I'm working on IPMI protocol which communicates with a BMC board present in a server. I've got a doubt which basically is that can we get all the server related info(majorily boot options and reseting the system) through IPMI protocol using its commands(including RAW commands) ?
I tried to search it on internet but I didn't get much info (even on oracle's website) !
So if the answer is yes, please redirect me to the resource.
What I want is clear elaboration of the commands so that I can make use of it.
Here you are, the IPMI Spec.
http://www.intel.com.tw/content/www/tw/zh/servers/ipmi/ipmi-second-gen-interface-spec-v2-rev1-1.html
For an example:
ipmitool bmc info
Equal to ipmitool raw 0x06 0x01
0x06 is Network Function Code
0x01 is commad of NetFn
Find section,Get Device Command, to parse response data
Related
For an application I had to develop a simple telnet module. (Which I could do quickly with the help of minimalistic telnet)
I did create a telnet server and a client and I used PuTTY to test the behavior of the server.
PuTYY was configured for 'Telnet negotiation mode: active'.
In the server I did see the expected telnet negotiation sequence (and it was giving correct replies).
The last action in the PuTTY negotiation was a strange one.
0xff 0xfe 0x32 or IAC DONT 0x32
The third byte, 0x32, is the option and that is where the problem is (for me).
I did look up all Telnet options at IANA and according to them the options range 50-137 is unassigned (0x32 is 62).
What is behind PuTTY option 0x32?
I have looked on the internet but can't find anything.
I have looked up the PuTTY documentation, same thing.
Putting up a request to the PuTTY development is severely discouraged due to capacity problems (not enough hands to type a sensible reply).
It is perfectly sensible not to use this unknown option but it still is nagging me.
What is option 0x3e supposed to do?
Problem solved.
There was an error in my server code causing PuTTY to generate a reply for a non existing option.
I hereby thank Simon Tatham for the suggestion he gave leading me to find the real problem.
I was trying to save a SPA packet created via the fwknop-client on a client in client-server architecture.
The command that I have used is as follows
"fwknop -A tcp/22 -D server-ip --key-gen --use-hmac --save-packet --save-packet-file filename.pkt --save-rc-stanza -vv"
The command executes successfully but I'm not able to save the packet, the packet is not found in the system.
I also trying to append the packet using the "--save-packet-append" tag, but still not able to get the output.
The purpose for doing to above is to obtain the SPA packet and append the client certificate (asymmetric encryption) to this SPA packet which will be sent.
How can i save this packet to fulfil my purpose ?
Thank You
I am trying to do External authentication on SMART CARD, using SCP03. The value of P1 is the security level that we are using for the authentication.
May I know the possible values of the security levels and the data I need to pass for them?
If i understand your question correctly, there are three security level for command apdu:
none: nothing is secure
mac: command is clear but mac of command is calculated and added to the command
enc: command is encrypted and mac of command added to thecommand
there are another two option for response security:
rmac: like mac for response apdu
renc: like enc for response apdu
I suggest you to study Globalplatform specification:
https://www.globalplatform.org/specificationscard.asp
Currently, I am working on Telit UL865-EUD 3G Modem. I worked with Telit GL865 GPRS modem before that's why I am familiar with AT commands and initialization logic.
My problem is that my UL865 modem always returns "NO CARRIER" to ATD*99***1# command. It is not only specific to ATD*99***1#. It results also same with other ATD Commands.
UL865 module is working. I attached to the network and open TCP socket. I tested them. But Dial Commands are not working.
My AT Command Log output is in below
Firmware Version : 12.00.616
AT
OK
AT&K=0
OK
AT#SCFG=3,1,300,600,300,10
OK
AT#SLED=2,2,2
OK
AT#SLEDSAV
OK
AT#FRWL=1,"1.2.3.4","0.0.0.0"
OK
AT+CGDCONT=1,"IP","mgbs"
OK
AT+CGSN
355855052854914
OK
AT#ICMP=2
OK
AT+CSQ
+CSQ: 24,2
OK
AT+CCLK?
+CCLK: "17/10/05,18:55:06+12"
OK
AT#SGACT=1,1,"mgbs",""
#SGACT: 5.***.***.***
OK
AT+COPS?
+COPS: 0,0,"TURKCELL",2
OK
AT+CREG?
+CREG: 0,1
OK
AT#USERID=""
OK
AT#PASSW=""
OK
AT#GAUTH=0
OK
AT#GPPPCFG="000.000.000.000"
OK
ATD*99***1#
NO CARRIER
Is there anybody who faces the same problem before?
Before the ATD command, try send:
AT+CUSD=1
This command in the Telit 3G Modules AT Commands Reference Guide:
allows control of the Unstructured Supplementary Service Data
Note that you are sending an USSD command (e.g. ATD*100#).
I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the call gets hung up).
This is my setup:
What I see:
In the CLI:
[2015-11-24 01:01:53] NOTICE[43619][C-00000002]: res_rtp_asterisk.c:4441 ast_rtp_read: Unknown RTP codec 95 received from '(null)'
In Firefox:
InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute
Attachments:
SIP Dialogue (Asterisk CLI)
Webphone Log
Config Files (httpd.conf, sip.conf, rtp.conf)
Asterisk Compiled with Libuuid & Friends
What I've tried so far:
Changed webRTC implementations (tried chrome and firefox both with SIPML and SIP.JS)
Set the STUN server to null on the client side (stunServers: ['stun:null'])
Configured properly (I hope) my sip.conf and rtp.conf and httpd.conf
Made sure I have libuuid, uuid and their -devel companions and after i've recompiled asterisk.
What I've read:
http://forums.asterisk.org/viewtopic.php?p=201702
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
http://jssip.net/documentation/misc/interoperability/asterisk/
http://sipjs.com/guides/server-configuration/asterisk/
https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/
http://forums.digium.com/viewtopic.php?f=1&t=89798
Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.
Faced same issue and followed instruction in http://forums.digium.com/viewtopic.php?f=1&t=90167 realise that:
This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).
I did recompile my Astersik 11.16.0 with patch for ECDH support and fallback to prime256v1 https://issues.asterisk.org/jira/browse/ASTERISK-25265 and looks like lost uuid support at that time. Reverting back to non-patched version (with uuid support, use to be compiled before) resolved my issue with "no ice-ufrag attribute" error in Firefox console and calls are going well now from WebRTC client SIPML5 based to asterisk, but not in opposit direction