The problem is - fake links in e-mails, which are sent by scammers. They look like:
<a href="http://www.ebrug.eu/lnfo/power-SEIIER/portal/" target=
"_blank">http://pages.ebay.de/aboutebay/contact.html</a>
Is there any simple already known solution to discover such fake links using Java? I'm thinking about using regex expressions, filter both links, check if they valid and then to comapare them. Later, if it is fake message, SMTP server will refuse to accept it.
I would like to forward emails from my Lotus Notes inbox to my gmail account.
Lotus Notes rules and agents are disabled on our server, so I developed external application for that.
I am using document.send method and mail successfully arrives to my gmail box.
The only problem is that often the email also duplicated in my Lotus Notes inbox.
I just found that the reason of that is "CC" and "BCC" fields, which I don't clean up,
however, I am looking for the way to forward email as it is - which means keep original CC and BCC and TO fields - exactly on the same way as it is done by forwarding agent.
I am using "IBM Notes 9" on Windows 7 64 bit.
I've prepared a code sample that demonstrates what I am doing.
package com.example;
import lotus.domino.*;
public class TestMailForwarder {
public static void main(String[] args) throws NotesException {
NotesThread.sinitThread();
try {
Session notesSession = NotesFactory.createSession(
(String) null, (String) null, Consts.NOTES_PASSWORD);
DbDirectory dir = notesSession.getDbDirectory(Consts.NOTES_SERVER);
Database mailDb = dir.openDatabaseByReplicaID(Consts.MAILDB_REPLICA_ID);
forwardAllEmails(mailDb);
} finally {
NotesThread.stermThread();
}
}
private static void forwardAllEmails(Database mailDb) throws NotesException {
View inbox = mailDb.getView("$Inbox");
//noinspection LoopStatementThatDoesntLoop
for (Document document = inbox.getFirstDocument();
null != document;
document = inbox.getNextDocument(document)) {
document.send(Consts.GMAIL_ADDRESS);
break;
}
}
}
Instead of trying to send the messages to your GMail, why not upload them using Gmail's IMAP interface. You would require to get the message as MIME content - which probably they are already for external incoming eMails and then push them to GMail.
I don't have a ready code sample, just one for the opposite pulling GMail into Notes, but you should be able to use that as a starting point.
A code sample for the MIME conversion is in an IBM Technote.
Hope that helps
You can't do a transparent forward with code running at the client level. Pure SMTP systems do it by preserving the RFC-822 header content while altering the RFC-821 RCPT TO data. Domino does not give client-level code independent control over these. It just uses the SendTo, CopyTo, and BlindCopyTo items. (There are some tricks that mail management and archiving vendors play in order to do things like this, but they require special changes to the Domino server's router configuration, and software on the other end as well.
Another way of accomplishing this (in response to the question you asked in your comment) would be to have your Java code make a direct connection to the gmail SMTP servers. I'm not sure how easy it is. A comment on this question states that the Java Mail API allows you to control the RCPT TO separately from the RFC822 headers, but I've not looked into the specifics other than taking note that there's an SMTPTransport class -- which is where I'd look for anything related to RFC-821 protocol. The bigger issue is that you will have to take control of converting messages into MIME format. With Notes mail, you may have a mix of Notes rich text and MIME. Theres a convertToMIME method in Notes 8.5.1 and above, but this will only convert the message body. You'll have to deal with any header content separately. (I'm not really up to speed on Notes 9, but AFAIK even though there is functionality in the client to create a .EML file when you drag a message to the desktop, there's no API there to do that for you.)
Finally, I've found a ready solution: AWESYNC.MAIL.
It is a commercial software but it does exactly what I need.
I'm looking for best technique to verify internet over WiFi or Cell. Let's say you are connected to some AP but you don't now if internet exists.
Today I have 2 options by using HttpURLConnection:
Download file from cdn server
to send request to trust web page like google.com and get response 200.
Any other ways?
All suggestions would be appreciated.
On some level, 'ability to access internet endpoints' is equivalent to 'having internet access'. You could consider some algorithm:
for (Endpoint site : theEntireInternet) {
if (can connect to site) return true;
}
return false;
which will conclusively establish connectivity. In other words, being able to connect to any one site is sufficient positive proof, but theoretically you would need to enumerate every site to prove non-connectivity. In practice, checking a few major sites is obviously sufficient. Otherwise (without some sort of meta-information about networks, ISPs, etc; which is unavailable) there's no way to conclusively demonstrate "internet" connectivity other than... connecting to the internet.
Of course as you comment, checking various internet-based applications can't hurt either; it's just a different form of an equivalent technique.
See the Telephony Manager and it's getDataSate() method it has four state
DATA_DISCONNECTED
DATA_CONNECTING
DATA_CONNECTED
DATA_SUSPENDED
You can get the instance of telephony manager --
TelephonyManager tm = (TelephonyManager)getSystemService(Telephony_Service);
Here it is:
to see the state of the data connection on the cell network
and for the WIFI, I would use this public class WifiInfo. From there you can use getIpAddress() or getDetailedStateOf(...)
Hope it's helpful.
I was wondering can anyone point me to a good tutorial on how to construct a SDP message.
I have read the basics and can construct and understand the parameters but I just can't seem to get it to work.
I either get a not acceptable here reply or no reply at all, this is after I get 100 Trying and 180 ringing back.
So my SIP works but it doesn't like the SDP Data.
Its currently constructed like this:
String sdpData = "v=0\r\n"
+ "o=- 019078020 0"
+ " IN IP4 sip.ciceronetworks.com\r\n" + "s=MySession\r\n"
+ "c=IN IP4 sip.ciceronetworks.com\r\n"
+ "t=0 0\r\n" + "m=audio 6002 RTP/AVP 0\r\n"
+ "a=sendrecv\r\n" + "a=rtpmap:0 PCMU/8000\r\n" + "a=ptime:20\r\n"+ "a=fmtp:97 mode=20\r\n";
byte[] contents = sdpData.getBytes();
request.setContent(contents, contentTypeHeader);
And while like that I get 100 trying then 180 ringing but when I accept the call on the other end I get nothing back at all, it seems to just crash, I also get "Audio device Error" on the pc client that I try ringing.
Anyone any ideas?
The issue could be really simple: you seem to forgot the newline after "a=sendrecv". :-)
Anyway, here's an advice:
For testing purposes you are probably better off using a tool rather than jumping right in and writing parts of the protocol. You can use sipp for this purpose, it makes a great tool for testing SIP networks. Other than that you could of course just sniff the network traffic between two working SIP devices and see how it differs from your traffic.
EDIT:
I missed this one before:
You should omit a=fmtp:97 mode=20, as the session description is invalid this way: You may only use the format parameter attribute for codecs that are mentioned in the media line. Codecs are identified via the payload type number (0=PCMU, 8=PCMA, 18=G723, ...). Some codecs don't have officially assigned numbers, for these the dynamic range 96-127 should be used: user agents are free to assign a number in this range via an rtpmap attribute. So, unless you specify which codec you mean by 97, there is no way for the other user agent to know which codec the format parameters should be applied to.
Paprika is right: the a=fmtp:97 mode=20 is simply wrong (and looks like it's part of an iLBC codec offer). You didn't offer codec 97, you offered codec 0 (PCMU).
Note that the a=fmtp:97 shouldn't hurt you, it's just spurious.
The most likely problem is that you are not sip.ciceronetworks.com - i.e. your c= line (and m= line) said "send my media to port 6002 at sip.ciceronetworks.com". I suspect your PC's IP address is not the same as sip.ciceronetworks.com, and/or there's a firewall/NAT between you and the other end.
It probably isn't your problem, but the o= line is wrong per the spec
From RFC 4566:
o=<username> <sess-id> <sess-version> <nettype> <addrtype> <unicast-address>
Getting VoIP to work is not as simple as the RFCs or cookbook explanations would imply....
I found a good article which deals with SDP (Session Description Protocol). It is also in relationship with an SDK which is called Ozeki VoIP SIP SDK. If you combine these things you will be able to create a softphone for instance.
There is also a brief overall about SDP.
Working with SDP in VoIP SIP calls is an interesting topic for those who wish to develop their own softphone or webphone application or what you desire.
SDP describes multimedia communication session for the purposes of session announcement, session invitation, and parameter negotiation.
The usage of an SDK can take a lot of burden from one's shoulder bucasue flexibility and high compatibility is assured.
For more information regarding SDP in connection with an SDK to build own applications you can have a look at the mentioned article if you Google for: "Working with SDP in VoIP SIP calls"
I've been writing a little application that will let people upload & download files to me. I've added a web service to this applciation to provide the upload/download functionality that way but I'm not too sure on how well my implementation is going to cope with large files.
At the moment the definitions of the upload & download methods look like this (written using Apache CXF):
boolean uploadFile(#WebParam(name = "username") String username,
#WebParam(name = "password") String password,
#WebParam(name = "filename") String filename,
#WebParam(name = "fileContents") byte[] fileContents)
throws UploadException, LoginException;
byte[] downloadFile(#WebParam(name = "username") String username,
#WebParam(name = "password") String password,
#WebParam(name = "filename") String filename) throws DownloadException,
LoginException;
So the file gets uploaded and downloaded as a byte array. But if I have a file of some stupid size (e.g. 1GB) surely this will try and put all that information into memory and crash my service.
So my question is - is it possible to return some kind of stream instead? I would imagine this isn't going to be terribly OS independent though. Although I know the theory behind web services, the practical side is something that I still need to pick up a bit of information on.
Cheers for any input,
Lee
Yes, it is possible with Metro. See the Large Attachments example, which looks like it does what you want.
JAX-WS RI provides support for sending and receiving large attachments in a streaming fashion.
Use MTOM and DataHandler in the programming model.
Cast the DataHandler to StreamingDataHandler and use its methods.
Make sure you call StreamingDataHandler.close() and also close the StreamingDataHandler.readOnce() stream.
Enable HTTP chunking on the client-side.
Stephen Denne has a Metro implementation that satisfies your requirement. My answer is provided below after a short explination as to why that is the case.
Most Web Service implementations that are built using HTTP as the message protocol are REST compliant, in that they only allow simple send-receive patterns and nothing more. This greatly improves interoperability, as all the various platforms can understand this simple architecture (for instance a Java web service talking to a .NET web service).
If you want to maintain this you could provide chunking.
boolean uploadFile(String username, String password, String fileName, int currentChunk, int totalChunks, byte[] chunk);
This would require some footwork in cases where you don't get the chunks in the right order (Or you can just require the chunks come in the right order), but it would probably be pretty easy to implement.
When you use a standardized web service the sender and reciever do rely on the integrity of the XML data send from the one to the other. This means that a web service request and answer only are complete when the last tag was sent. Having this in mind, a web service cannot be treated as a stream.
This is logical because standardized web services do rely on the http-protocol. That one is "stateless", will say it works like "open connection ... send request ... receive data ... close request". The connection will be closed at the end, anyway. So something like streaming is not intended to be used here. Or he layers above http (like web services).
So sorry, but as far as I can see there is no possibility for streaming in web services. Even worse: depending on the implementation/configuration of a web service, byte[] - data may be translated to Base64 and not the CDATA-tag and the request might get even more bloated.
P.S.: Yup, as others wrote, "chuinking" is possible. But this is no streaming as such ;-) - anyway, it may help you.
I hate to break it to those of you who think a streaming web service is not possible, but in reality, all http requests are stream based. Every browser doing a GET to a web site is stream based. Every call to a web service is stream based. Yes, all. We don't notice this at the level where we are implementing services or pages because lower levels of the architecture are dealing with this for you - but it is being done.
Have you ever noticed in a browser that sometimes it can take a while to fetch a page - the browser just keeps cranking away showing the hourglass? That is because the browser is waiting on a stream.
Streams are the reason mime/types have to be sent before the actual data - it's all just a byte stream to the browser, it wouldn't be able to identify a photo if you didn't tell it what it was first. It's also why you have to pass the size of a binary before sending - the browser won't be able to tell where the image stops and the page picks up again.
It's all just a stream of bytes to the client. If you want to prove this for yourself, just get a hold of the output stream at any point in the processing of a request and close() it. You will blow up everything. The browser will immediately stop showing the hourglass, and will display a "cannot find" or "connection reset at server" or some other such message.
That a lot of people don't know that all of this stuff is stream based shows just how much stuff has been layered on top of it. Some would say too much stuff - I am one of those.
Good luck and happy development - relax those shoulders!
For WCF I think its possible to define a member on a message as stream and set the binding appropriately - I've seen this work with wcf talking to Java web service.
You need to set the transferMode="StreamedResponse" in the httpTransport configuration and use mtomMessageEncoding (need to use a custom binding section in the config).
I think one limitation is that you can only have a single message body member if you want to stream (which kind of makes sense).
Apache CXF supports sending and receiving streams.
One way to do it is to add a uploadFileChunk(byte[] chunkData, int size, int offset, int totalSize) method (or something like that) that uploads parts of the file and the servers writes it the to disk.
Keep in mind that a web service request basically boils down to a single HTTP POST.
If you look at the output of a .ASMX file in .NET , it shows you exactly what the POST request and response will look like.
Chunking, as mentioned by #Guvante, is going to be the closest thing to what you want.
I suppose you could implement your own web client code to handle the TCP/IP and stream things into your application, but that would be complex to say the least.
I think using a simple servlet for this task would be a much easier approach, or is there any reason you can not use a servlet?
For instance you could use the Commons open source library.
The RMIIO library for Java provides for handing a RemoteInputStream across RMI - we only needed RMI, though you should be able to adapt the code to work over other types of RMI . This may be of help to you - especially if you can have a small application on the user side. The library was developed with the express purpose of being able to limit the size of the data pushed to the server to avoid exactly the type of situation you describe - effectively a DOS attack by filling up ram or disk.
With the RMIIO library, the server side gets to decide how much data it is willing to pull, where with HTTP PUT and POSTs, the client gets to make that decision, including the rate at which it pushes.
Yes, a webservice can do streaming. I created a webservice using Apache Axis2 and MTOM to support rendering PDF documents from XML. Since the resulting files could be quite large, streaming was important because we didn't want to keep it all in memory. Take a look at Oracle's documentation on streaming SOAP attachments.
Alternately, you can do it yourself, and tomcat will create the Chunked headers. This is an example of a spring controller function that streams.
#RequestMapping(value = "/stream")
public void hellostreamer(HttpServletRequest request, HttpServletResponse response) throws CopyStreamException, IOException
{
response.setContentType("text/xml");
OutputStreamWriter writer = new OutputStreamWriter (response.getOutputStream());
writer.write("this is streaming");
writer.close();
}
It's actually not that hard to "handle the TCP/IP and stream things into your application". Try this...
class MyServlet extends HttpServlet
{
public void doGet(HttpServletRequest request, HttpServletResponse response)
{
response.getOutputStream().println("Hello World!");
}
}
And that is all there is to it. You have, in the above code, responded to an HTTP GET request sent from a browser, and returned to that browser the text "Hello World!".
Keep in mind that "Hello World!" is not valid HTML, so you may end up with an error on the browser, but that really is all there is to it.
Good Luck in your development!
Rodney