fileinsert's questions

I am trying to manipulate how Rsyslog sends data to a syslog server on an embedded system that runs Rsyslog version 5. As it is an embedded system there is no option to upgrade to a later version without vendor involvement.

I have the following template:

 $template precise,": %HOSTNAME%-dev123: %timegenerated%: %syslogfacility-text%-%syslogpriority%-%syslogtag%%msg%"

In the above example, the text '123' is a serialised asset number that identifies the device. This number is stored in a file and I would like to dynamically fetch this rather than hard-code it to make the rsyslog config more universal and portable within our environment.

As it is pre-version6, advanced templates are not supported that would allow for command substition.

Is there a method available in this version of rsyslog to allow me to do this?

I have two computers running Windows 10 Pro 1903 and have recently created a shared folder on one.

I initially tried to connect using the IP address and when trying to access the share it displays a message saying:

"You do not have permission to access \\ip\share."

I then tried to connect using the system name and it successfully connected with no issues.

The system event viewer reports a successful login on both occasions. Sharing is configured with password-protected sharing.

Although I got it working, I want to understand why it didn't work without the name. It seems like something has changed in windows networking, and so far my travels of the googlesphere has come back with nothing.

Is there something in Windows 10 network shares that prevents shares accessed directly via IP address?

I have some clients connected to my Asterisk server behind a NAT device. When using TCP everything works OK. If I change the client and server config to use UDP (from transport=tcp to transport=udp,tcp or even simply transport=udp) the phone can no longer register and Asterisk sends SIP: SIP/2.0 401 Unauthorized to the client.

From what I have read it is possible to use UDP behind a NAT with Asterisk, however experience seems to dictate otherwise. Have I missed another mandatory step or is there something else that I am unaware of?

To clarify: it is the clients behind the NAT device; not the server.

Here's the relevant portions of my sip.conf:

[general]
context=bogus
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
videosupport=yes
canreinvite=no
nat=force_rport,comedia
session-timers=refuse
localnet=x.x.x.x/255.255.0.0     ; blanked out for serverfault
tcpenable=yes                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0:5060         ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp,tcp

[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
;        allow=ilbc
;        allow=gsm
;        allow=g723
        allow=alaw
        allow=ulaw
;       allow=g722

[basic-options](!,my-codecs)                ; another template
        dtmfmode=rfc2833
        context=internal
        type=friend


[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic
        transport=tcp,udp

[101](natted-phone)
callerid="Office"
secret=12345678901234567890
context=internal
mailbox=101@main
transport=udp,tcp

UPDATE: On investigation I have found that this only seems to affect Avaya handsets. Other softphones connected to the same server do not seem to suffer this problem. Here is a sample of register and un authorized packets:

asterisk*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:x.x.x.x:1032 --->
REGISTER sip:asterisk SIP/2.0
From: sip:2003@asterisk;tag=46c2abf4589053ec5895ffc0_F2003192.168.0.174
To: sip:2003@asterisk
Call-ID: [email protected]
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.0.174;branch=z9hG4bK1_5a7d12149bcad5895ffc4_R2003
Content-Length: 0
Max-Forwards: 70
Contact: <sip:[email protected];avaya-sc-enabled;transport=udp>;q=1;expires=900;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-2cf4c54ef19b>"
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,MESSAGE,INFO,PUBLISH,REFER,UPDATE
User-Agent: Avaya one-X Deskphone
Supported: eventlist

<------------->
--- (12 headers 0 lines) ---
Sending to x.x.x.x:1032 (NAT)
Sending to x.x.x.x:1032 (NAT)

<--- Transmitting (NAT) to x.x.x.x:1032 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.174;branch=z9hG4bK1_5a7d12149bcad5895ffc4_R2003;received=x.x.x.x;rport=1032
From: sip:2003@asterisk;tag=46c2abf4589053ec5895ffc0_F2003192.168.0.174
To: sip:2003@asterisk;tag=as5f37e30b
Call-ID: [email protected]
CSeq: 1 REGISTER
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2350c222"
Content-Length: 0