I have a VPN configured by ovpn in my mikrotik that already works, but whenever I connect both by android and Windows, I get the error:
ovpn,debug,error,,,,,,,,,l2tp,info,,debug,,,critical,,,,,,,,,,,,,warning duplicate packet, droppin
On console and terminal:
I have already looked for the official forum of mikrotik but neither has no clear answer or a definitive solution.
My goal is to remove this error from my terminal and from my console log.
My current setting:
OVPN SERVER:
PPP PROFILE:
#Edit:
After contacting support, I got this error message to have no impact on the VPN. Below is the email from Mikrotik Support:
Email:
"Hello,
This error message does not have any impact on the VPN connection
establishment, it simply warns you that the client sent duplicate
message which some client software (for example Windows) do.
Best regards, Emils Z."
Thank you all for your help.
I would advise that this is probably a bug in the OVPN implementation on Mikrotik's side. Please log a support ticket with them, and provide the supout.inf file as per the usual process.
Please see this link on how to make the support info file:
https://wiki.mikrotik.com/wiki/Manual:Support_Output_File
In short, open winbox, click the make support file on the left, and then go to files and download the file to your pc. Then attach this file to the support ticket.
Related
I got the above error when try to connect two soft phones which is successfully registered. I was trying to make a voice call in local not connecting and instead returns the error:
chan_sip.c:4274 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
I am using asterisk 13.5.0 and not using freepbx. Simply try to make a call between two peers
Make changes to sip.conf edit the line bind address like bindaddr=0.0.0.0
Likly firewall or permissions(for socket) issues.
Solved
OK I've just solved this issue in my network.
First of all , about the situation : i had an "Issabel VoIP server" based on Asterisk 13 but when i wanted to make calls ,it just failed and i got the same error as yours on Asterisk CLI. I had successful ping requests from the server and extension had been registered without any errors.
Solution: I had doubt about network devices, so i installed 2 soft-phones (like 3cx or Zoiper) on my laptop and connected it straightly to the server using a LAN patch cord ( to see if there's an issue on server side or its related to my network infra.) and registered 2 extensions on my soft-phone and they called each other without problem.so i understood that its one of my network devices that was limiting the transition of VoIP packets between server and extensions . finally i got that it was my TDD-LTE Modem that intercept my connection.
So to make long story short :
1- First create a small network between your server and laptop
2- Register 2 extensions on your soft-phone
3- make call between them and if it goes fine , then look for an obstacle in your main network.
hope it could help you
I have configured bro on my system successfully. OS is centos 7. I have to monotor multimedia traffic e.g. youtube and some social site like facebook. I started bro for some miniutes while using facebook and youtube but their is no information about youtube in http log file nithir facebook. As for I think that this is a protocol problem as facebook use https rather than http but I do not know why youtube.
I have followed following steps after setting correct interface.
[BroControl] > install
Then
[BroControl] > start
But I have not found any youtube or facebook info in http.log. How to get traffic info of such websites?
The problem is that you are expecting SSL encrypted traffic to be magically decrypted and appear in your http.log. If you look again, you will find that YouTube also runs over HTTPS.
Unless you are doing something to intercept and act as a man-in-the-middle for the SSL/TLS connections, you cannot expect to be able to see the content. If you can't see it, Bro can't see it either. :)
If you want to verify that you are properly configured, you would be best served looking at the conn.log to verify that the connections are occurring. Once you do that, search for the UID values in the other logs and I strongly suspect that you will see that you are finding SSL certificate data.
Several things come to mind
1) What are the contents of /usr/local/bro/etc/node.cfg? Make sure it is the interface you expect traffic to cross via a span or tap.
2) Run tcpdump -i <interface> where interface comes from question 1.
3) Run /usr/local/bro/bin/broctl diag to see if there are any issues.
4) Run /usr/local/bro/bin/broctl status to verify everything is running.
If the interface is wrong, the solution may be that easy.
I have faced this issue many times, when I call on my sip line the server responds that the line is busy, though no call is going on when I see with asterisk -vvvr command. When I reload it, it starts working.
Is there a way to troubleshoot this problem or at least get a trigger form some command that sip needs to be reloaded.
Thanks in Advance
Asterisk work for years without reload.
Try update it to latest system or find bug in your config(like no dns, nat settings changed, bad router etc).
Asterisk have no any triggers for such case. You can use external monitoring system(like nagios) which will check sip channel and reload if needed. But better find issue and fix it.
Watch the CLI with a high debug/verbosity to see what's going wrong.
Set verbosity to a high value:
core set verbose 10
Set Debug to a high value:
core set debug 10
You can also check the status of the SIP endpoint with:
sip show peer <extension> (Replace with the number you'd use to dial the endpoint).
If you add verbose CLI output and the output of sip show peer to your question we'll be able to tell you how to fix the issue. Use pastebin or a similar site if the output is very long.
I just went to https://www.sharedrop.io/ and it says my private LAN IP address 192.168.0.3, which is correct. How can it know this? This information isn't sent out via the browser. Or is it?
I'm using Chrome, no extensions installed.
Go here and view the source - http://net.ipcalf.com/ They parse the metadata from the SDP while creating a WebRTC connection. The code is commented with links to the relevant RFCs.
This is probably how ShareDrop does it too.
Edit: ShareDrop is open source and they do this exactly the same way as mentioned above, see https://github.com/cowbell/sharedrop/blob/master/app/scripts/app/controllers/index_controller.js
I want to send a message to everyone(broadcast) in my subnet(LAN) so as to prevent them using Internet due to repair work that's going to happen. How do i do that ? I can't use "wall" because no one is logged on some server.
I want to send a packet that opens a new Tab in web browser and displays message that stop using Internet during a certain duration.
http://en.wikipedia.org/wiki/ARP_spoofing
You might be interested in something like this. I am not sure if this will work in your LAN environment, but typically it should mess up with the entire LAN :D
you can use sockets to send a message to everyone connected in LAN, you can use java and when you want send a message to all clients, every client will see a msgbox with the information you want display....
net send command will be helpful in windows machines. Im not sure about linux.
Plz refer the following link for more Info.
http://www.cezeo.com/tips-and-tricks/net-send-command/
This is a solution shared by my senior( Hope it helps anyone who views this post). What we can do is to do DNS spoofing and redirect everyone's request to a server where you can show the required message.