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• Mysql password hashing method old vs new

  - by The Disintegrator

  I'm trying to connect to a mysql server at dreamhost from a php scrip located in a server at slicehost (two different hosting companies). I need to do this so I can transfer new data at slicehost to dreamhost. Using a dump is not an option because the table structures are different and i only need to transfer a small subset of data (100-200 daily records) The problem is that I'm using the new MySQL Password Hashing method at slicehost, and dreamhost uses the old one, So i get $link = mysql_connect($mysqlHost, $mysqlUser, $mysqlPass, FALSE); Warning: mysql_connect() [function.mysql-connect]: OK packet 6 bytes shorter than expected Warning: mysql_connect() [function.mysql-connect]: mysqlnd cannot connect to MySQL 4.1+ using old authentication Warning: mysql_query() [function.mysql-query]: Access denied for user 'nodari'@'localhost' (using password: NO) facts: I need to continue using the new method at slicehost and i can't use an older php version/library The database is too big to transfer it every day with a dump Even if i did this, the tables have different structures I need to copy only a small subset of it, in a daily basis (only the changes of the day, 100-200 records) Since the tables are so different, i need to use php as a bridge to normalize the data Already googled it Already talked to both support stafs The more obvious option to me would be to start using the new MySQL Password Hashing method at dreamhost, but they will not change it and i'm not root so i can't do this myself. Any wild idea? By VolkerK sugestion: mysql> SET SESSION old_passwords=0; Query OK, 0 rows affected (0.01 sec) mysql> SELECT @@global.old_passwords,@@session.old_passwords, Length(PASSWORD('abc')); +------------------------+-------------------------+-------------------------+ | @@global.old_passwords | @@session.old_passwords | Length(PASSWORD('abc')) | +------------------------+-------------------------+-------------------------+ | 1 | 0 | 41 | +------------------------+-------------------------+-------------------------+ 1 row in set (0.00 sec) The obvious thing now would be run a mysql SET GLOBAL old_passwords=0; But i need SUPER privilege to do that and they wont give it to me if I run the query SET PASSWORD FOR 'nodari'@'HOSTNAME' = PASSWORD('new password'); I get the error ERROR 1044 (42000): Access denied for user 'nodari'@'67.205.0.0/255.255.192.0' to database 'mysql' I'm not root... The guy at dreamhost support insist saying thet the problem is at my end. But he said he will run any query I tell him since it's a private server. So, I need to tell this guy EXACTLY what to run. So, telling him to run SET SESSION old_passwords=0; SET GLOBAL old_passwords=0; SET PASSWORD FOR 'nodari'@'HOSTNAME' = PASSWORD('new password'); grant all privileges on *.* to nodari@HOSTNAME identified by 'new password'; would be a good start?

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• Using native MySQL driver in Erlang

  - by Mickey Shine

  I am using native MySQL driver (http://code.google.com/p/erlang-mysql-driver/) with mochiweb. When I tried that MySQL driver in shell mode, all woked fine. But when I write some code with Mochiweb, it reported me the following error: =CRASH REPORT==== 4-Jul-2009::04:44:29 === crasher: initial call: mochiweb_socket_server:acceptor_loop/1 pid: <0.61.0> registered_name: [] exception error: no function clause matching mysql:fetch(p1,<<"SELECT * FROM cdb_forums LIMIT 10">>) in function perly_web:loop/2 in call from mochiweb_http:headers/5 ancestors: [perly_web,perly_sup,<0.58.0>] messages: [] links: [<0.60.0>,#Port<0.965>] dictionary: [{mochiweb_request_body,undefined}, {mochiweb_request_qs,[]}, {mochiweb_request_post,[]}, {mochiweb_request_path,"/online"}, {mochiweb_request_cookie, [{"04c_sid","hG9Oyv"}, {"04c_visitedfid","2"}, {"kQx_cookietime","2592000"}, {"kQx_loginuser","admin"}, {"kQx_activationauth", "98b3mdX86fKT9dI4WyKuL61Tqxk%2BW1r6ACpHp9y8itH2xQ"}, {"smile","1D1"}]}] trap_exit: false status: running heap_size: 1597 stack_size: 24 reductions: 5188 neighbours: The code I write in Mochiweb is start(Options) -> {DocRoot, Options1} = get_option(docroot, Options), Loop = fun (Req) -> ?MODULE:loop(Req, DocRoot) end, % we’ll set our maximum to 1 million connections. (default: 2048) mochiweb_http:start([{max, 1000000}, {name, ?MODULE}, {loop, Loop} | Options1]), mysql:start_link(p1, "10.0.0.123", "root", "root", "test"). stop() -> mochiweb_http:stop(?MODULE). loop(Req, DocRoot) -> "/" ++ Path = Req:get(path), case Req:get(method) of Method when Method =:= 'GET'; Method =:= 'HEAD' -> case Path of "online" -> Result1 = mysql:fetch(p1, <<"SELECT * FROM cdb_forums LIMIT 10">>), Body1 = io:format("Result1: ~p~n", [Result1]), Req:ok({"text/plain", Body1}); The connection looks good but when I added Result1 = mysql:fetch(p1, <<"SELECT * FROM cdb_forums LIMIT 10">>), it crashed. Can someone help me? Thanks in advance~ //================================================== updated: I noticed the follwoing information. If that is correct? =PROGRESS REPORT==== 4-Jul-2009::05:49:32 === supervisor: {local,kernel_safe_sup} started: [{pid,<0.65.0>}, {name,inet_gethost_native_sup}, {mfa,{inet_gethost_native,start_link,[]}}, {restart_type,temporary}, {shutdown,1000}, {child_type,worker}] mysql_conn: greeting version "5.1.33-log" (protocol 10) salt "ne0_m'vA" caps 63487 serverchar <<8,2,0,0, 0,0,0,0, 0,0,0,0, 0,0,0,0>> salt2 "!|o;vabJ*4bt" mysql_auth send packet 1: <<5,162,0,0,64,66,15,0,8,0,0,0,0,0,0,0,0,0,0,0,0,0,0, 0,0,0,0,0,0,0,0,0,114,111,111,116,0,20,52,235,78, 173,36,251,201,242,172,139,113,231,253,181,245,3, 91,198,111,135>> Link: {ok,<0.62.0>} =SUPERVISOR REPORT==== 4-Jul-2009::05:49:32 === Supervisor: {local,perly_sup} Context: start_error Reason: ok Offender: [{pid,undefined}, {name,perly_web}, {mfa, {perly_web,start, [[{ip,"0.0.0.0"}, {port,8000}, {docroot, "/work/mochiweb-read-only/scripts/perly/priv/www"}]]}}, {restart_type,permanent}, {shutdown,5000}, {child_type,worker}]

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• "RFC 2833 RTP Event" Consecutive Events and the E "End" Bit

  - by brian_d

  Hello, I can send out a RFC 2833 dtmf event as outlined at http://www.ietf.org/rfc/rfc2833.txt When I do set the E "End" bit, but leave it as 0, I get the following behaviour: If for example keys 7874556332111111145855885#3 were pressed, then ALL events would be sent and show up in a program like wireshark, however only 87456321458585#3 would sound. So the first key (which I figure could be a separate issue) and any repeats of an event (ie 11111) are failing to sound. In section 3.9, figure 2 of the above linked document, they give a 911 example. Here all but the last event have the E bit set. When I set the bit for all numbers, I never get an event to sound. I have thought of a couple possible thing but do not know if they are the reason: 1) figure 2 shows payload types of 96 and 97 sent. I have not nor know how to exactly. In section 3.8, codes 96 and 97 are described as "the dynamic payload types 96 and 97 have been assigned for the redundancy mechanism and the telephone event payload respectively" 2) In section 3.5, "E:", "A sender MAY delay setting the end bit until retransmitting the last packet for a tone, rather than on its first transmission" Does anyone have an idea of how to actually do this? I have also fiddled around with timestamp intervals and the RTP marker. Any help is greatly appreciated. Here is a sample wireshark event capture of the relevant areas: 6590 31.159045000 xx.x.x.xxx --.--.---.-- RTP EVENT Payload type=RTP Event, DTMF Pound # (end) Real-Time Transport Protocol Stream setup by SDP (frame 6225) Setup frame: 6225 Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: telephone-event (101) Sequence number: 0 Extended sequence number: 65536 Timestamp: 0 Synchronization Source identifier: 0x15f27104 (368210180) RFC 2833 RTP Event Event ID: DTMF Pound # (11) 1... .... = End of Event: True .0.. .... = Reserved: False ..00 0000 = Volume: 0 Event Duration: 2048

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• Publishing WCF .NET 3.5 to IIS 6 (Windows Server 2003)

  - by Adam

  I've been developing a WCF web service using .NET 3.5 with IIS7 and it works perfectly on my local computer. I tried publishing it to a server running IIS 6 and even though I can view the WSDL in my browser, the client application doesn't seem to be connecting to it correctly. I launched a packet sniffing app (Charles Proxy) and the response for the first message comes back to the client empty (0 bytes). Every message after the first one times out. The WCF service is part of a larger application that uses ASP .NET 3.5. That application has been working fine on IIS 6 for awhile now so I think it's something specific to WCF. I also tried throwing an exception in the SVC file to see if it made it that far and the exception never got thrown so I have a feeling it's something more low level that's not working. Any thoughts? Is there anything I need to install on the IIS5 server? If so how am I still able to view the WSDL in my browser? The service is being consumed via an SVC file using basicHttpBinding Here's the meat of the Web.Config (let me know if you need any other part of it): <system.net> <defaultProxy> <proxy usesystemdefault="False" proxyaddress="http://127.0.0.1:80" bypassonlocal="True"/> </defaultProxy> </system.net> ... <system.serviceModel> <services> <service name="Nexternal.Service.XMLTools.VNService" behaviorConfiguration="VNServiceBehavior"> <!--The first endpoint would be picked up from the confirg this shows how the config can be overriden with the service host--> <endpoint address="" binding="basicHttpBinding" contract="Nexternal.Service.XMLTools.IVNService" /> <endpoint address="mex" binding="mexHttpBinding" contract="IMetadataExchange" name="mexHttpBinding" /> </service> </services> <behaviors> <serviceBehaviors> <behavior name="VNServiceBehavior"> <serviceMetadata httpGetEnabled="true" /> <serviceDebug includeExceptionDetailInFaults="true" /> </behavior> </serviceBehaviors> </behaviors> <serviceHostingEnvironment aspNetCompatibilityEnabled="true" /> </system.serviceModel>

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• Providing DNS redirection to honeypot server for known bad domains

  - by syn-

  Currently running BIND on RHEL 5.4 and am looking for a more efficient manner of providing DNS redirection to a honeypot server for a large (30,000+) list of forbidden domains. Our current solution for this requirement is to include a file containing a zone master declaration for each blocked domain in named.conf. Subsequently, each of these zone declarations point to the same zone file, which resolves all hosts in that domain to our honeypot servers. ...basically this allows us to capture any "phone home" attempts by malware that may infiltrate the internal systems. The problem with this configuration is the large amount of time taken to load all 30,000+ domains as well as management of the domain list configuration file itself... if any errors creep into this file, the BIND server will fail to start, thereby making automation of the process a little frightening. So I'm looking for something more efficient and potentially less error prone. named.conf entry: include "blackholes.conf"; blackholes.conf entry example: zone "bad-domain.com" IN { type master; file "/var/named/blackhole.zone"; allow-query { any; }; notify no; }; blackhole.zone entries: $INCLUDE std.soa @ NS ns1.ourdomain.com. @ NS ns2.ourdomain.com. @ NS ns3.ourdomain.com.                        IN            A                192.168.0.99 *                      IN            A                192.168.0.99

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• System.Net.Dns.GetHostAddresses("")

  - by dbasnett

  Yesterday s**ked, and today ain't (sic) looking better. I have an application I have been working on and it can be slow to start when my ISP is down because of DNS. My ISP was down for 3 hours yesterday, so I didn't think much about this piece of code I had added, until I found that it is always slow to start. This code is supposed to return your IP address and my reading of the link suggests that should be immediate, but it isn't, at least on my machine. Oh, and yesterday before the internet went down, I upgraded (oymoron) to XP SP3, and have had other problems. So my questions / request: 1. Am I doing this right? 2. If you run this on your machine does it take 39 seconds to return your IP address? It does on mine. One other note, I did a packet capture and the first request did NOT go on the wire, but the second did, and was answered quickly. So the question is what happened in XP SP3 that I am missing, besides a brain. One last note. If I resolve a FQDN all is well. Public Class Form1 'http://msdn.microsoft.com/en-us/library/system.net.dns.gethostaddresses.aspx ' 'excerpt 'The GetHostAddresses method queries a DNS server 'for the IP addresses associated with a host name. ' 'If hostNameOrAddress is an IP address, this address 'is returned without querying the DNS server. ' 'When an empty string is passed as the host name, 'this method returns the IPv4 addresses of the local host Private Sub Button1_Click(ByVal sender As System.Object, _ ByVal e As System.EventArgs) Handles Button1.Click Dim stpw As New Stopwatch stpw.Reset() stpw.Start() 'originally Dns.GetHostEntry, but slow also Dim myIPs() As System.Net.IPAddress = System.Net.Dns.GetHostAddresses("") stpw.Stop() Debug.WriteLine("'" & stpw.Elapsed.TotalSeconds) If myIPs.Length > 0 Then Debug.WriteLine("'" & myIPs(0).ToString) 'debug '39.8990525 '192.168.1.2 stpw.Reset() stpw.Start() 'originally Dns.GetHostEntry, but slow also myIPs = System.Net.Dns.GetHostAddresses("www.vbforums.com") stpw.Stop() Debug.WriteLine("'" & stpw.Elapsed.TotalSeconds) If myIPs.Length > 0 Then Debug.WriteLine("'" & myIPs(0).ToString) 'debug '0.042212 '63.236.73.220 End Sub End Class

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• Java multithreaded server - each connection returns data. Processing on main thread?

  - by oliwr

  I am writing a client with an integrated server that should wait indefinitely for new connections - and handle each on a Thread. I want to process the received byte array in a system wide available message handler on the main thread. However, currently the processing is obviously done on the client thread. I've looked at Futures, submit() of ExecutorService, but as I create my Client-Connections within the Server, the data would be returned to the Server thread. How can I return it from there onto the main thread (in a synchronized packet store maybe?) to process it without blocking the server? My current implementation looks like this: public class Server extends Thread { private int port; private ExecutorService threadPool; public Server(int port) { this.port = port; // 50 simultaneous connections threadPool = Executors.newFixedThreadPool(50); } public void run() { try{ ServerSocket listener = new ServerSocket(this.port); System.out.println("Listening on Port " + this.port); Socket connection; while(true){ try { connection = listener.accept(); System.out.println("Accepted client " + connection.getInetAddress()); connection.setSoTimeout(4000); ClientHandler conn_c= new ClientHandler(connection); threadPool.execute(conn_c); } catch (IOException e) { System.out.println("IOException on connection: " + e); } } } catch (IOException e) { System.out.println("IOException on socket listen: " + e); e.printStackTrace(); threadPool.shutdown(); } } } class ClientHandler implements Runnable { private Socket connection; ClientHandler(Socket connection) { this.connection=connection; } @Override public void run() { try { // Read data from the InputStream, buffered int count; byte[] buffer = new byte[8192]; InputStream is = connection.getInputStream(); ByteArrayOutputStream out = new ByteArrayOutputStream(); // While there is data in the stream, read it while ((count = is.read(buffer)) > 0) { out.write(buffer, 0, count); } is.close(); out.close(); System.out.println("Disconnect client " + connection.getInetAddress()); connection.close(); // handle the received data MessageHandler.handle(out.toByteArray()); } catch (IOException e) { System.out.println("IOException on socket read: " + e); e.printStackTrace(); } return; } }

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• Filtering Security Logs by User and Logon Type

  - by Trido

  I have been asked to find out when a user has logged on to the system in the last week. Now the audit logs in Windows should contain all the info I need. I think if I search for Event ID 4624 (Logon Success) with a specific AD user and Logon Type 2 (Interactive Logon) that it should give me the information I need, but for the life of my I cannot figure out how to actually filter the Event Log to get this information. Is it possible inside of the Event Viewer or do you need to use an external tool to parse it to this level? I found http://nerdsknowbest.blogspot.com.au/2013/03/filter-security-event-logs-by-user-in.html which seemed to be part of what I needed. I modified it slightly to only give me the last 7 days worth. Below is the XML I tried. <QueryList> <Query Id="0" Path="Security"> <Select Path="Security">*[System[(EventID=4624) and TimeCreated[timediff(@SystemTime) &lt;= 604800000]]]</Select> <Select Path="Security">*[EventData[Data[@Name='Logon Type']='2']]</Select> <Select Path="Security">*[EventData[Data[@Name='subjectUsername']='Domain\Username']]</Select> </Query> </QueryList> It only gave me the last 7 days, but the rest of it did not work. Can anyone assist me with this? EDIT Thanks to the suggestions of Lucky Luke I have been making progress. The below is my current query, although as I will explain it isn't returning any results. <QueryList> <Query Id="0" Path="Security"> <Select Path="Security"> *[System[(EventID='4624')] and System[TimeCreated[timediff(@SystemTime) &lt;= 604800000]] and EventData[Data[@Name='TargetUserName']='john.doe'] and EventData[Data[@Name='LogonType']='2'] ] </Select> </Query> </QueryList> As I mentioned, it wasn't returning any results so I have been messing with it a bit. I can get it to produce the results correctly until I add in the LogonType line. After that, it returns no results. Any idea why this might be? EDIT 2 I updated the LogonType line to the following: EventData[Data[@Name='LogonType'] and (Data='2' or Data='7')] This should capture Workstation Logons as well as Workstation Unlocks, but I still get nothing. I then modify it to search for other Logon Types like 3, or 8 which it finds plenty of. This leads me to believe that the query works correctly, but for some reason there are no entries in the Event Logs with Logon Type equalling 2 and this makes no sense to me. Is it possible to turn this off?

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• Understanding PTS and DTS in video frames

  - by theateist

  I had fps issues when transcoding from avi to mp4(x264). Eventually the problem was in PTS and DTS values, so lines 12-15 where added before av_interleaved_write_frame function: 1. AVFormatContext* outContainer = NULL; 2. avformat_alloc_output_context2(&outContainer, NULL, "mp4", "c:\\test.mp4"; 3. AVCodec *encoder = avcodec_find_encoder(AV_CODEC_ID_H264); 4. AVStream *outStream = avformat_new_stream(outContainer, encoder); 5. // outStream->codec initiation 6. // ... 7. avformat_write_header(outContainer, NULL); 8. // reading and decoding packet 9. // ... 10. avcodec_encode_video2(outStream->codec, &encodedPacket, decodedFrame, &got_frame) 11. 12. if (encodedPacket.pts != AV_NOPTS_VALUE) 13. encodedPacket.pts = av_rescale_q(encodedPacket.pts, outStream->codec->time_base, outStream->time_base); 14. if (encodedPacket.dts != AV_NOPTS_VALUE) 15. encodedPacket.dts = av_rescale_q(encodedPacket.dts, outStream->codec->time_base, outStream->time_base); 16. 17. av_interleaved_write_frame(outContainer, &encodedPacket) After reading many posts I still do not understand: outStream->codec->time_base = 1/25 and outStream->time_base = 1/12800. The 1st one was set by me but I cannot figure out why and who set 12800? I noticed that before line (7) outStream->time_base = 1/90000 and right after it it changes to 1/12800, why? When I transcode from avi to avi, meaning changing the line (2) to avformat_alloc_output_context2(&outContainer, NULL, "avi", "c:\\test.avi"; , so before and after line (7) outStream->time_base remains always 1/25 and not like in mp4 case, why? What is the difference between time_base of outStream->codec and outStream? To calc the pts av_rescale_q does: takes 2 time_base, multiplies their fractions in cross and then compute the pts. Why it does this in this way? As I debugged, the encodedPacket.pts has value incremental by 1, so why changing it if it does has value? At the beginning the dts value is -2 and after each rescaling it still has negative number, but despite this the video played correctly! Shouldn't it be positive?

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• a problem in socks.h

  - by janathan

  i use this (http://www.codeproject.com/KB/IP/Socks.aspx) lib in my socket programing in c++ and copy the socks.h in include folder and write this code: include include include include include include "socks.h" define PORT 1001 // the port client will be connecting to define MAXDATASIZE 100 static void ReadThread(void* lp); int socketId; int main(int argc, char* argv[]) { const char temp[]="GET / HTTP/1.0\r\n\r\n"; CSocks cs; cs.SetVersion(SOCKS_VER4); cs.SetSocksPort(1080); cs.SetDestinationPort(1001); cs.SetDestinationAddress("192.168.11.97"); cs.SetSocksAddress("192.168.11.97"); //cs.SetVersion(SOCKS_VER5); //cs.SetSocksAddress("128.0.21.200"); socketId = cs.Connect(); // if failed if (cs.m_IsError) { printf( "\n%s", cs.GetLastErrorMessage()); getch(); return 0; } // send packet for requesting to a server if(socketId > 0) { send(socketId, temp, strlen(temp), 0); HANDLE ReadThreadID; // handle for read thread id HANDLE handle; // handle for thread handle handle = CreateThread ((LPSECURITY_ATTRIBUTES)NULL, // No security attributes. (DWORD)0, // Use same stack size. (LPTHREAD_START_ROUTINE)ReadThread, // Thread procedure. (LPVOID)(void*)NULL, // Parameter to pass. (DWORD)0, // Run immediately. (LPDWORD)&ReadThreadID); WaitForSingleObject(handle, INFINITE); } else { printf("\nSocks Server / Destination Server not started.."); } closesocket(socketId); getch(); return 0; } // Thread Proc for reading from server socket. static void ReadThread(void* lp) { int numbytes; char buf[MAXDATASIZE]; while(1) { if ((numbytes=recv(socketId, buf, MAXDATASIZE-1, 0)) == -1) { printf("\nServer / Socks Server has been closed Receive thread Closed\0"); break; } if (numbytes == 0) break; buf[numbytes] = '\0'; printf("Received: %s\r\n",buf); send(socketId,buf,strlen(buf),0); } } but when compile this i get an error . pls help me thanks

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• Write STDOUT & STDERR to a logfile, also write STDERR to screen

  - by Stefan Lasiewski

  I would like to run several commands, and capture all output to a logfile. I also want to print any errors to the screen (or optionally mail the output to someone). Here's an example. The following command will run three commands, and will write all output (STDOUT and STDERR) into a single logfile. { command1 && command2 && command3 ; } > logfile.log 2>&1 Here is what I want to do with the output of these commands: STDERR and STDOUT for all commands goes to a logfile, in case I need it later--- I usually won't look in here unless there are problems. Print STDERR to the screen (or optionally, pipe to /bin/mail), so that any error stands out and doesn't get ignored. It would be nice if the return codes were still usable, so that I could do some error handling. Maybe I want to send email if there was an error, like this: { command1 && command2 && command3 ; } logfile.log 2&1 || mailx -s "There was an error" [email protected] The problem I run into is that STDERR loses context during I/O redirection. A '2&1' will convert STDERR into STDOUT, and therefore I cannot view errors if I do 2 error.log Here are a couple juicier examples. Let's pretend that I am running some familiar build commands, but I don't want the entire build to stop just because of one error so I use the '--keep-going' flag. { ./configure && make --keep-going && make install ; } > build.log 2>&1 Or, here's a simple (And perhaps sloppy) build and deploy script, which will keep going in the event of an error. { ./configure && make --keep-going && make install && rsync -av --keep-going /foo devhost:/foo} > build-and-deploy.log 2>&1 I think what I want involves some sort of Bash I/O Redirection, but I can't figure this out.

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• How to know the source of certain TCP traffic on AIX

  - by A.Rashad

  We have two AIX boxes, one for production system and another for testing. both systems are running ATM machine switches, where the ATM device is connected via TCP socket. we had an issue on production system where the machine would power off or get disconnected but the netstat -na | grep <IP of machine > would still mention that the socket is up when simulated that case on the UAT environment, the problem did not happen, where the socket would terminate in 3 to 5 minutes. when sniffed on the traffic between the machine and ATM we found that no traffic takes place on production while there is some sort of heartbeat on UAT. but it is not initiated by the application. $>tcpdump | grep -v "10.2.2.71" | grep -v "HSRP" | grep "10.3.1.30" tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on en6, link-type 1, capture size 96 bytes 09:08:13.323421 IP server073.afs3-callback > 10.3.1.30.impera: . 278204201:278204202(1) ack 3307884029 win 164 09:08:13.335334 IP 10.3.1.30.impera > server073.afs3-callback: . ack 1 win 64180 09:08:23.425771 IP 10.3.1.30.impera > server073.afs3-callback: . 1:2(1) ack 1 win 64180 09:08:23.425789 IP server073.afs3-callback > 10.3.1.30.impera: . ack 2 win 65535 09:09:13.628985 IP server073.afs3-callback > 10.3.1.30.impera: . 0:1(1) ack 1 win 164 09:09:13.633900 IP 10.3.1.30.impera > server073.afs3-callback: . ack 1 win 64180 09:09:23.373634 IP 10.3.1.30.impera > server073.afs3-callback: . 1:2(1) ack 1 win 64180 09:09:23.373647 IP server073.afs3-callback > 10.3.1.30.impera: . ack 2 win 65535 while on production, that traffic is not there. we want to know where this traffic is initiated from to implement on production to sense disconnection our comms parameters are: tcp_keepcnt = 2 tcp_keepidle = 100 tcp_keepinit = 150 tcp_keepintvl = 150 tcp_finwait2 = 1200 can anyone help?

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• No exception, no error, still i dont recieve the json object from my http post

  - by user2978538

  My source code: final Thread t = new Thread() { public void run() { Looper.prepare(); HttpClient client = new DefaultHttpClient(); HttpConnectionParams.setConnectionTimeout(client.getParams(), 10000); HttpResponse response; JSONObject obj = new JSONObject(); try { HttpPost post = new HttpPost("http://pc.dyndns-office.com/mobile.asp"); obj.put("Model", ReadIn1); obj.put("Product", ReadIn2); obj.put("Manufacturer", ReadIn3); obj.put("RELEASE", ReadIn4); obj.put("SERIAL", ReadIn5); obj.put("ID", ReadIn6); obj.put("ANDROID_ID", ReadIn7); obj.put("Language", ReadIn8); obj.put("BOARD", ReadIn9); obj.put("BOOTLOADER", ReadIn10); obj.put("BRAND", ReadIn11); obj.put("CPU_API", ReadIn12); obj.put("DISPLAY", ReadIn13); obj.put("FINGERPRINT", ReadIn14); obj.put("HARDWARE", ReadIn15); obj.put("UUID", ReadIn16); StringEntity se = new StringEntity(obj.toString()); se.setContentType(new BasicHeader(HTTP.CONTENT_TYPE, "application/json")); post.setEntity(se); post.setHeader("host", "http://pc.dyndns-office.com/mobile.asp"); response = client.execute(post); if (response != null) { InputStream in = response.getEntity().getContent(); } } catch (Exception e) { e.printStackTrace(); } Looper.loop(); } }; t.start(); } } i want to send an Json object to a Website. As far as I can see, I set the header, but still I get this exception, can someone help me? (I'm using Android-Studio) __ Edit: i don't get any exceptions anymore, but still i do not receive the json packet. When i manually call the website i get a log file entry. Does anyone know, what's wrong? Edit2: When i debug i get as response "HTTP/1.1 400 bad request" i'm sure its not an permission problem. Any ideas?

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• WinPE, Startnet.CMD and passing variables to second batch file not working

  - by user140892

  I don't know scripting or PowerShell (yes I need to learn something). I'm not an expert batch file maker either. I have a WinPE flash drive which I used to deploy OS images. I have the WIM, drivers and anything needed else outside the WinPE environment to ensure that Updates, changes are easier for me to make. I use the "STARTNET.CMD" batch file which is part of the WinPE. The reason to go through the letter drives is that the WinPE always gets the X letter drive assigned. The flash drive itself can receive a random letter which always changes. My deployment menu is located on the flash drive it self and not inside the WinPE. This is so that if I need to make a change I don't have to re-do the WinPE. I am able to locate the "menu.bat" batch file and launch it. I use a variable to capture the letter drive. I call the second batch file named "menu.bat" and pass the variable to it. When the second batch file loads, I believe that I am calling the variable correctly. If I break out of the batch file I can echo the variable and see the expected reply. The issue is that I can't use the variable to work with anything on the second batch file. In my test, I can get this to work over and over. When it runs from the real USB flash drive it does not work. I removed comments from the second batch file to make it smaller. My issue is that files below all get a message stating that the system cannot find the path specified. Diskpart Imagex.exe bcdboot.exe Why can't I get the varible to properly function when I try to using example "ImageX.exe"? Contents of the Startnet.cmd @echo off for %%p in (a b c d e f g h i j k l m n o p q r s t u v w x y z) do if exist %%p:\Tools\ set w=%%p Set execpatch=%w%\Tools\ call %w%:\Menu.bat \Tools\ Contents of the Menu.BAT @echo off set SecondPath=%1 cls :Start cls Echo. Echo.============================================================== Echo. Windows 7 64 Bit Ent Basic Desktops Echo.============================================================== Echo. Echo A. 790 Windows 7 - Basic Echo. Echo. Echo I. Exit Echo. Echo. set /p choice=Choose your option = if not '%choice%'=='' set choice=%choice:~0,1% if '%choice%'=='a' goto 790_Windows_7_Basic echo "%choice%" is not a valid (answer/command) echo. goto start :790_Windows_7_Basic REM DISKPART /s %SecondPath%BatchFiles\Make-Partition.txt %SecondPath%imagex.exe /apply %SecondPath%Images\Win7-64b-Ent-Basic-SysPreped.wim 1 o:\ /verify %SecondPath%bcdboot.exe o:\Windows /s S: Copy %SecondPath%Unattended\unattend.XML o:\Windows\System32\sysprep\unattend.XML /y xcopy %SecondPath%Drivers\790\*.* o:\Windows\INF\790\ /E /Q /Y MD o:\Windows\Setup\Scripts\ Copy %SecondPath%BatchFiles\SetupComplete.cmd o:\Windows\Setup\Scripts\ /y Goto Done :Done Exit

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• BOM in a PHP page auto generated by Wordpress

  - by Paolo63

  I admin two different blogs. They are both wordpress 2.8.6 (so they have exactly the same source code, plugins apart) but they are located on two different hosting platform (hostmonster.com and aruba.it). To explain my problem I've dumped with SmartSniff a session with each one of the sites. Here is the dump from hostmonster: GET /blog/paolo/ HTTP/1.1 Host: www.e-venturi.com Accept-Encoding: identity Accept-Language: en-us Accept: text/html, text/plain, text/xml, image/gif, image/x-xbitmap, image/x-icon,image/jpeg, image/pjpeg, application/vnd.ms-powerpoint, application/vnd.ms-excel, application/msword, */* User-Agent: Mozilla/4.0 (compatible; MSIE 6.0; Windows NT 5.0;) HTTP/1.1 200 OK Date: Sat, 28 Nov 2009 23:47:38 GMT Server: Apache/2.2.14 (Unix) mod_ssl/2.2.14 OpenSSL/0.9.8l DAV/2 mod_auth_passthrough/2.1 FrontPage/5.0.2.2635 X-Powered-By: PHP/5.2.11 X-Pingback: http://www.e-venturi.com/blog/paolo/xmlrpc.php Vary: Accept-Encoding Transfer-Encoding: chunked Content-Type: text/html; charset=UTF-8 a6 <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> and now from aruba: GET /blog/ HTTP/1.1 Host: www.cubanite.net Accept-Encoding: identity Accept-Language: en-us Accept: text/html, text/plain, text/xml, image/gif, image/x-xbitmap, image/x-icon,image/jpeg, image/pjpeg, application/vnd.ms-powerpoint, application/vnd.ms-excel, application/msword, */* User-Agent: Mozilla/4.0 (compatible; MSIE 6.0; Windows NT 5.0;) HTTP/1.1 200 OK Date: Sat, 28 Nov 2009 23:49:19 GMT Server: Apache/2.2 X-Pingback: http://www.cubanite.net/blog/xmlrpc.php Vary: Accept-Encoding Transfer-Encoding: chunked Content-Type: text/html; charset=UTF-8 100b ...<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> (note: a6 and 100b are the packet size reported by SmartSniff) Ok, the big difference are the three dots in front of the <!DOCTYPE in aruba. They are the UTF-8 BOM (0xef 0xbb 0xbf). Being the same PHP source on both the servers, why does it appears only on one server ? The content is generated so the post author can't deliberately insert a BOM and I've verified the template to be BOM free too. Naturally there are different PHP and Apache versions on the servers... what can I check or set to diagnose and resolve the problem ? By the way I don't want the BOM. Many thanks in advance.

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• Given a trace of packets, how would you group them into flows?

  - by zxcvbnm

  I've tried it these ways so far: 1) Make a hash with the source IP/port and destination IP/port as keys. Each position in the hash is a list of packets. The hash is then saved in a file, with each flow separated by some special characters/line. Problem: Not enough memory for large traces. 2) Make a hash with the same key as above, but only keep in memory the file handles. Each packet is then put into the hash[key] that points to the right file. Problems: Too many flows/files (~200k) and it might run out of memory as well. 3) Hash the source IP/port and destination IP/port, then put the info inside a file. The difference between 2 and 3 is that here the files are opened and closed for each operation, so I don't have to worry about running out of memory because I opened too many at the same time. Problems: WAY too slow, same number of files as 2 so also impractical. 4) Make a hash of the source IP/port pairs and then iterate over the whole trace for each flow. Take the packets that are part of that flow and place them into the output file. Problem: Suppose I have a 60 MB trace that has 200k flows. This way, I would process, say, a 60 MB file 200k times. Maybe removing the packets as I iterate would make it not so painful, but so far I'm not sure this would be a good solution. 5) Split them by IP source/destination and then create a single file for each one, separating the flows by special characters. Still too many files (+50k). Right now I'm using Ruby to do it, which might've been a bad idea, I guess. Currently I've filtered the traces with tshark so that they only have relevant info, so I can't really make them any smaller. I thought about loading everything in memory as described in 1) using C#/Java/C++, but I was wondering if there wouldn't be a better approach here, especially since I might also run out of memory later on even with a more efficient language if I have to use larger traces. In summary, the problem I'm facing is that I either have too many files or that I run out of memory. I've also tried searching for some tool to filter the info, but I don't think there is one. The ones I've found only return some statistics and wouldn't scan for every flow as I need.

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• Ubuntu with KVM guest VM and broken bridges

  - by MadPsy

  I have an Ubuntu box with a KVM guest VM running. They use bridging so the guest VM attaches to the physical network of its host. The guest VM has 2 NICs in 2 different bridges. First NIC of the VM is tap5 and is in bridge br0 br0 8000.46720f5c572e no eth0.500 tap5 Second NIC of the VM is tap2 and is in bridge br100 br100 8000.76ad2fc96661 no eth0.100 eth0.101 eth0.103 eth0.104 eth0.105 tap2 On the host, br0 has an IP and br100 does not 21: br0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc noqueue state UP link/ether 46:72:0f:5c:57:2e brd ff:ff:ff:ff:ff:ff inet 192.168.100.4/24 brd 192.168.10.255 scope global br0 inet6 fe80::d6ae:52ff:febe:777/64 scope link valid_lft forever preferred_lft forever On the guest, its eth0 and eth1 interfaces both have IP addresses 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP qlen 1000 link/ether 00:3e:61:fb:7a:da brd ff:ff:ff:ff:ff:ff inet 192.168.100.6/24 brd 192.168.100.255 scope global eth0 inet6 fe80::23e:61ff:fefb:7ada/64 scope link valid_lft forever preferred_lft forever 3: eth1: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP qlen 1000 link/ether 00:3e:61:fb:7a:ea brd ff:ff:ff:ff:ff:ff inet 172.16.50.129/25 brd 172.16.50.255 scope global eth1 inet6 fe80::23e:61ff:fefb:7aea/64 scope link valid_lft forever preferred_lft forever On the guest VM, a tcpdump of its eth1 interface (tap2) shows traffic from its eth0 interface (tap5), as if the 2 bridges are themselves bridged. This means any interface on br100 is now bridged across to br0 - which is completely broken. root@chillispot:~# tcpdump -c 1 -n -v -i eth1 net 192.168.100.0/24 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 65535 bytes 16:31:24.175583 IP (tos 0x0, ttl 64, id 48054, offset 0, flags [DF], proto TCP (6), length 148) 192.168.100.6.22 > 192.168.100.4.59505: Flags [P.], cksum 0x6c2b (correct), seq 1056321648:1056321744, ack 398642983, win 1700, options [nop,nop,TS val 197473436 ecr 200655363], length 96 What could be bridging the 2 bridges, except the guest VM (which is a stock Ubuntu install)? I am at a complete loss! Thanks.

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• How to configure the framesize using AudioUnit.framework on iOS

  - by Piperoman

  I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

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• Lag spikes at full CPU usage, maybe video card

  - by Roberts

  I am posting this thread in hurry so few things may be missed (I will update tomorrow). My PC specs: Motherboard Name - Gigabyte GA-945PL-S3 CPU Type - DualCore Intel Core 2 Duo E4300, 1800 MHz (9 x 200) OS - Microsoft Windows 7 Ultimate OS Kernel Type - 32-bit OS Version - 6.1.7601 I bougth a new video card one month ago. GeForce 210. I didn't have any problems. I wanted to overclock it, in other words: "Play with it". So I installed Gigabyte EasyBoost from CD and overclocked the GPU 590 + 110 mhz, memory to max to 960mhz from 800mhz. Benchmarks showed a little bit bigger score. Then I overclocked shader clock from 1405 to [..] (don't remeber really). So I was playing Modern Warfare 2 when off sudden computer froze when I wanted to select team, I was afk before that. I had to reset CMOS. After that I had problems with Skype: unread messages and no sound. Then I figured it out that when ever I open EasyBoost - Skype starts to glitch again. Now I use EVGA Precission X. Now after a month, I cleaned computer and closed the case, it was open all the time. I started to overclock GPU clock only (just a bit) because there was no problems that would stop me. So sometimes on heavy CPU load graphics starts to lag. Dragging a window is painful to watch too. Sometimes the screen freezes for 5 to 10 seconds (I can see that hard disk activity is maximal). You may say that CPU fault it is, isn't it? But sometimes lag spikes starts randomly when CPU load is at maximum. All 3 benchmark softwares (PerformanceTest, NovaBench and MSI Kombustor) shows that performance of my video card has dropped about 25%. BUT! CPU score is lower too. I ignored these problems but when I refreshed Windows Experience Index I was shocked. Month before (in latvian language but not so hard to understand): Now (upgraded RAM): This happened when I tried to capture Minecraft with Fraps on underclocked GPU to 580mhz (def: 590mhz): All drivers are up to date. Average CPU temperature from 55°C to 75°C (at 70°C sometimes starts these lag spikes). Video card's tempratures are from 45°C to 60°C (very hard to reach 60°C). So my hope is that the video card is fine, cause this card is very new and I want to upgrade CPU anyways. Aplogies for my mistakes in vocabulary (I am trying to type this as fast I can).

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• Problem with tcp server when converting to service

  - by djerry

  Hello lads, I'm working on monitoring some object (cdr-packets). I'm setting up a tcp-server and am listening on port 50043 for the packages. The program as a console application is working just fine, my server is working like it should and i'm receiving the packets. When i try to use it as a service, i cannot seem to get a client connected to my server. Is there something i need to change to deploy this as a service? Code below is from my application: this is my service class where i start protected override void OnStart(string[] args) { server = new TcpServer(); server.StartServer(); } this is the constructor of TcpServer public TcpServer() { try { _server = new TcpListener(IPAddress.Any, 50043); } catch (Exception) { _server = null; } } this is the method i call after initialising the class public void StartServer() { if (_server != null) { // Create a ArrayList for storing SocketListeners before starting the server. _socketListenersList = new ArrayList(); // Start the Server and start the thread to listen client requests. _server.Start(); _serverThread = new Thread(new ThreadStart(ServerThreadStart)); _serverThread.Start(); // Create a low priority thread that checks and deletes client // SocktConnection objcts that are marked for deletion. _purgingThread = new Thread(new ThreadStart(PurgingThreadStart)); _purgingThread.Priority = ThreadPriority.Lowest; _purgingThread.Start(); } } this is the thread that keep checking if any client tries to connect private void ServerThreadStart() { // Client Socket variable; Socket clientSocket = null; TcpSocketListener socketListener = null; while (!_stopServer) { try { // Wait for any client requests and if there is any request from any //client accept it (Wait indefinitely). clientSocket = _server.AcceptSocket(); // Create a SocketListener object for the client. socketListener = new TcpSocketListener(clientSocket); // Add the socket listener to an array list in a thread safe fashon. lock (_socketListenersList) { _socketListenersList.Add(socketListener); } // Start a communicating with the client in a different thread. socketListener.StartSocketListener(); } catch (SocketException se) { _stopServer = true; } } } when for the first time a packet waits to be read, and i get to "clientSocket = _server.AcceptSocket();", it throws an exception (service, not very good debugable) Does anyone recognize this problem or can help me? Thanks in advance

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• How can I send multiple types of objects across Protobuf?

  - by cyclotis04

  I'm implementing a client-server application, and am looking into various ways to serialize and transmit data. I began working with Xml Serializers, which worked rather well, but generate data slowly, and make large objects, especially when they need to be sent over the net. So I started looking into Protobuf, and protobuf-net. My problem lies in the fact that protobuf doesn't sent type information with it. With Xml Serializers, I was able to build a wrapper which would send and receive any various (serializable) object over the same stream, since object serialized into Xml contain the type name of the object. ObjectSocket socket = new ObjectSocket(); socket.AddTypeHandler(typeof(string)); // Tells the socket the types socket.AddTypeHandler(typeof(int)); // of objects we will want socket.AddTypeHandler(typeof(bool)); // to send and receive. socket.AddTypeHandler(typeof(Person)); // When it gets data, it looks for socket.AddTypeHandler(typeof(Address)); // these types in the Xml, then uses // the appropriate serializer. socket.Connect(_host, _port); socket.Send(new Person() { ... }); socket.Send(new Address() { ... }); ... Object o = socket.Read(); Type oType = o.GetType(); if (oType == typeof(Person)) HandlePerson(o as Person); else if (oType == typeof(Address)) HandleAddress(o as Address); ... I've considered a few solutions to this, including creating a master "state" type class, which is the only type of object sent over my socket. This moves away from the functionality I've worked out with Xml Serializers, though, so I'd like to avoid that direction. The second option would be to wrap protobuf objects in some type of wrapper, which defines the type of object. (This wrapper would also include information such as packet ID, and destination.) It seems silly to use protobuf-net to serialize an object, then stick that stream between Xml tags, but I've considered it. Is there an easy way to get this functionality out of protobuf or protobuf-net? I've come up with a third solution, and posted it below, but if you have a better one, please post it too!

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• What Sort of Server Setup Am I Likely to Need? - School A/V streaming

  - by DeathMagus

  My prior experience with servers has generally been limited to home file-sharing servers, low-traffic web-servers, and the like. This leaves me with the technical knowledge of how to set up a system, but little experience in terms of scaling said system. My current project, however, has me as the technical lead in setting up a school for online audio and video streaming. The difficulty I'm running into is that I don't quite have the experience to guess what they'll need, and they don't have the experience to tell me - so I've tried to ask as many pertinent questions about what they want to do with their server, and here's what I found out: About 1000 simultaneous users, and hoping to expand (possibly significantly) Both video and audio streaming, at obviously the highest quality possible Support for both live and playlist-based streaming. Probably only one channel, but as it's an educational opportunity, I imagine letting them have a few more wouldn't hurt. No word on whether they're locked into Windows or whether Linux is acceptable. Approximate budget - $7000. It may actually be about $2k less than this, because of a mishap with another technology firm (they ordered a $7000 DV tape deck for some reason, and now the company wants them to pay a 30% restocking fee). The tentative decisions I've already made: I'm planning on using Icecast 2 for my streaming server, fed by VLC Shoutcast encoding. Since the school already has a DMZ set up, I plan on placing the Icecast server in there, and feeding it through their intranet from a simple workstation computer in their studios. This system isn't in any way mission critical - it's an education tool (they're a media magnet school), so I figure redundancy is not worthwhile to them from a cost:benefit perspective. What I don't know is this: How powerful of a server will I need? What is likely to be my major throttle - bandwidth? How can I mitigate that? Will I need anything special for the encoding workstation other than professional video and audio capture cards and a copy of VLC? Are there any other considerations that I'm simply missing? Thanks a lot for any help - if there's more information you need, let me know and I'll tell you all I can.

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• waiting for 2 different events in a single thread

  - by João Portela

  component A (in C++) - is blocked waiting for alarm signals (not relevant) and IO signals (1 udp socket). has one handler for each of these. component B (java) - has to receive the same information the component A udp socket receives. periodicaly gives instructions that should be sent through component A udp socket. How to join both components? it is strongly desirable that: the changes to attach component B to component A are minimal (its not my code and it is not very pleasent to mess with). the time taken by the new operations (usually communicating with component B) interfere very little with the usual processing time of component A - this means that if the operations are going to take a "some" time I would rather use a thread or something to do them. note: since component A receives udp packets more frequently that it has component B instructions to forward, if necessary, it can only forward the instructions (when available) from the IO handler. my initial ideia was to develop a component C (in C++) that would sit inside the component A code (is this called an adapter?) that when instanciated starts the java process and makes the necessary connections (that not so little overhead in the initialization is not a problem). It would have 2 stacks, one for the data to give component B (lets call it Bstack) and for the data to give component A (lets call it Astack). It would sit on its thread (lets call it new-thread) waiting for data to be available in Bstack to send it over udp, and listen on the udp socket to put data on the Astack. This means that the changes to component A are only: when it receives a new UDP packet put it on the Bstack, and if there is something on the Astack sent it over its UDP socket (I decided for this because this socket would only be used in the main thread). One of the problems is that I don't know how to wait for both of these events at the same time using only one thread. so my questions are: Do I really need to use the main thread to send the data over component A socket or can I do it from the new-thread? (I think the answer is no, but I'm not sure about race conditions on sockets) how to I wait for both events? boost::condition_variable or something similar seems the solution in the case of the stack and boost::asio::io_service io_service.run() seems like the thing to use for the socket. Is there any other alternative solution for this problem that I'm not aware of? Thanks for reading this long text but I really wanted you to understand the problem.

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• L-Soft LISTSERV TCPGUI Interface for PHP Creation

  - by poolnoodl

  I'm trying to use LISTSERV's "API" in PHP. L-Soft calls this TCPGUI, and essentially, you can request data like over Telnet. To do this, I'm using PHP's TCP socket functions. I've seen this done in other languages but can't quite convert it to PHP. I can connect, I can change set ASCII or BINARY mode. But I can never quite craft the header packet the way I need to authenticate, so I'm thinking I'm messing up my conversion. C: http://www.lsoft.com/manuals/16.0/htmlhelp/advanced%20topics/TCPGUI.html#2334328 $origin = '[email protected]'; $pwd = 'password'; $host = "example.com"; $port = 2306; $email = "[email protected]"; $list = "mailinglist"; $command = "Query $list FOR $email"; $fp = stream_socket_client("tcp://$host:$port", $errno, $errstr, 30); $cmd = $command . " PW=" . $pwd; $len = strlen($cmd); $orglen = strlen($origin); $n = $len + $orglen + 1; $headerPacket[0] = "1"; $headerPacket[1] = "B"; $headerPacket[2] = "\r"; $headerPacket[3] = "\n"; $headerPacket[4] = ord($n / 256); $headerPacket[5] = ord($n + 255); $headerPacket[6] = ord($orglen); for ($i = 0; $i < $orglen; $i++) { $headerPacket[$i + 7] = ord($origin[$i]); } for ($i = 0; $i < $len; $i++) { $cmdPacket[$i] = ord($cmd[$i]); } fwrite($fp, implode($headerPacket)); while (!feof($fp)) { echo fgets($fp, 1024); } Any thoughts on where I'm going wrong? I'd much appreciate it if anyone could point me toward some code to do this, days of googling and searching here on SO has only lead me to examples in other languages. Of course, if you know C (or Java or Perl as linked below in my comment to bypass the spam filter), PHP, and socket programming fairly well, you could probably rewrite the whole of the code in an hour, maybe a few minutes. You'd have my eternal thanks for that.

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• How do you unit test the real world?

  - by Kim Sun-wu

  I'm primarily a C++ coder, and thus far, have managed without really writing tests for all of my code. I've decided this is a Bad Idea(tm), after adding new features that subtly broke old features, or, depending on how you wish to look at it, introduced some new "features" of their own. But, unit testing seems to be an extremely brittle mechanism. You can test for something in "perfect" conditions, but you don't get to see how your code performs when stuff breaks. A for instance is a crawler, let's say it crawls a few specific sites, for data X. Do you simply save sample pages, test against those, and hope that the sites never change? This would work fine as regression tests, but, what sort of tests would you write to constantly check those sites live and let you know when the application isn't doing it's job because the site changed something, that now causes your application to crash? Wouldn't you want your test suite to monitor the intent of the code? The above example is a bit contrived, and something I haven't run into (in case you haven't guessed). Let me pick something I have, though. How do you test an application will do its job in the face of a degraded network stack? That is, say you have a moderate amount of packet loss, for one reason or the other, and you have a function DoSomethingOverTheNetwork() which is supposed to degrade gracefully when the stack isn't performing as it's supposed to; but does it? The developer tests it personally by purposely setting up a gateway that drops packets to simulate a bad network when he first writes it. A few months later, someone checks in some code that modifies something subtly, so the degradation isn't detected in time, or, the application doesn't even recognize the degradation, this is never caught, because you can't run real world tests like this using unit tests, can you? Further, how about file corruption? Let's say you're storing a list of servers in a file, and the checksum looks okay, but the data isn't really. You want the code to handle that, you write some code that you think does that. How do you test that it does exactly that for the life of the application? Can you? Hence, brittleness. Unit tests seem to test the code only in perfect conditions(and this is promoted, with mock objects and such), not what they'll face in the wild. Don't get me wrong, I think unit tests are great, but a test suite composed only of them seems to be a smart way to introduce subtle bugs in your code while feeling overconfident about it's reliability. How do I address the above situations? If unit tests aren't the answer, what is? Thanks!

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