i have my first question after years.
i configured a remote syslog to send events to my syslogs event.
now i want to resend this events to a siem via syslog but i can't.
I see some events in var/log/messages the are not sending to the siem
### end of the forwarding rule
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I'm implementing hot update for my service behind a proxy. During the update process my service goes into a quiescent state for some time so I want to buffer the gRPC messages received during this time so that I can process them later (after update). Does envoy or Nginx proxy provide such a feature?
Story: In a client-server system I use a long time connection(an http(s) request from client to server with a long time timeout) in order to use notify client to do some actions(Most of data transfer is from client to server but some commands send to client in response of this http(s) request).
Problem: If client cancel the connection server can understand that but if internet connection of client loses(e.g, unplug the LAN cable or it loses the WLAN/GPRS antenna) neither client nor server understand this. Connection still remains until (some time spends and) somebody writes something in it which is too late.
PS: 0) I googled with AKC/NACK, Keep-alive, ping-pong and heartbeat key words for http(s) request and could not find a protocol which it periodically check the status of request.
1) In this you can find an argument for curl command which sets a time interval to send a props(also I monitored this with wireshark). But if still if you unplug the cable neither curl command nor server can understand the connection lost.
curl -k --keepalive-time 5 https://exampel.com/v1/v/f9a64e73/notification
2) Also here explains that there is an http header which is used to use a connection multiple time.
In server side and with nginx web-server we can enable TCP keep-alive probe with so_keepalive=on as listen input argument. Find more information in this link.
Setup
180.87.13.77 ===Router=========GRE TUNNEL=========ROUTER=====Internet===Webserver
The client .77 cannot open few web sites.
We did a packet capture and noticed in all the TCP sessions
after HTTP GET is requested, Server is Sending RST/ACK
THis terminate the session.
No idea on what is happening.Wireshark Capture
I have a mirth channel that listens to a source and then deploys the inbound communications to several channels. One of these channels sends the HL7 to an application I am developing, and I do not know where to send my ack message. Should I send it to the inbound port of the original message, or does MIRTH have a specific process for sending acknowledgements?
Acknowledgements in Mirth are handled in destination's Response Transformer. To get there, go to the Destinations tab, select your destination if there are more than one, under Channel Tasks menu on the left side select Edit Response. The msg variable there is your response message. To generate an acknowledgement use Postprocessor script or place your acknowledgement into responseMap directly and configure the Response setting of the Source connector.
I am working on a sip client - asterisk server. I am using tcp connections.
The client side is Zoiper as for a first test.
Registration and outbound calls do work as expected, but after 3-4 minutes from registration process or an outgoing call, when testing incoming calls I do get this message on the server:
tcptls.c:446 ast_tcptls_client_start: Unable to connect SIP socket to ip:port: Connection timed out
The invite message (incoming call) never gets on the client (Zoiper softphone).
Why is this error showing up?
The reason why this appears from my assumption is because of the fact that neither the client or the server are sending keep alive messages, so after a tcp socket timeout the client which is behind a nat will not be reachable from the server side anymore.
This error come because your NAT (or 3g if you use 3g) drop connection. As result there are no way use same connection anymore.
Correct behavour of you app - send SIP OPTIONS message, if timeout - do registration again.
And yes, you need send keepalives(recomended method - OPTIONS message) or setup keepalive on asterisk side and setup in your side correct answer.