. I read in the this article http://blogs.msdn.com/karstenj/archive/2006/10/09/activex-wpf-gadget.aspx that " ... These upsides come with a price: the ActiveX control must be registered in the registry, which requires some kind of installation such as an .msi. The default gadget installation process cannot install ActiveX. The ActiveX control can't be access via reg-free COM. ..." Has anybody had a similar experience? Can anyone explain what is going on? Additional details: When the control is activated after it has been registered it appears as part of the COM client's UI. The control does not receive focus, its elements receive it. When using reg-free-com the control does not load correctly. 1) The control receives focus instead of its sub elements 2) The control has areas that are black instead of the windows default color 3) when I tab in and out of the control or double click it, it's subelements receive focus, the control starts receiving events and the black areas are replaced by the correct color

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  • unable to find a MessageBodyReader

    - by Cristian Boariu
    Hi guys, I have this interface: @Path("inbox") public interface InboxQueryResourceTest { @POST @Path("{membershipExternalId}/query") @Consumes(MediaType.APPLICATION_XML) @Produces("multipart/mixed") public MultipartOutput query(@PathParam("membershipExternalId") final String membershipExternalId, @QueryParam("page") @DefaultValue("0") final int page, @QueryParam("pageSize") @DefaultValue("10") final int pageSize, @QueryParam("sortProperty") final List<String> sortPropertyList, @QueryParam("sortReversed") final List<Boolean> sortReversed, @QueryParam("sortType") final List<String> sortTypeString, final InstanceQuery instanceQuery) throws IOException; } I have implemented the method to return a MultipartOutput. I am posting an xml query from Fiddler and i receive the result without any problem. BUT i have done an integration test for the same interface, i send the same objects and i put the response like: final MultipartOutput multiPartOutput = getClient().query(getUserRestAuth(), 0, 25, null, null, null, instanceQuery); But here, so from integration tests, i receive a strange error: Unable to find a MessageBodyReader of content-type multipart/mixed;boundary="74c5b6b4-e820-452d-abea-4c56ffb514bb" and type class org.jboss.resteasy.plugins.providers.multipart.MultipartOutput Anyone has any ideea why only in integration tests i receive this error? PS: Some of you will say that i do not send application/xml as ContentType but multipart, which of course is false because the objects are annotated with the required @XmlRootElement etc, otherways neither the POST from Fiddler would work.

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  • video calling (center)

    - by rrejc
    We are starting to develop a new application and I'm searching for information/tips/guides on application architecture. Application should: read the data from an external (USB) device send the data to the remote server (through internet) receive the data from the remote server perform a video call with to the calling (support) center receive a video call call from the calling (support) center support touch screens In addition: some of the data should also be visible through the web page. So I was thinking about: On the server side: use the database (probably MS SQL) use ORM (nHibernate) to map the data from the DB to the domain objects create a layer with business logic in C# create a web (WCF) services (for client application) create an asp.net mvc application (for item 7.) to enable data view through the browser On the client side I would use WPF 4 application which will communicate with external device and the wcf services on the server. So far so good. Now the problem begins. I have no idea how to create a video call (outgoing or incoming) part of the application. I believe that there is no problem to communicate with microphone, speaker, camera with WPF/C#. But how to communicate with the call center? What protocol and encoding should be used? I think that I will need to create some kind of server which will: have a list of operators in the calling center and track which operator is occupied and which operator is free have a list of connected end users receive incoming calls from end users and delegate call to free operator delegate calls from calling center to the end user Any info, link, anything on where to start would be much appreciated. Many thanks!

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  • how to continuously send data without blocking?

    - by Donal Rafferty
    I am trying to send rtp audio data from my Android application. I currently can send 1 RTP packet with the code below and I also have another class that extends Thread that listens to and receives RTP packets. My question is how do I continuously send my updated buffer through the packet payload without blocking the receiving thread? public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); Log.d("BUFFERSIZE","Buffer size = " + buffersize); arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); short[] readBuffer = new short[80]; byte[] buffer = new byte[160]; arec.startRecording(); while(arec.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING){ int frames = arec.read(readBuffer, 0, 80); @SuppressWarnings("unused") int lenghtInBytes = codec.encode(readBuffer, 0, buffer, frames); RtpPacket rtpPacket = new RtpPacket(); rtpPacket.setV(2); rtpPacket.setX(0); rtpPacket.setM(0); rtpPacket.setPT(0); rtpPacket.setSSRC(123342345); rtpPacket.setPayload(buffer, 160); try { rtpSession2.sendRtpPacket(rtpPacket); } catch (UnknownHostException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (RtpException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } } } So when I send on one device and receive on another I get decent audio, but when I send and receive on both I get broken sound like its taking turns to send and receive audio. I have a feeling it could be to do with the while loop? it could be looping around in there and not letting anything else run?

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  • Testing Broadcasting and receiving messages

    - by Avik
    Guys am having some difficulty figuring this out: I am trying to test whether the code(in c#) to broadcast a message and receiving the message works: The code to send the datagram(in this case its the hostname) is: public partial class Form1 : Form { String hostName; byte[] hostBuffer = new byte[1024]; public Form1() { InitializeComponent(); StartNotification(); } public void StartNotification() { IPEndPoint notifyIP = new IPEndPoint(IPAddress.Broadcast, 6000); hostName = Dns.GetHostName(); hostBuffer = Encoding.ASCII.GetBytes(hostName); UdpClient newUdpClient = new UdpClient(); newUdpClient.Send(hostBuffer, hostBuffer.Length, notifyIP); } } And the code to receive the datagram is: public partial class Form1 : Form { byte[] receivedNotification = new byte[1024]; String notificationReceived; StringBuilder listBox; UdpClient udpServer; IPEndPoint remoteEndPoint; public Form1() { InitializeComponent(); udpServer = new UdpClient(new IPEndPoint(IPAddress.Any, 1234)); remoteEndPoint=null; startUdpListener1(); } public void startUdpListener1() { receivedNotification = udpServer.Receive(ref remoteEndPoint); notificationReceived = Encoding.ASCII.GetString(receivedNotification); listBox = new StringBuilder(this.listBox1.Text); listBox.AppendLine(notificationReceived); this.listBox1.Items.Add(listBox.ToString()); } } For the reception of the code I have a form that has only a listbox(listBox1). The problem here is that when i execute the code to receive, the program runs but the form isnt visible. However when I comment the function call( startUdpListener1() ), the purpose isnt served but the form is visible. Whats going wrong?

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  • Silverlight Socket Constantly Returns With Empty Buffer

    - by Benny
    I am using Silverlight to interact with a proxy application that I have developed but, without the proxy sending a message to the Silverlight application, it executes the receive completed handler with an empty buffer ('\0's). Is there something I'm doing wrong? It is causing a major memory leak. this._rawBuffer = new Byte[this.BUFFER_SIZE]; SocketAsyncEventArgs receiveArgs = new SocketAsyncEventArgs(); receiveArgs.SetBuffer(_rawBuffer, 0, _rawBuffer.Length); receiveArgs.Completed += new EventHandler<SocketAsyncEventArgs>(ReceiveComplete); this._client.ReceiveAsync(receiveArgs); if (args.SocketError == SocketError.Success && args.LastOperation == SocketAsyncOperation.Receive) { // Read the current bytes from the stream buffer int bytesRecieved = this._client.ReceiveBufferSize; // If there are bytes to process else the connection is lost if (bytesRecieved > 0) { try { //Find out what we just received string messagePart = UTF8Encoding.UTF8.GetString(_rawBuffer, 0, _rawBuffer.GetLength(0)); //Take out any trailing empty characters from the message messagePart = messagePart.Replace('\0'.ToString(), ""); //Concatenate our current message with any leftovers from previous receipts string fullMessage = _theRest + messagePart; int seperator; //While the index of the seperator (LINE_END defined & initiated as private member) while ((seperator = fullMessage.IndexOf((char)Messages.MessageSeperator.Terminator)) > 0) { //Pull out the first message available (up to the seperator index string message = fullMessage.Substring(0, seperator); //Queue up our new message _messageQueue.Enqueue(message); //Take out our line end character fullMessage = fullMessage.Remove(0, seperator + 1); } //Save whatever was NOT a full message to the private variable used to store the rest _theRest = fullMessage; //Empty the queue of messages if there are any while (this._messageQueue.Count > 0) { ... } } catch (Exception e) { throw e; } // Wait for a new message if (this._isClosing != true) Receive(); } } Thanks in advance.

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  • Place JComponent On The Top Of JXLayer

    - by Yan Cheng CHEOK
    Hello, currently, I had successful applying JXLayer on my charting component, to draw a yellow information box on the top of it. final org.jdesktop.jxlayer.JXLayer<ChartPanel> layer = new org.jdesktop.jxlayer.JXLayer<ChartPanel>(this.chartPanel); this.chartLayerUI = new ChartLayerUI<ChartPanel>(this); layer.setUI(this.chartLayerUI); At the same time, I wish to add the following JComponent (DefaultDrawingView) on the top of JXLayer. This JComponent has the ability 1) To receive mouse event to draw figures on itself. Within ChartLayerUI, I add the following code @Override @SuppressWarnings("unchecked") public void installUI(JComponent c) { super.installUI(c); JXLayer<JComponent> l = (JXLayer<JComponent>) c; l.getGlassPane().setLayout(new java.awt.BorderLayout()); // this.view is DefaultDrawingView drawing object. l.getGlassPane().add(this.view, java.awt.BorderLayout.CENTER); } However, after having the above code, I get the following outcome 1) My charting component ChartPanel are being blocked by DefaultDrawingView 2) My charting component no longer able to receive mouse event. What I wish is that A) ChartPanel and DefaultDrawingView able to show up B) ChartPanel and DefaultDrawingView able to receive mouse event Is there other steps I had missed out, or did wrong? Thanks.

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  • Passing ByteArray from flash (as3) to AMFPHP (2.0.1)

    - by Mauro
    i have a problem passing ByteArray from flash (as3) to amfphp to save an image. With old version of amfphp, all worked in the past… now, with new version i have many problem. I'm using version 2.0.1 and the first problem is that i have to do this, for access to my info: function SaveAsJPEG($json) { $string = json_encode($json); $obj = json_decode($string); $compressed = $obj->{'compressed'}; } in the past i wrote only: function SaveAsJPEG($json) { $compressed = $json['compressed']; } Anyway… now i can take all data (if i use " $json['compressed']" i receive an error) but i can't receive my ByteArray data. From flash i write this: var tempObj:Object = new Object(); tempObj["jpgStream "]= createBitStream(myBitmmapData); // return ByteArray tempObj["compressed"] = false; tempObj["dir"] = linkToSave; tempObj["name"] = this.imageName; So.. in my php class i receive all correct info, except "jpgStream" that seems "null". Do you have any idea?

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  • Why do the outputs differ when I run this code using Netbeans 6.8 and Eclipse?

    - by Vimal Basdeo
    When I am running the following codes using Eclipse and Netbeans 6.8. I want to see the available COM ports on my computer. When running in Eclipse it is returning me all available COm ports but when running t in Netbeans, it does not seem to find any ports .. public static void test(){ Enumeration lists=CommPortIdentifier.getPortIdentifiers(); System.out.println(lists.hasMoreElements()); while (lists.hasMoreElements()){ CommPortIdentifier cn=(CommPortIdentifier)lists.nextElement(); if ((CommPortIdentifier.PORT_SERIAL==cn.getPortType())){ System.out.println("Name is serail portzzzz "+cn.getName()+" Owned status "+cn.isCurrentlyOwned()); try{ SerialPort port1=(SerialPort)cn.open("ComControl",800000); port1.setSerialPortParams(9600, SerialPort.DATABITS_8, SerialPort.STOPBITS_1, SerialPort.PARITY_NONE); System.out.println("Before get stream"); OutputStream out=port1.getOutputStream(); InputStream input=port1.getInputStream(); System.out.println("Before write"); out.write("AT".getBytes()); System.out.println("After write"); int sample=0; //while((( sample=input.read())!=-1)){ System.out.println("Before read"); //System.out.println(input.read() + "TEsting "); //} System.out.println("After read"); System.out.println("Receive timeout is "+port1.getReceiveTimeout()); }catch(Exception e){ System.err.println(e.getMessage()); } } else{ System.out.println("Name is parallel portzzzz "+cn.getName()+" Owned status "+cn.isCurrentlyOwned()+cn.getPortType()+" "); } } } Output with Netbeans false Output using Eclipse true Name is serail portzzzz COM1 Owned status false Before get stream Before write After write Before read After read Receive timeout is -1 Name is serail portzzzz COM2 Owned status false Before get stream Before write After write Before read After read Receive timeout is -1 Name is parallel portzzzz LPT1 Owned status false2 Name is parallel portzzzz LPT2 Owned status false2

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  • Packet fragmentation when sending data via SSLStream

    - by Ive
    When using an SSLStream to send a 'large' chunk of data (1 meg) to a (already authenticated) client, the packet fragmentation / dissasembly I'm seeing is FAR greater than when using a normal NetworkStream. Using an async read on the client (i.e. BeginRead()), the ReadCallback is repeatedly called with exactly the same size chunk of data up until the final packet (the remainder of the data). With the data I'm sending (it's a zip file), the segments happen to be 16363 bytes long. Note: My receive buffer is much bigger than this and changing it's size has no effect I understand that SSL encrypts data in chunks no bigger than 18Kb, but since SSL sits on top of TCP, I wouldn't think that the number of SSL chunks would have any relevance to the TCP packet fragmentation? Essentially, the data is taking about 20 times longer to be fully read by the client than with a standard NetworkStream (both on localhost!) What am I missing? EDIT: I'm beginning to suspect that the receive (or send) buffer size of an SSLStream is limited. Even if I use synchronous reads (i.e. SSLStream.Read()), no more data ever becomes available, regardless of how long I wait before attempting to read. This would be the same behavior as if I were to limit the receive buffer to 16363 bytes. Setting the Underlying NetworkStream's SendBufferSize (on the server), and ReceiveBufferSize (on the client) has no effect.

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  • one two-directed tcp socket OR two one-directed? (linux, high volume, low latency)

    - by osgx
    Hello I need to send (interchange) a high volume of data periodically with the lowest possible latency between 2 machines. The network is rather fast (e.g. 1Gbit or even 2G+). Os is linux. Is it be faster with using 1 tcp socket (for send and recv) or with using 2 uni-directed tcp sockets? The test for this task is very like NetPIPE network benchmark - measure latency and bandwidth for sizes from 2^1 up to 2^13 bytes, each size sent and received 3 times at least (in teal task the number of sends is greater. both processes will be sending and receiving, like ping-pong maybe). The benefit of 2 uni-directed connections come from linux: http://lxr.linux.no/linux+v2.6.18/net/ipv4/tcp_input.c#L3847 3847/* 3848 * TCP receive function for the ESTABLISHED state. 3849 * 3850 * It is split into a fast path and a slow path. The fast path is 3851 * disabled when: ... 3859 * - Data is sent in both directions. Fast path only supports pure senders 3860 * or pure receivers (this means either the sequence number or the ack 3861 * value must stay constant) ... 3863 * 3864 * When these conditions are not satisfied it drops into a standard 3865 * receive procedure patterned after RFC793 to handle all cases. 3866 * The first three cases are guaranteed by proper pred_flags setting, 3867 * the rest is checked inline. Fast processing is turned on in 3868 * tcp_data_queue when everything is OK. All other conditions for disabling fast path is false. And only not-unidirected socket stops kernel from fastpath in receive

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  • UDP server doesnt accept calls from outside.

    - by rayman
    Hi, ive implement simple udp server on my Android device.(sdk 1.5) it works fine when i am runnning a local client on the phone sends through it trigger to my server. but when i try to get udp call from an outside server to my phone, it doesnt work. already make sure the outside server isnt blocked by firewall and it's sending the udp trigger to the right port, which my phone is listening to. i used natstat on the phone and checked that the phone is realy listening to the it's local ip and the port ive setted it to. here is my code of the server:(on the device) // server will listen to one client try { Thread udpServerThread = new Thread() { @Override public void run() { try { // Retrieve the ServerName InetAddress serverAddr = InetAddress .getByName("localhost"); Log.d("UDP", "S: Connecting..."); // Create new UDP-Socket socket = new DatagramSocket(SERVERPORT,serverAddr); byte[] buf = new byte[17]; // * Prepare a UDP-Packet that can contain the data we // * want to receive DatagramPacket packet = new DatagramPacket(buf, buf.length); Log.d("UDP", "S: Receiving..."); // wait to Receive the UDP-Packet socket.receive(packet); Log.d("UDP", "S: Received: '" + new String(packet.getData()) + "'"); acceptedMsg=new String(packet.getData()); notifyService(acceptedMsg); Log.d("UDP", "S: Done."); } catch (Exception e) { Log.e("UDP", "S: Error", e); } } }; udpServerThread.start(); } catch (Exception E) { Log.e("r",E.getMessage()) ; } so as i said, when i try it with local client(seperate thread) which sends udp trigger it works fine, but when i take this client implementation and put it on an outside real server, after UDP being sent, the phone doesnt respond to it. any idea? thanks, ray.

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  • one two-directed tcp socket of two one-directed? (linux, high volume, low latency)

    - by osgx
    Hello I need to send (interchange) a high volume of data periodically with the lowest possible latency between 2 machines. The network is rather fast (e.g. 1Gbit or even 2G+). Os is linux. Is it be faster with using 1 tcp socket (for send and recv) or with using 2 uni-directed tcp sockets? The test for this task is very like NetPIPE network benchmark - measure latency and bandwidth for sizes from 2^1 up to 2^13 bytes, each size sent and received 3 times at least (in teal task the number of sends is greater. both processes will be sending and receiving, like ping-pong maybe). The benefit of 2 uni-directed connections come from linux: http://lxr.linux.no/linux+v2.6.18/net/ipv4/tcp_input.c#L3847 3847/* 3848 * TCP receive function for the ESTABLISHED state. 3849 * 3850 * It is split into a fast path and a slow path. The fast path is 3851 * disabled when: ... 3859 * - Data is sent in both directions. Fast path only supports pure senders 3860 * or pure receivers (this means either the sequence number or the ack 3861 * value must stay constant) ... 3863 * 3864 * When these conditions are not satisfied it drops into a standard 3865 * receive procedure patterned after RFC793 to handle all cases. 3866 * The first three cases are guaranteed by proper pred_flags setting, 3867 * the rest is checked inline. Fast processing is turned on in 3868 * tcp_data_queue when everything is OK. All other conditions for disabling fast path is false. And only not-unidirected socket stops kernel from fastpath in receive

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  • Jtable live updating problem

    - by Fishinastorm
    Hi all, I'm trying to display a jtable in a pop up Jframe and am running into some problems. What i'am trying to do is as follows: catch a button action event in the main frame and display a pop up frame with a jtable populated with some data. The problem i have is that the jtable is populated with metadata i receive from a website and if i'm receiving many records, then the jtable is not displayed until all the records(metadata) is received from the website. I would like to change it such that as soon as the button event is detected in the main frame, I display the pop up frame along with the jtable and insert/update rows "as and when i receive the data from the website". In another words, i want to display the table and have the records appearing one at a time rather than displaying the jtable only after i receive all records.Below is how i'm trying to do it (but in vain :( ): ......... //add the table to the popup frame when application is started, but don't display the frame `until button action is //detected` extraInfoFrame.add(tblMetadata); extraInfoFrame.setVisible(false); //handle code for button press; display the popup private void butMetadataActionPerformed(java.awt.event.ActionEvent evt) { extraInfoFrame.pack(); extraInfoFrame.toFront(); //frame.setSize(350, 250); extraInfoFrame.setVisible(true); //retrieve rows data for the table for(int i=0;i<len;i++){ Object[] data=new Object[4];data=getMetadata(); //get model and insert row ((javax.swing.table.DefaultTableModel)tblMetadata.getModel()).insertRow(i,data); //tried something to notify the view abt change in table data ((javax.swing.table.DefaultTableModel)tblMetadata.getModel()).fireTableRowsInserted(0, 0); tblMetadata.revalidate(); tblMetadata.repaint(); } } Have been racking my head to try and figure something out. A sample example would be greatly appreciated.

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  • How can i get a file from remote machine?

    - by programmerist
    How can i get a file from remote computer? i know remote computer ip and 51124 port is open. i need this algorith: 1) Connect 192.xxx.x.xxx ip via 51124 port 2) filename:123456 (i want to search it on remote machine) 3) Get File 4) Save C:\ 51124 port is open. can i access and can i search any file according to filename? My code is below: IPEndPoint ipEnd = new IPEndPoint(IPAddress.Any, 51124); Socket sock = new Socket(AddressFamily.InterNetwork, SocketType.Stream, ProtocolType.IP); sock.Bind(ipEnd); sock.Listen(maxConnections); Socket serverSocket = sock.Accept(); byte[] data = new byte[bufferSize]; int received = serverSocket.Receive(data); int filenameLength = BitConverter.ToInt32(data, 0); string filename = Encoding.ASCII.GetString(data, 4, filenameLength); BinaryWriter bWrite = new BinaryWriter(File.Open(outPath + filename, FileMode.Create)); bWrite.Write(data, filenameLength + 4, received - filenameLength - 4); int received2 = serverSocket.Receive(data); while (received2 0) { bWrite.Write(data, 0, received2); received2 = serverSocket.Receive(data); } bWrite.Close(); serverSocket.Close(); sock.Close();

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  • programs hangs during socket interaction

    - by herrturtur
    I have two programs, sendfile.py and recvfile.py that are supposed to interact to send a file across the network. They communicate over TCP sockets. The communication is supposed to go something like this: sender =====filename=====> receiver sender <===== 'ok' ======= receiver or sender <===== 'no' ======= receiver if ok: sender ====== file ======> receiver I've got The sender and receiver code is here: Sender: import sys from jmm_sockets import * if len(sys.argv) != 4: print "Usage:", sys.argv[0], "<host> <port> <filename>" sys.exit(1) s = getClientSocket(sys.argv[1], int(sys.argv[2])) try: f = open(sys.argv[3]) except IOError, msg: print "couldn't open file" sys.exit(1) # send filename s.send(sys.argv[3]) # receive 'ok' buffer = None response = str() while 1: buffer = s.recv(1) if buffer == '': break else: response = response + buffer if response == 'ok': print 'receiver acknowledged receipt of filename' # send file s.send(f.read()) elif response == 'no': print "receiver doesn't want the file" # cleanup f.close() s.close() Receiver: from jmm_sockets import * s = getServerSocket(None, 16001) conn, addr = s.accept() buffer = None filename = str() # receive filename while 1: buffer = conn.recv(1) if buffer == '': break else: filename = filename + buffer print "sender wants to send", filename, "is that ok?" user_choice = raw_input("ok/no: ") if user_choice == 'ok': # send ok conn.send('ok') #receive file data = str() while 1: buffer = conn.recv(1) if buffer=='': break else: data = data + buffer print data else: conn.send('no') conn.close() I'm sure I'm missing something here in the sorts of a deadlock, but don't know what it is.

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  • Error while closing SQL Connection

    - by Wickedman84
    I have a problem with closing the SQLconnection in my application. My application is in VB.net. I have a reference in my application to a class with code to open and close the database connection and to execute all sql scripts. The error occurs when i close my application. In the formClosing event of my main form I call a function that closes all the connections. But just before I close the connections I perform an SQLquery to delete a row from a table with the function below. Public Function DeleteFunction(ByVal mySQLQuery As String, ByVal cmd As SqlCommand) As Boolean Try cmd.Connection = myConnection cmd.CommandText = mySQLQuery cmd.ExecuteNonQuery() Return True Catch ex As Exception WriteErrorMessage("DeleteFunction", ex, Logpath, "SQL Error: " & mySQLQuery) Return False End Try End Function In my application I check the result of the boolean. If it returns True, then i call the function to close the database connection. The returned boolean is True and the requested row is deleted in my database. This means i can close my connection which I do with the function below. Public Sub DatabaseConnClose() myCommand.CommandText = "" myConnection.Close() myCommand = Nothing myConnection = Nothing End Sub After executing this code I receive an error in my logfile from the DeleteFunction. It says: "Connection property has not been initialized." It seems very strange to receive an error from a function that was completely executed, or am i wrong to think that? Can anyone tell me why I receive this error and how I can solve the problem?

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  • c# How to get string from byte?

    - by Kade
    I have a form-console application which does a TCP socket connections for send and receive. I need help getting the following response to STRING. The following code does write the RESPONSE to the console, but i also want to byte[] b = new byte[100]; int k = s.Receive(b); Console.WriteLine("Recieved..."); for (int i = 0; i < k; i++) Console.Write(Convert.ToChar(b[i])); ASCIIEncoding asen = new ASCIIEncoding(); s.Send(asen.GetBytes("RECEIVED :")); i want to get something like String GETSTRING; byte[] b = new byte[100]; int k = s.Receive(b); Console.WriteLine("Recieved..."); for (int i = 0; i < k; i++) Console.Write(Convert.ToChar(b[i])); GETSTRING = *WHATEVER RESPONSE RECEIVED ABOVE* ASCIIEncoding asen = new ASCIIEncoding(); s.Send(asen.GetBytes("RECEIVED :"));

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  • Scaling-out Your Services by Message Bus based WCF Transport Extension &ndash; Part 1 &ndash; Background

    - by Shaun
    Cloud computing gives us more flexibility on the computing resource, we can provision and deploy an application or service with multiple instances over multiple machines. With the increment of the service instances, how to balance the incoming message and workload would become a new challenge. Currently there are two approaches we can use to pass the incoming messages to the service instances, I would like call them dispatcher mode and pulling mode.   Dispatcher Mode The dispatcher mode introduces a role which takes the responsible to find the best service instance to process the request. The image below describes the sharp of this mode. There are four clients communicate with the service through the underlying transportation. For example, if we are using HTTP the clients might be connecting to the same service URL. On the server side there’s a dispatcher listening on this URL and try to retrieve all messages. When a message came in, the dispatcher will find a proper service instance to process it. There are three mechanism to find the instance: Round-robin: Dispatcher will always send the message to the next instance. For example, if the dispatcher sent the message to instance 2, then the next message will be sent to instance 3, regardless if instance 3 is busy or not at that moment. Random: Dispatcher will find a service instance randomly, and same as the round-robin mode it regardless if the instance is busy or not. Sticky: Dispatcher will send all related messages to the same service instance. This approach always being used if the service methods are state-ful or session-ful. But as you can see, all of these approaches are not really load balanced. The clients will send messages at any time, and each message might take different process duration on the server side. This means in some cases, some of the service instances are very busy while others are almost idle. For example, if we were using round-robin mode, it could be happened that most of the simple task messages were passed to instance 1 while the complex ones were sent to instance 3, even though instance 1 should be idle. This brings some problem in our architecture. The first one is that, the response to the clients might be longer than it should be. As it’s shown in the figure above, message 6 and 9 can be processed by instance 1 or instance 2, but in reality they were dispatched to the busy instance 3 since the dispatcher and round-robin mode. Secondly, if there are many requests came from the clients in a very short period, service instances might be filled by tons of pending tasks and some instances might be crashed. Third, if we are using some cloud platform to host our service instances, for example the Windows Azure, the computing resource is billed by service deployment period instead of the actual CPU usage. This means if any service instance is idle it is wasting our money! Last one, the dispatcher would be the bottleneck of our system since all incoming messages must be routed by the dispatcher. If we are using HTTP or TCP as the transport, the dispatcher would be a network load balance. If we wants more capacity, we have to scale-up, or buy a hardware load balance which is very expensive, as well as scaling-out the service instances. Pulling Mode Pulling mode doesn’t need a dispatcher to route the messages. All service instances are listening to the same transport and try to retrieve the next proper message to process if they are idle. Since there is no dispatcher in pulling mode, it requires some features on the transportation. The transportation must support multiple client connection and server listening. HTTP and TCP doesn’t allow multiple clients are listening on the same address and port, so it cannot be used in pulling mode directly. All messages in the transportation must be FIFO, which means the old message must be received before the new one. Message selection would be a plus on the transportation. This means both service and client can specify some selection criteria and just receive some specified kinds of messages. This feature is not mandatory but would be very useful when implementing the request reply and duplex WCF channel modes. Otherwise we must have a memory dictionary to store the reply messages. I will explain more about this in the following articles. Message bus, or the message queue would be best candidate as the transportation when using the pulling mode. First, it allows multiple application to listen on the same queue, and it’s FIFO. Some of the message bus also support the message selection, such as TIBCO EMS, RabbitMQ. Some others provide in memory dictionary which can store the reply messages, for example the Redis. The principle of pulling mode is to let the service instances self-managed. This means each instance will try to retrieve the next pending incoming message if they finished the current task. This gives us more benefit and can solve the problems we met with in the dispatcher mode. The incoming message will be received to the best instance to process, which means this will be very balanced. And it will not happen that some instances are busy while other are idle, since the idle one will retrieve more tasks to make them busy. Since all instances are try their best to be busy we can use less instances than dispatcher mode, which more cost effective. Since there’s no dispatcher in the system, there is no bottleneck. When we introduced more service instances, in dispatcher mode we have to change something to let the dispatcher know the new instances. But in pulling mode since all service instance are self-managed, there no extra change at all. If there are many incoming messages, since the message bus can queue them in the transportation, service instances would not be crashed. All above are the benefits using the pulling mode, but it will introduce some problem as well. The process tracking and debugging become more difficult. Since the service instances are self-managed, we cannot know which instance will process the message. So we need more information to support debug and track. Real-time response may not be supported. All service instances will process the next message after the current one has done, if we have some real-time request this may not be a good solution. Compare with the Pros and Cons above, the pulling mode would a better solution for the distributed system architecture. Because what we need more is the scalability, cost-effect and the self-management.   WCF and WCF Transport Extensibility Windows Communication Foundation (WCF) is a framework for building service-oriented applications. In the .NET world WCF is the best way to implement the service. In this series I’m going to demonstrate how to implement the pulling mode on top of a message bus by extending the WCF. I don’t want to deep into every related field in WCF but will highlight its transport extensibility. When we implemented an RPC foundation there are many aspects we need to deal with, for example the message encoding, encryption, authentication and message sending and receiving. In WCF, each aspect is represented by a channel. A message will be passed through all necessary channels and finally send to the underlying transportation. And on the other side the message will be received from the transport and though the same channels until the business logic. This mode is called “Channel Stack” in WCF, and the last channel in the channel stack must always be a transport channel, which takes the responsible for sending and receiving the messages. As we are going to implement the WCF over message bus and implement the pulling mode scaling-out solution, we need to create our own transport channel so that the client and service can exchange messages over our bus. Before we deep into the transport channel, let’s have a look on the message exchange patterns that WCF defines. Message exchange pattern (MEP) defines how client and service exchange the messages over the transportation. WCF defines 3 basic MEPs which are datagram, Request-Reply and Duplex. Datagram: Also known as one-way, or fire-forgot mode. The message sent from the client to the service, and no need any reply from the service. The client doesn’t care about the message result at all. Request-Reply: Very common used pattern. The client send the request message to the service and wait until the reply message comes from the service. Duplex: The client sent message to the service, when the service processing the message it can callback to the client. When callback the service would be like a client while the client would be like a service. In WCF, each MEP represent some channels associated. MEP Channels Datagram IInputChannel, IOutputChannel Request-Reply IRequestChannel, IReplyChannel Duplex IDuplexChannel And the channels are created by ChannelListener on the server side, and ChannelFactory on the client side. The ChannelListener and ChannelFactory are created by the TransportBindingElement. The TransportBindingElement is created by the Binding, which can be defined as a new binding or from a custom binding. For more information about the transport channel mode, please refer to the MSDN document. The figure below shows the transport channel objects when using the request-reply MEP. And this is the datagram MEP. And this is the duplex MEP. After investigated the WCF transport architecture, channel mode and MEP, we finally identified what we should do to extend our message bus based transport layer. They are: Binding: (Optional) Defines the channel elements in the channel stack and added our transport binding element at the bottom of the stack. But we can use the build-in CustomBinding as well. TransportBindingElement: Defines which MEP is supported in our transport and create the related ChannelListener and ChannelFactory. This also defines the scheme of the endpoint if using this transport. ChannelListener: Create the server side channel based on the MEP it’s. We can have one ChannelListener to create channels for all supported MEPs, or we can have ChannelListener for each MEP. In this series I will use the second approach. ChannelFactory: Create the client side channel based on the MEP it’s. We can have one ChannelFactory to create channels for all supported MEPs, or we can have ChannelFactory for each MEP. In this series I will use the second approach. Channels: Based on the MEPs we want to support, we need to implement the channels accordingly. For example, if we want our transport support Request-Reply mode we should implement IRequestChannel and IReplyChannel. In this series I will implement all 3 MEPs listed above one by one. Scaffold: In order to make our transport extension works we also need to implement some scaffold stuff. For example we need some classes to send and receive message though out message bus. We also need some codes to read and write the WCF message, etc.. These are not necessary but would be very useful in our example.   Message Bus There is only one thing remained before we can begin to implement our scaling-out support WCF transport, which is the message bus. As I mentioned above, the message bus must have some features to fulfill all the WCF MEPs. In my company we will be using TIBCO EMS, which is an enterprise message bus product. And I have said before we can use any message bus production if it’s satisfied with our requests. Here I would like to introduce an interface to separate the message bus from the WCF. This allows us to implement the bus operations by any kinds bus we are going to use. The interface would be like this. 1: public interface IBus : IDisposable 2: { 3: string SendRequest(string message, bool fromClient, string from, string to = null); 4:  5: void SendReply(string message, bool fromClient, string replyTo); 6:  7: BusMessage Receive(bool fromClient, string replyTo); 8: } There are only three methods for the bus interface. Let me explain one by one. The SendRequest method takes the responsible for sending the request message into the bus. The parameters description are: message: The WCF message content. fromClient: Indicates if this message was came from the client. from: The channel ID that this message was sent from. The channel ID will be generated when any kinds of channel was created, which will be explained in the following articles. to: The channel ID that this message should be received. In Request-Reply and Duplex MEP this is necessary since the reply message must be received by the channel which sent the related request message. The SendReply method takes the responsible for sending the reply message. It’s very similar as the previous one but no “from” parameter. This is because it’s no need to reply a reply message again in any MEPs. The Receive method takes the responsible for waiting for a incoming message, includes the request message and specified reply message. It returned a BusMessage object, which contains some information about the channel information. The code of the BusMessage class is 1: public class BusMessage 2: { 3: public string MessageID { get; private set; } 4: public string From { get; private set; } 5: public string ReplyTo { get; private set; } 6: public string Content { get; private set; } 7:  8: public BusMessage(string messageId, string fromChannelId, string replyToChannelId, string content) 9: { 10: MessageID = messageId; 11: From = fromChannelId; 12: ReplyTo = replyToChannelId; 13: Content = content; 14: } 15: } Now let’s implement a message bus based on the IBus interface. Since I don’t want you to buy and install the TIBCO EMS or any other message bus products, I will implement an in process memory bus. This bus is only for test and sample purpose. It can only be used if the service and client are in the same process. Very straightforward. 1: public class InProcMessageBus : IBus 2: { 3: private readonly ConcurrentDictionary<Guid, InProcMessageEntity> _queue; 4: private readonly object _lock; 5:  6: public InProcMessageBus() 7: { 8: _queue = new ConcurrentDictionary<Guid, InProcMessageEntity>(); 9: _lock = new object(); 10: } 11:  12: public string SendRequest(string message, bool fromClient, string from, string to = null) 13: { 14: var entity = new InProcMessageEntity(message, fromClient, from, to); 15: _queue.TryAdd(entity.ID, entity); 16: return entity.ID.ToString(); 17: } 18:  19: public void SendReply(string message, bool fromClient, string replyTo) 20: { 21: var entity = new InProcMessageEntity(message, fromClient, null, replyTo); 22: _queue.TryAdd(entity.ID, entity); 23: } 24:  25: public BusMessage Receive(bool fromClient, string replyTo) 26: { 27: InProcMessageEntity e = null; 28: while (true) 29: { 30: lock (_lock) 31: { 32: var entity = _queue 33: .Where(kvp => kvp.Value.FromClient == fromClient && (kvp.Value.To == replyTo || string.IsNullOrWhiteSpace(kvp.Value.To))) 34: .FirstOrDefault(); 35: if (entity.Key != Guid.Empty && entity.Value != null) 36: { 37: _queue.TryRemove(entity.Key, out e); 38: } 39: } 40: if (e == null) 41: { 42: Thread.Sleep(100); 43: } 44: else 45: { 46: return new BusMessage(e.ID.ToString(), e.From, e.To, e.Content); 47: } 48: } 49: } 50:  51: public void Dispose() 52: { 53: } 54: } The InProcMessageBus stores the messages in the objects of InProcMessageEntity, which can take some extra information beside the WCF message itself. 1: public class InProcMessageEntity 2: { 3: public Guid ID { get; set; } 4: public string Content { get; set; } 5: public bool FromClient { get; set; } 6: public string From { get; set; } 7: public string To { get; set; } 8:  9: public InProcMessageEntity() 10: : this(string.Empty, false, string.Empty, string.Empty) 11: { 12: } 13:  14: public InProcMessageEntity(string content, bool fromClient, string from, string to) 15: { 16: ID = Guid.NewGuid(); 17: Content = content; 18: FromClient = fromClient; 19: From = from; 20: To = to; 21: } 22: }   Summary OK, now I have all necessary stuff ready. The next step would be implementing our WCF message bus transport extension. In this post I described two scaling-out approaches on the service side especially if we are using the cloud platform: dispatcher mode and pulling mode. And I compared the Pros and Cons of them. Then I introduced the WCF channel stack, channel mode and the transport extension part, and identified what we should do to create our own WCF transport extension, to let our WCF services using pulling mode based on a message bus. And finally I provided some classes that need to be used in the future posts that working against an in process memory message bus, for the demonstration purpose only. In the next post I will begin to implement the transport extension step by step.   Hope this helps, Shaun All documents and related graphics, codes are provided "AS IS" without warranty of any kind. Copyright © Shaun Ziyan Xu. This work is licensed under the Creative Commons License.

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  • Sonicwall SSL VPN Login : I need help with a NetExtender initialization error.

    - by jacke672
    I receive the error message: "Server is busy now, please try it later!" after logging into our Sonicwall successfully and attempting to initialize NetExtender for the "virtual office" function. It was set up yesterday and I am able to log in without any issues, but I keep getting hung up on the installation and/or initialization of NetExtender. I have attempted to connect remotely on XP and 7 using both FireFox and IE. I am using a Sonicwall NSA-240 with load balancing active (1 ISP and 2 different connections)- I have tried turning off load balancing and disabling the secondary connection but still receive the same error. I've been in contact with SonicWall support but I haven't heard from them as of yet so I'm asking the Server Fault community in the meantime... Does anyone have any ideas as per what could be the issue? Thanks -Jack

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  • Send email from server to Google Apps email address (same domains)

    - by Orlando
    I'm sending email from a server, let's say domain.com. I also have Google Apps email set up for hosted email, same domain, domain.com. If I get mail sent to me from anywhere else, I receive things just fine. However, if the email originates from my server, it just ends up in /var/mail/root as a delivery error saying the user is unknown. I created a user on the server for the name which is having trouble, [email protected]. Retried sending and it sends, but not to my hosted email at Google Apps. I just receive it at /var/mail/webmaster now. I'm using sendmail. I messed around with /etc/aliases but adding webmaster: [email protected] looked useless (and I was right.) Any help?

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  • How to start networking on a wired interface before logon in Ubuntu Desktop Edition

    - by Burly
    Problem Ubuntu 9.10 Desktop Edition (and possibly previous versions as well, I haven't tested them) has no network connections after boot until at least 1 user logs in. This means any services that require networking (e.g. openssh-server) are not available until someone logs in locally either via gdm, kdm, or a TTY. Background Ubuntu 9.10 Desktop Edition uses the NetworkManager service to take commands from the nm-applet in Gnome (or it's equivalent in KDE). As I understand it, while NetworkManager is running at boot, it is not issued any commands to connect until you login for the first time because nm-applet isn't running until you login and your Gnome session starts (or similar for KDE). I'm not sure what prompts NetworkManager to connect to the network when you login via a TTY. There are several relevant variables involved in starting up the network connections including: Wired vs Wireless (and the resulting drivers, SSID, passwords, and priorities) Static vs DHCP Multiple interfaces Constraints Support Ubuntu 9.10 Karmic Koala (bonus points for additional supported versions) Support wired eth0 interface Receive an IP address via DHCP Receive DNS information via DHCP (obviously the DHCP server must provide this information) Enable networking at the proper time (e.g. some time after file systems are loaded but before network services like ssh start) Switching distros or versions (e.g. to Server Edition) is not an acceptable solution Switching to a Static IP configuration is not an acceptable solution Question How to start networking on a wired interface before logon in Ubuntu Desktop Edition? What I have tried Per this guide, adding the following entry into /etc/network/interfaces so that NetworkManager won't manage the eth0 interface: auth eth0 iface inet dhcp After reboot eth0 is down. Issuing ifconfig eth0 up brings the interface up but it receives no IP address. Issuing dhclient eth0 instead Does bring up the interface and it Does receive an IP address. Completely removing the NetworkManager package in addition to the settings above. I'm a bit confused with the whole UpStart/SysVinit mangling that's going in Ubuntu currently (I'm more familiar with the CentOS world). However, directly issuing sudo /etc/init.d/networking start Or sudo start networking does not bring up the eth0 interface at all, much less get an IP address. See-Also How to force NetworkManager to make a connection before login? References Ubuntu Desktop Edition Ubuntu Networking Configuration Using Command Line Automatic Network Configuration Via Command-Line Start network connection before login

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  • UDP multicast streaming of media content over WIFI

    - by sajad
    I am using vlc to stream media content over wireless network in scenario like this (from content streamer to stream receiver client): The bandwidth of wireless network is 54 Mb/s and UDP stream's required bandwidth is only 4 Mb/s; however there is trouble in receiving media stream and quality of playing specifically in multicast mode; means I can play the stream but it has jitter and does not play smoothly. In uni-cast I can stream up to 5 media streams correctly, but in multicast mode there is problem with streaming just one media! However when I stream from client some multicast streams; the wifi access-point can receive data correctly and I can see the video in "udp streamer" side correctly even when number of multicast streams increases to 9; But as you see I want to stream from streaming server and receive media in client size. Is this a typical problem of streaming real-time contents over wireless networks? Is it necessary to change configurations of my WIFI switch or it is just a software trouble? thank you

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  • diagnostic multicast issue using wireshark

    - by Abruzzo Forte e Gentile
    I have a network that is setup for multicast traffic. My setup is the following -Machine A : a server generates multicast traffic. -Machine A : few clients subscribing to that multicast traffic -Machine B : few clients subscribing to that multicast traffic # Address I am using IP : 239.193.0.21 PORT: 20401 The clients in machine A , even if they join the group (I can see IGMP messages through wireshark), don't receive any data while (and this is the funny part) machine B,C and D receive everything. I sorted that issue by completely disabling Linux firewall. Before doing that, I enabled the multicast on the firwall ('reject all'). iptables -A INPUT -m addrtype --src-type MULTICAST -j ACCEPT My question is the following: what I can check in wireshark that can help me in spot such firewall issues in the futures? For TCP/IP I realize by using ping and looking at ICMP packets rejected. What I can check/monitor for multicast? I am using LInux/Red-Hat Enterprise 6.2

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  • SonicOS Enhanced 5.8.1.2 L2TP VPN Authentication Failed

    - by Dean A. Vassallo
    I have a SonicWall TZ 215 running SonicOS Enhanced 5.8.1.2-6o. I have configured the L2TP VPN using the default crypto suite ESP: 3DES/HMAC SHA1 (IKE). Proposals are as such: IKE (Phase 1) Proposal DH Group: Group 2 Encryption: 3DES Authentication: SHA1 Life Time (seconds): 28800 Ipsec (Phase 2) Proposal Protocol: ESP Encryption: 3DES Authentication: SHA1 Enable Perfect Forward Secrecy DISABLED Life Time (seconds): 28800 When attempting to connect via my Mac OS X client I get an authentication error. It appears to pass the pre-authentication but fails to complete. I am at a complete loss. I reconfigured from scratch multiple times...used simple usernames and passwords to verify this wasn't a miskeyed password issue. I have Here are the logs (noted IP has been removed for privacy): 7/1/13 8:19:05.174 PM pppd[1268]: setup_security_context server port: 0x1503 7/1/13 8:19:05.190 PM pppd[1268]: publish_entry SCDSet() failed: Success! 7/1/13 8:19:05.191 PM pppd[1268]: publish_entry SCDSet() failed: Success! 7/1/13 8:19:05.191 PM pppd[1268]: pppd 2.4.2 (Apple version 727.1.1) started by dean, uid 501 7/1/13 8:19:05.192 PM pppd[1268]: L2TP connecting to server ‘0.0.0.0’ (0.0.0.0)... 7/1/13 8:19:05.193 PM pppd[1268]: IPSec connection started 7/1/13 8:19:05.208 PM racoon[1269]: accepted connection on vpn control socket. 7/1/13 8:19:05.209 PM racoon[1269]: Connecting. 7/1/13 8:19:05.209 PM racoon[1269]: IPSec Phase 1 started (Initiated by me). 7/1/13 8:19:05.209 PM racoon[1269]: IKE Packet: transmit success. (Initiator, Main-Mode message 1). 7/1/13 8:19:05.209 PM racoon[1269]: >>>>> phase change status = Phase 1 started by us 7/1/13 8:19:05.231 PM racoon[1269]: >>>>> phase change status = Phase 1 started by peer 7/1/13 8:19:05.231 PM racoon[1269]: IKE Packet: receive success. (Initiator, Main-Mode message 2). 7/1/13 8:19:05.234 PM racoon[1269]: IKE Packet: transmit success. (Initiator, Main-Mode message 3). 7/1/13 8:19:05.293 PM racoon[1269]: IKE Packet: receive success. (Initiator, Main-Mode message 4). 7/1/13 8:19:05.295 PM racoon[1269]: IKE Packet: transmit success. (Initiator, Main-Mode message 5). 7/1/13 8:19:05.315 PM racoon[1269]: IKEv1 Phase 1 AUTH: success. (Initiator, Main-Mode Message 6). 7/1/13 8:19:05.315 PM racoon[1269]: IKE Packet: receive success. (Initiator, Main-Mode message 6). 7/1/13 8:19:05.315 PM racoon[1269]: IKEv1 Phase 1 Initiator: success. (Initiator, Main-Mode). 7/1/13 8:19:05.315 PM racoon[1269]: IPSec Phase 1 established (Initiated by me). 7/1/13 8:19:06.307 PM racoon[1269]: IPSec Phase 2 started (Initiated by me). 7/1/13 8:19:06.307 PM racoon[1269]: >>>>> phase change status = Phase 2 started 7/1/13 8:19:06.308 PM racoon[1269]: IKE Packet: transmit success. (Initiator, Quick-Mode message 1). 7/1/13 8:19:06.332 PM racoon[1269]: attribute has been modified. 7/1/13 8:19:06.332 PM racoon[1269]: IKE Packet: receive success. (Initiator, Quick-Mode message 2). 7/1/13 8:19:06.332 PM racoon[1269]: IKE Packet: transmit success. (Initiator, Quick-Mode message 3). 7/1/13 8:19:06.333 PM racoon[1269]: IKEv1 Phase 2 Initiator: success. (Initiator, Quick-Mode). 7/1/13 8:19:06.333 PM racoon[1269]: IPSec Phase 2 established (Initiated by me). 7/1/13 8:19:06.333 PM racoon[1269]: >>>>> phase change status = Phase 2 established 7/1/13 8:19:06.333 PM pppd[1268]: IPSec connection established 7/1/13 8:19:07.145 PM pppd[1268]: L2TP connection established. 7/1/13 8:19:07.000 PM kernel[0]: ppp0: is now delegating en0 (type 0x6, family 2, sub-family 3) 7/1/13 8:19:07.146 PM pppd[1268]: Connect: ppp0 <--> socket[34:18] 7/1/13 8:19:08.709 PM pppd[1268]: MS-CHAPv2 mutual authentication failed. 7/1/13 8:19:08.710 PM pppd[1268]: Connection terminated. 7/1/13 8:19:08.710 PM pppd[1268]: L2TP disconnecting... 7/1/13 8:19:08.711 PM pppd[1268]: L2TP disconnected 7/1/13 8:19:08.711 PM racoon[1269]: IPSec disconnecting from server 0.0.0.0 7/1/13 8:19:08.711 PM racoon[1269]: IKE Packet: transmit success. (Information message). 7/1/13 8:19:08.712 PM racoon[1269]: IKEv1 Information-Notice: transmit success. (Delete IPSEC-SA). 7/1/13 8:19:08.712 PM racoon[1269]: IKE Packet: transmit success. (Information message). 7/1/13 8:19:08.712 PM racoon[1269]: IKEv1 Information-Notice: transmit success. (Delete ISAKMP-SA). 7/1/13 8:19:08.713 PM racoon[1269]: glob found no matches for path "/var/run/racoon/*.conf" 7/1/13 8:19:08.714 PM racoon[1269]: pfkey DELETE failed: No such file or directory

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