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  • JS window.close() let IE hang

    - by p4bl0.666
    Hi all, for reason I won't bore you with, I'm writing an asp.net application that must open some pages in new browser windows. I managed to open them within a postback (don't ask why, I just needed to) with this code: script = String.Format(@"window.open(""{0}"", ""{1}"");", url, target); ScriptManager.RegisterStartupScript(page, typeof(Page), "Redirect", script, true); Now I have new windows each one with a button that should close it. I have simply an onclick="window.close()" (but that prompts me when I'm closing the browser) or window.open('','_self','');window.close() (horrible, I agree but it's the only way I found to avoid the JS prompt) On firefox it works perfectly but on IE7 (the browser our customers have) after 2-3 times I use that button to close the window I can't open other windows (in both cases, with or without the JS prompt). With the method above it does nothing, and with a <a href="mypage.aspx" target="_blank">click me</a> a new window is opened but hangs on loading (it doesn't even calls the Page_Load). What could be the cause? How can I solve this? Thank you.

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  • GDI+ not clearing my window on repaint for vista

    - by Michael
    on WM_PAINT i do the following: //RectF mNameRect; //WCHAR* mName; //HWND mWin; // this is the window handle { PAINTSTRUCT ps; HDC hdc = BeginPaint(mWin, &ps); Graphics g(hdc); g.Clear(Color::White); StringFormat stringForm; stringForm.SetLineAlignment(StringAlignmentCenter); stringForm.SetAlignment(StringAlignmentCenter); // set the rectangle to the size of the whole window mNameRect.Width = static_cast<float>(size.cx); mNameRect.Height = static_cast<float>(size.cy); g.DrawString(mName, -1, &mNameFont, mNameRect, &stringForm, &mNameBrush); EndPaint(mWin, &ps); } In XP this works fine, the mName is displayed in the middle of the window. However on Vista the text doesn't move, it stays in its location no matter how I resize the window. the g.Clear(Color::White) doesn't seem to do any difference. The text doesn't even change position when the window is hidden behind another window and on focus needs to be repainted again. How do I make mName change position in Vista?

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  • Cannot start tor with vidalia, failed to bind listening port because of tor-socks running

    - by ganjan
    I get these errors trying to run tor with vidalia Apr 19 21:55:15.371 [Notice] Tor v0.2.1.30. This is experimental software. Do not rely on it for strong anonymity. (Running on Linux i686) Apr 19 21:55:15.372 [Notice] Initialized libevent version 1.4.13-stable using method epoll. Good. Apr 19 21:55:15.373 [Notice] Opening Socks listener on 127.0.0.1:9050 Apr 19 21:55:15.373 [Warning] Could not bind to 127.0.0.1:9050: Address already in use. Is Tor already running? Apr 19 21:55:15.373 [Warning] Failed to parse/validate config: Failed to bind one of the listener ports. Apr 19 21:55:15.373 [Error] Reading config failed--see warnings above. I don't think tor is running. Here is a nmap scan of my localhost Starting Nmap 5.21 ( http://nmap.org ) at 2011-04-19 21:59 CEST Nmap scan report for localhost (127.0.0.1) Host is up (0.0000050s latency). Hostname localhost resolves to 2 IPs. Only scanned 127.0.0.1 rDNS record for 127.0.0.1: localhost.localdomain Not shown: 989 closed ports PORT STATE SERVICE 22/tcp open ssh 53/tcp open domain 80/tcp open http 139/tcp open netbios-ssn 445/tcp open microsoft-ds 631/tcp open ipp 3128/tcp open squid-http 3306/tcp open mysql 9000/tcp open cslistener 9050/tcp open tor-socks 10000/tcp open snet-sensor-mgmt I see tor-socks is running here, probably be the cause of the problem. How do I stop this from starting up? I want to use vidalia so I can monitor whats going on.

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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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  • Per-pixel collision detection - why does XNA transform matrix return NaN when adding scaling?

    - by JasperS
    I looked at the TransformCollision sample on MSDN and added the Matrix.CreateTranslation part to a property in my collision detection code but I wanted to add scaling. The code works fine when I leave scaling commented out but when I add it and then do a Matrix.Invert() on the created translation matrix the result is NaN ({NaN,NaN,NaN},{NaN,NaN,NaN},...) Can anyone tell me why this is happening please? Here's the code from the sample: // Build the block's transform Matrix blockTransform = Matrix.CreateTranslation(new Vector3(-blockOrigin, 0.0f)) * // Matrix.CreateScale(block.Scale) * would go here Matrix.CreateRotationZ(blocks[i].Rotation) * Matrix.CreateTranslation(new Vector3(blocks[i].Position, 0.0f)); public static bool IntersectPixels( Matrix transformA, int widthA, int heightA, Color[] dataA, Matrix transformB, int widthB, int heightB, Color[] dataB) { // Calculate a matrix which transforms from A's local space into // world space and then into B's local space Matrix transformAToB = transformA * Matrix.Invert(transformB); // When a point moves in A's local space, it moves in B's local space with a // fixed direction and distance proportional to the movement in A. // This algorithm steps through A one pixel at a time along A's X and Y axes // Calculate the analogous steps in B: Vector2 stepX = Vector2.TransformNormal(Vector2.UnitX, transformAToB); Vector2 stepY = Vector2.TransformNormal(Vector2.UnitY, transformAToB); // Calculate the top left corner of A in B's local space // This variable will be reused to keep track of the start of each row Vector2 yPosInB = Vector2.Transform(Vector2.Zero, transformAToB); // For each row of pixels in A for (int yA = 0; yA < heightA; yA++) { // Start at the beginning of the row Vector2 posInB = yPosInB; // For each pixel in this row for (int xA = 0; xA < widthA; xA++) { // Round to the nearest pixel int xB = (int)Math.Round(posInB.X); int yB = (int)Math.Round(posInB.Y); // If the pixel lies within the bounds of B if (0 <= xB && xB < widthB && 0 <= yB && yB < heightB) { // Get the colors of the overlapping pixels Color colorA = dataA[xA + yA * widthA]; Color colorB = dataB[xB + yB * widthB]; // If both pixels are not completely transparent, if (colorA.A != 0 && colorB.A != 0) { // then an intersection has been found return true; } } // Move to the next pixel in the row posInB += stepX; } // Move to the next row yPosInB += stepY; } // No intersection found return false; }

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  • Troubleshooting Network Speeds -- The Age Old Inquiry

    - by John K
    I'm looking for help with what I'm sure is an age old question. I've found myself in a situation of yearning to understand network throughput more clearly, but I can't seem to find information that makes it "click" We have a few servers distributed geographically, running various versions of Windows. Assuming we always use one host (a desktop) as the source, when copying data from that host to other servers across the country, we see a high variance in speed. In some cases, we can copy data at 12MB/s consistently, in others, we're seeing 0.8 MB/s. It should be noted, after testing 8 destinations, we always seem to be at either 0.6-0.8MB/s or 11-12 MB/s. In the building we're primarily concerned with, we have an OC-3 connection to our ISP. I know there are a lot of variables at play, but I guess I was hoping the experts here could help answer a few basic questions to help bolster my understanding. 1.) For older machines, running Windows XP, server 2003, etc, with a 100Mbps Ethernet card and 72 ms typical latency, does 0.8 MB/s sound at all reasonable? Or do you think that slow enough to indicate a problem? 2.) The classic "mathematical fastest speed" of "throughput = TCP window / latency," is, in our case, calculated to 0.8 MB/s (64Kb / 72 ms). My understanding is that is an upper bounds; that you would never expect to reach (due to overhead) let alone surpass that speed. In some cases though, we're seeing speeds of 12.3 MB/s. There are Steelhead accelerators scattered around the network, could those account for such a higher transfer rate? 3.) It's been suggested that the use SMB vs. SMB2 could explain the differences in speed. Indeed, as expected, packet captures show both being used depending on the OS versions in play, as we would expect. I understand what determines SMB2 being used or not, but I'm curious to know what kind of performance gain you can expect with SMB2. My problem simply seems to be a lack of experience, and more importantly, perspective, in terms of what are and are not reasonable network speeds. Could anyone help impart come context/perspective?

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  • How to resize YouTube player when Window Resize

    - by Permana
    I want to show Popup window contain YouTube video. My question is how to resize YouTube Player when user resize the Popup Window ? Head section of Popup windows PHP/HTML code <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html xmlns="http://www.w3.org/1999/xhtml"> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /> <title>Wavin Video</title> <script src="jquery-1.4.1.min.js" type="text/javascript"></script> <script type="text/javascript" charset="utf-8"> $(document).ready(function(){ //If the User resizes the window //$(window).bind("resize", resizeWindow); $(window).resize(resizeWindow).resize() function resizeWindow( e ) { var target = "#youtubebox2"; var newWindowHeight = $(window).height(); var newWindowWidth = $(window).width(); console.log("Width : "+newWindowWidth); console.log("Height: "+newWindowHeight); console.log("---------"); $(target).html("<object width=\""+newWindowWidth+"\" height=\""+newWindowHeight+"\" id=\"youtubebox\"><param name=\"movie\" value=\"http://www.youtube.com/v/<?php echo $_GET['filename'];?>\"></param><param name=\"allowFullScreen\" value=\"true\"></param><param name=\"allowscriptaccess\" value=\"always\"></param><embed src=\"http://www.youtube.com/v/<?php echo $_GET['filename'];?>\" type=\"application/x-shockwave-flash\" allowscriptaccess=\"always\" allowfullscreen=\"true\" width=\""+newWindowWidth+"\" height=\""+newWindowHeight+"\"></embed></object>"); } }); </script> </head> <body> <div align="center" style="padding:6px 0px 0px 0px;background-color: #ccc;" id="youtubebox2"> <object width="480" height="385" id="youtubebox"> <param name="movie" value="http://www.youtube.com/v/<?php echo $_GET['filename'];?>"></param> <param name="allowFullScreen" value="true"></param> <param name="allowscriptaccess" value="always"></param> <embed src="http://www.youtube.com/v/<?php echo $_GET['filename'];?>" type="application/x-shockwave-flash" allowscriptaccess="always" allowfullscreen="true" width="480" height="385"> </embed> </object> </div> </body> </html> the page receive youtube video ID through $_GET variable. the code above didn't work, the YouTUbe Player is not resized. Is there something I miss ?

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  • problem with filtering the dropdownlist in scroll window

    - by Rahul
    Hi all, Problem with filtering Dropdown list. The scenario is : in scroll window there are two fields Document types and type id: Document type is Dropdown list: As per document type selection type id look should display the values. For ex. If I select quote type from document type and if I open type id look up it should display only quotation in the look window. It should work for all the values of document type drop down list values. Its working fine. The item in the document types are: Quote, Order, Invoice, Return, BackOrder. The problem is after saving the data when I am displaying the same record in scroll window, suppose after displaying document type is QUOTE and document id is QTOARD, and in this position I am changing the document type from dropdown QUOTE to ORDER at this time warning message should c ome this range entered is in valid. Because in database table there is no document QTOARD for ORDER type. The same should work for all the condition. The table name is SOP_ID_Setp and key is SOP Type and DocumentID. For that I have written the Stored procedure : create procedure DocTypeFilter @DocumentType as int, @DocumentID as varchar(30) as --declare --@documentype int, --@documentID varchar(30), select * from sop40200 where soptype=@DocumentType and docid=@DocumentID and I have called this SP in Dropdownlist change event. local long retcode; range clear table SOP_ID_SETP; clear field 'SOP Type' of table SOP_ID_SETP; clear field 'Document ID' of table SOP_ID_SETP; range start table SOP_ID_SETP; fill field 'SOP Type' of table SOP_ID_SETP; fill field 'Document ID' of table SOP_ID_SETP; range end table SOP_ID_SETP; if err()=OKAY then call DocTypeFilter,retcode,'Document Type' of window 'Is_Document Type Site_Scroll','Document ID' of window 'Is_Document Type Site_Scroll'; else warning "The range entered is invalid"; clear window 'Is_Document Type Site_Scroll'; fill window 'Is_Document Type Site_Scroll' table is_sop_site_line_temp; end if; Above code not giving the expected output any help pls.

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  • How to fix this window.open memory leak?

    - by DotnetShadow
    Hi there, I was recently looking at this memory leak tool sIEve: http://home.orange.nl/jsrosman/ So I decided to test out the tool by creating a main page that will open up a popup window. I started by creating 3 pages: index.html, page1.html and page2.html, the index.html page will open a child window (popup) linking to page1.html. Page1 will have a anchor tag that links to page2.html, while page2 will have a link back to page1.html PROBLEM So in the tool I entered the index.html page, popup window opened to page1.html, I then clicked the page2 link, no leaks detected yet. While I'm on page2 I click the link back to page1, and that's where the tool claims there is a link. The leak seems to be happening on the index.html page and I have no idea as to why it would be doing that. Even more concerning is that I can see elements that the tool detects that aren't even on my page. Does anyone have any experience with this tool or know if this really is a memory leak? Any samples of showing how to achieve what I'm doing without memory leaks? INDEX.HTML <script type="text/javascript"> MYLEAK = function() { var childWindow = null; function showWindow() { childWindow = window.open("page1.html", "myWindow"); return false; } return { init: function() { $("#window-link").bind("click", showWindow); } } }(); </script> </head> <body> <a id="window-link" href="#" on>Open Window</a> <script type="text/javascript"> $(document).ready(function() { MYLEAK.init(); }); </script> </body> </html> PAGE1.HTML <html> <body> <h1>Page 1</h1> <a href="page2.html">Page2</a> </body> </html> PAGE2.HTML <html> <body> <h1>Page 2</h1> <a href="page1.html">Page1</a> </body> </html> Appreciate your efforts.

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  • Objective c TCP/IP client

    - by charith
    Hi all, I have a TCP server running on a machine. (implemented in Java). I need to connect to that server from a iPhone and send data to the server and also, receive data on the iphone when server pushes me data. So I need to be notified when data pushes from the server. Is there a way to do this in Objective C(socket programming). Although I googled I couldn't find a solution. But I saw CFSocket etc. Please anyone have a solution?

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  • Max TCP Connections to a machine

    - by A9S6
    I am creating a Windows Service in .NET to which N number of client can connect. The service starts a TCP listener and accepts the client connections. The problem I am facing is that I can only open 10 connections to this service. The listener::AcceptTcpClient() method accepts only 10 connection and throws an exception for 11th one. The client application uses the System.Net.Sockets.TcpClient class and the service is using System.Net.Sockets.TcpListener class. This is the exception that I am getting when I try to make a number of connections in a for loop to this service (after the 10th connection is made): "Unable to read data from Transport connection: An exsting connection was forcibly closed by remote host"

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  • Stream a continously growing file over tcp/ip

    - by Grinner
    Hello, I have a project I'm working on, where a piece of Hardware is producing output that is continuously being written into a textfile. What I need to do is to stream that file as it's being written over a simple tcp/ip connection. I'm currently trying to that through simple netcat, but netcat only sends the part of the file that is written at the time of execution. It doesn't continue to send the rest. Right now I have a server listening to netcat on port 9000 (simply for test-purposes): netcat -l 9000 And the send command is: netcat localhost 9000 < c:\OUTPUTFILE So in my understanding netcat should actually be streaming the file, but it simply stops once everything that existed at the beginning of the execution has been sent. It doesn't kill the connection, but simply stops sending new data. How do I get it to stream the data continuously? Thanks for any help!

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  • TCP echo server

    - by khera-satinder
    I have written a code for TCP echo server code in C i.e. whatever I receive I have to send it back. I am also successful in doing so but I am facing a problem. Sometimes the packets that are received are not echoed back. For this I have introduced a delay after receiving and the no. of failures reduced but the problem still exists. Can someone suggest something? Later on I would like to run two server applications simultaneously on two different ports. Regards, Satinder Singh

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  • Close resources before exiting JFrame and TCP communication in Java

    - by Oz Molaim
    1. I'm writing a chat based application on TCP communication. I'm using NetBeans and I want to add functionality to the default EXIT_ON_CLOSE when exiting JFrame. The reason of course is because I want to clean resources and end threads safely. How can I call a method that clear resources and only then close the JFrame safely and end the process. 2. I need to implement the server side. The server has List/HashMap/Queue of 'Socket' with their chat nick-names. Is there any simple design pattern to do it correctly because I don't want to re-invent the wheel. thanks.

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  • Choking experienced while using the TCP/IP Adapter for BizTalk Server 2006

    - by Burhan
    I am using the TCP/IP Adapter for BizTalk Server 2006 which was obtained from codeplex: http://www.codeplex.com/BTSTCPIP Once the application was deployed in production, we started to experience choking in the performance of the application. The more the requests, the more the performance degradation. Sometimes, it happens that the receive ports become non-responsive and we have to forcefully restart the host instances to temporarily let the services respond again but we experience the same problems again and again. I would like to ask if any of you have used the same adapter and have you ever experienced the similar issues? If yes, how can we overcome theses issues. Thanks.

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  • Replacing TCP/IP pipe with WCF

    - by msarchet
    So currently my company is using a TCP/IP connection to talk between server and client programs, right now we are building this connection using System.RunTime.Remoting, which is clunky and not that reliable. It was built about 5 years ago and the model keeps getting reused and it's starting to propagate some issues, ports used, refused connections, etc. I'm trying to find some resources on how to change this over to WCF but I'm not really sure what I am looking for or what I should be searching. If you want some more information on what were actually doing with it I can go into some detail, but I'll need to pull up the code and make sure I explain it completely. thanks!

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  • checksum error with building an HTTP packet(but over TCP, like syn/ack its ok)

    - by Hila
    I am building a NAT program,I change each packet that comes from our internal subnet, change it's source IP address by libnet functions.( catch the packet with libpcap, put it sniff structures and build the new packet with libnet) I am trying to build an http packet. When I look on wireshark, I see that the new packet that I have built is exectly like the original packet(the only diffrent is that I changed the src port and ip), but there is a checksum error, So the server don't do anything with the packet that I have sent to him, beacuse the cheksum field is wrong. When I send a tcp packet(like syn or ack), the checksum is ok, and the server respons. Is anyone knows what can cause this problem? the new checksum in other packets is calculated as it should be.. but in the HTTP packet it doesn't..

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  • wcf net.tcp service fails to start when extra properties are set

    - by Pharabus
    i have a current project that runs fine with a self hosted net.tcp binding if I uses the following host.AddServiceEndpoint(typeof(IMonitorService), new NetTcpBinding() {PortSharingEnabled = false }, ""); host.AddServiceEndpoint(ServiceMetadataBehavior.MexContractName, MetadataExchangeBindings.CreateMexTcpBinding(), "mex"); however if I ammend to the below it fails to run with the message that there is already an endpoint on the port, can anyone explain why adding the extra properties causes it to fail? host.AddServiceEndpoint(typeof(IMonitorService), new NetTcpBinding() {PortSharingEnabled = false,ListenBacklog=1000,ReceiveTimeout=new TimeSpan(0,3,0) }, ""); host.AddServiceEndpoint(ServiceMetadataBehavior.MexContractName, MetadataExchangeBindings.CreateMexTcpBinding(), "mex"); Edit: testing confirms that the ReceiveTimeout property works Ok, as soon as I add the MaxConnections or ListenBacklog the service fails start Edit 2: this link seems to imply i ned portsharing is i want to modify these properies, not sure I am understanding it.

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  • Measure data transfer rate over tcp using c#

    - by publicENEMY
    i want to measure current download speed. im sending huge file over tcp. how can i capture the transfer rate every second? if i use IPv4InterfaceStatistics or similar method, instead of capturing the file transfer rate, i capture the device transfer rate. the problem with capturing device transfer rate is that it captures all ongoing data through the network device instead of the single file that i transfer. how can i capture the file transfer rate? im using c#.

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  • GNOME 2 + Compiz equivalent?

    - by virtualeyes
    Running Fedora 14 and realize I need to either change distros or find an alternative to GNOME 3 in Fedora 17. Based on what I have read to-date, XFCE and KDE are the go-to WMs if I want to avoid GNOME 3. I tried KDE 4 and I wasn't impressed; I like the simplicity of GNOME 2 with Compiz and Emerald. Can't stay on Fedora 14 forever, however, so...where to turn? Basically looking for these features in my desktop environment: GNOME Do or equivalent Snap to grid/Window tiling A must-have, the ability to hot key focused window to a monitor grid region is a huge productivity win. Zoom window to cursor In a multi-monitor setup sometimes it's nice to, say, GNOME Do terminal in one monitor and then hot key the opened window to the other monitor just by zipping the mouse cursor anywhere on target monitor (followed by, of course, snap-to-grid hotkey, all without a single mouse click) Polarization At night white background hurts the eyes, so I prefer to hot key polarize to black. Multi-monitor support I'm partial to Fedora given that I've worked with CentOS for years and have little experience with any other Linux distro; however, if the difference between Fedora and Arch, Mint, etc. is fairly subtle, I'll make the leap, just need a distro & desktop environment that allows me to be productive with keyboard hot keys and provides the above basic features. Any suggestions?

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  • How do I get my ubuntu server to listen for database connections?

    - by Bob Flemming
    I am having a problems connecting to my database outside of phpmyadmin. Im pretty sure this is because my server isn't listening on port 3306. When I type: sudo netstat -ntlp on my OTHER working server I can see the following line: tcp 0 0 0.0.0.0:3306 0.0.0.0:* LISTEN 20445/mysqld However, this line does not appear on the server I am having difficulty with. How do I make my sever listen for mysql connections? Here my my.conf file: # # The MySQL database server configuration file. # # You can copy this to one of: # - "/etc/mysql/my.cnf" to set global options, # - "~/.my.cnf" to set user-specific options. # # One can use all long options that the program supports. # Run program with --help to get a list of available options and with # --print-defaults to see which it would actually understand and use. # # For explanations see # http://dev.mysql.com/doc/mysql/en/server-system-variables.html # This will be passed to all mysql clients # It has been reported that passwords should be enclosed with ticks/quotes # escpecially if they contain "#" chars... # Remember to edit /etc/mysql/debian.cnf when changing the socket location. [client] port = 3306 socket = /var/run/mysqld/mysqld.sock # Here is entries for some specific programs # The following values assume you have at least 32M ram # This was formally known as [safe_mysqld]. Both versions are currently parsed. [mysqld_safe] socket = /var/run/mysqld/mysqld.sock nice = 0 [mysqld] # # * Basic Settings # user = mysql pid-file = /var/run/mysqld/mysqld.pid socket = /var/run/mysqld/mysqld.sock port = 3306 basedir = /usr datadir = /var/lib/mysql tmpdir = /tmp lc-messages-dir = /usr/share/mysql #skip-networking=off #skip_networking=off #skip-external-locking # # Instead of skip-networking the default is now to listen only on # localhost which is more compatible and is not less secure. #bind-address = 0.0.0.0 # # * Fine Tuning # key_buffer = 64M max_allowed_packet = 64M thread_stack = 650K thread_cache_size = 32 # This replaces the startup script and checks MyISAM tables if needed # the first time they are touched myisam-recover = BACKUP #max_connections = 100 #table_cache = 64 #thread_concurrency = 10 # # * Query Cache Configuration # query_cache_limit = 2M query_cache_size = 32M # # * Logging and Replication # # Both location gets rotated by the cronjob. # Be aware that this log type is a performance killer. # As of 5.1 you can enable the log at runtime! #general_log_file = /var/log/mysql/mysql.log #general_log = 1 # # Error logging goes to syslog due to /etc/mysql/conf.d/mysqld_safe_syslog.cnf. # # Here you can see queries with especially long duration #log_slow_queries = /var/log/mysql/mysql-slow.log #long_query_time = 2 #log-queries-not-using-indexes # # The following can be used as easy to replay backup logs or for replication. # note: if you are setting up a replication slave, see README.Debian about # other settings you may need to change. #server-id = 1 #log_bin = /var/log/mysql/mysql-bin.log expire_logs_days = 10 max_binlog_size = 100M #binlog_do_db = include_database_name #binlog_ignore_db = include_database_name # # * InnoDB # # InnoDB is enabled by default with a 10MB datafile in /var/lib/mysql/. # Read the manual for more InnoDB related options. There are many! # # * Security Features # # Read the manual, too, if you want chroot! # chroot = /var/lib/mysql/ # # For generating SSL certificates I recommend the OpenSSL GUI "tinyca". # # ssl-ca=/etc/mysql/cacert.pem # ssl-cert=/etc/mysql/server-cert.pem # ssl-key=/etc/mysql/server-key.pem [mysqldump] quick quote-names max_allowed_packet = 32M [mysql] #no-auto-rehash # faster start of mysql but no tab completition [isamchk] key_buffer = 32M # # * IMPORTANT: Additional settings that can override those from this file! # The files must end with '.cnf', otherwise they'll be ignored. # !includedir /etc/mysql/conf.d/

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  • How to transfer data between two networks efficiently

    - by Tono Nam
    I would like to transfer files between two places over the internet. Right now I have a VPN and I am able to browse, download and transfer files. So my question is not really how to transfer the files; Instead, I would like to use the most efficient approach because the two places constantly share a lot of data. The reason why I want to get rid of the VPN is because it is two slow. Having high upload speed is very expensive/impossible in residential places so I would like to use a different approach. I was thinking about using programs such as http://www.dropbox.com . The problem with Dropbox is that the free version comes with only 2 GB of storage. I think the deals they offer are OK and I might be willing to pay to get that increase in speed. But I am concerned with the speed of transferring data. Dropbox will upload the file to their server then send it from the server to the other location. I would like it to be even faster. Anyway I was thinking why not create a program myself. This is the algorithm that I was thinking of. Let me know if it sounds too crazy. (Remember my goal is to transfer files as fast as possible) Things that I will use in this algorithm: Server on the internet called S (Has fast download and upload speed. I pay to host a website and some services in there. I want to take advantage of it.) Client A at location 1 Client B at location 2 So lets say at location 1, 20 large files are created and need to be transferred to location 2. Client A compresses the files with the highest compression ratio possible. Client A starts sending data via UDP to client B. Because I am using UDP I will include the sequence number on each packet. Have server S help speed up things. For example every time a packet is lost we can use Server S to inform client A that it needs to resend a packet. Anyways I think this approach will increase the transfer rate. I do not know if it is possible to start sending data while it is being compressed. Or if it is possible to start decompressing data even if we are not done receiving the whole file. Maybe it will be faster to start sending the files right away without compressing. If I knew that I will always be sending large text files then I will obviously use the compression. I need this as a general algorithm. So I guess my question is could I increase performance by using UDP instead of TCP and by using an extra server to keep track of lost packets? And how should I compress files before sending? Compressing a 1 GB file with the highest compression ratio takes about 1 hour! I would like to take advantage of that time by sending it as it is being compressed.

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  • How to transfer data between two netowks efficiently

    - by Tono Nam
    I will like to transfer files between two places over the internet. Right now I have a VPN and I am able to browse, download and transfer files. So my question is not really how to transfer the files; Instead, I will like to use the most efficient approach because the two places constantly share a lot of data. The reason why I want to get rid of the vpn is because it is two slow. Having high upload speed is very expensive/impossible on residential places so I will like to use a different approach. I was thinking about using programs such as http://www.dropbox.com . The problem with dropbox is it only enables 2 GB of storage in order for it to be free. I think the deals they offer are ok and I might be willing to pay to get that increase in speed. But I am concerned with the speed of transferring data. Dropbox will upload the file to their server then send it from the server to the other location. I will like it even faster lol. Anyways I was thinking why not create a program my self. This is the algorithm that I was thinking let me know if it sounds to crazy. (remember my goal is to transfer files as fastest as possible) Things that I will use in this algorithm: Server on the internet called S ( has fast download and upload speed. I pay to host a website and some services in there. I want to take advantage of it) Client A on location 1 Client B on location 2 So lets say on location 1 20 large files are created and need to be transferred to location 2. Client A compresses the files with the highest compression ratio possible. Client A starts sending data via UDP to client B. Because I am using UDP I will include the sequence number on each package. Have server S help speed up things. For example every time a package is lost we can use Server S to inform client A that it needs to resend a package. Anyways I think this approach will increase the transfer rate. I do not know if it is possible to start sending data meanwhile it is being compressed. Also if it is possible to start decompressing data even if we are not done receiving all the info. Maybe it will be faster to start sending the files right away without compressing. If I knew that I will always be sending large text files then I will obviously use the compression. I need this as a general algorithm. So i guess my question is should using UDP over TCP could increase performance by using an extra server to keep track of lost packages? and How should I compress files before sending? compressing a 1 GB file with the highest compression ration takes about 1 hour! I will like to take advantage of that time by sending it meanwhile it is compressed.

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  • How can I make Cygwin open a new window each time I use a Window 7 keyboard shortcut?

    - by Michael Gundlach
    [Update: The short answer is, if an application in the 3rd thing on your taskbar, press WindowsKey+Shift+3 to open a new instance. Hooray!] I have Chrome and Cygwin on my taskbar. Chrome's shortcut is Ctrl-Alt-C (as set through right clicking the icon and putting Ctrl-Alt-C into Chrome - Properties - Shortcut Key). Cygwin's shortcut is Ctrl-Alt-T. When I press Ctrl-Alt-C, I get a new Chrome window. Great! It's as if I had shift-clicked on the Chrome icon. When I press Ctrl-Alt-T, I get a Cygwin window the first time, but after that I just get focused on the Cygwin window. As if I had simply clicked on the Cygwin icon and not shift-clicked. As if Cygwin wasn't able to have more than one instance open. I can still shift-click the icon to get more instances. I've tried with different keys than Ctrl-Alt-T and got the same behavior. Strangely, I've twice managed to get it into a state (via just clearing and setting the shortcut key over and over) where a shortcut WOULD open multiple instances -- but it was Ctrl-Alt-G both times, which doesn't make sense to my brain which has been trained to use Ctrl-Alt-T for years. Ctrl-Alt-G usually behaves just as poorly as Ctrl-Alt-T, except for the two times when magically it started behaving properly. So I'm thinking this is a Windows 7 bug (which has existed since Windows XP at least), but I'm hoping someone knows something I don't :)

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  • Chrome's new window disappear from desktop

    - by faulty
    I'm having this problem with Chrome on a Windows 7 x86 machine. I'm using Chrome for a locally hosted intranet web app. In the web app, there's frequent use of new window which popup, which works like a dialog box for the app. The problem started since yesterday. All the new windows doesn't appear on top and no where to be found on the desktop. But it does appear on the taskbar. Clicking the button on the taskbar doesn't bring up the window as well. I've also tried Alt+Space, then Move, it doesn't work. I only found 2 way to bring it up, Maximize or Show as tab. What could have caused this and how should I fix it? Thanks EDIT: I've configure to not group taskbar buttons. EDIT: I've just gone through the javascript of the button, it call window.open to display the popup. I can see the popup flicker and disappear. If I step through the code, the popup will display properly, sometime. Also, if I click the button twice, it will also display the popup, but at a much smaller size, less than 100x100

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