FreeBSD portupgrade missing key: categories: Cannot read the portsdb! SOLUTION!!!
No commentswhen the error is like this:
# portupgrade phpMyAdmin
[missing key: categories] [Updating the portsdb
in /usr/ports ... - 20687 port entries found .........1000.........2000.........3000.........4000.........5000.........6000.........7000.........8000.........9000.........10000.........11000.........12000.........13000.........14000.........15000.........16000.........17000.........18000.........19000.........20000...... ..... done]
missing key: categories: Cannot read the portsdb!
/usr/local/lib/ruby/site_ruby/1.8/portsdb.rb:567:in `open_db’: database file error (PortsDB::DBError)
from /usr/local/lib/ruby/site_ruby/1.8/portsdb.rb:736:in `port’
from /usr/local/lib/ruby/site_ruby/1.8/portsdb.rb:924:in `all_depends_list’
from /usr/local/lib/ruby/site_ruby/1.8/pkgdb.rb:915:in `tsort_build’
from /usr/local/lib/ruby/site_ruby/1.8/pkgdb.rb:907:in `each’
from /usr/local/lib/ruby/site_ruby/1.8/pkgdb.rb:907:in `tsort_build’
from /usr/local/lib/ruby/site_ruby/1.8/pkgdb.rb:929:in `sort_build’
from /usr/local/lib/ruby/site_ruby/1.8/pkgdb.rb:933:in `sort_build!’
from /usr/local/sbin/portupgrade:694:in `main’
from /usr/local/lib/ruby/1.8/optparse.rb:755:in `initialize’
from /usr/local/sbin/portupgrade:210:in `new’
from /usr/local/sbin/portupgrade:210:in `main’
from /usr/local/sbin/portupgrade:1981 Read the rest of this entry »
Skype protocol, skype connect algorithm, authentication servers and ports
No commentsHere because I dont wonna a forget it.
Protocol
A Skype network is a peer-to-peer network with three main entities: supernodes, ordinary nodes and the login server. It is an overlay network: each client builds and refreshes a list of reachable nodes known as the host cache. The host cache contains IP address and port numbers of supernodes. Communication is encrypted using RC4; the method used does not provide any privacy but instead merely obfuscates the traffic.
So, abbreviatures for host cache is HC , Skype Client is SC and Skype Network is SN
Asterisk 1.4 tunning tips and shits
No commentsIP Type Of Service settings:
——————————————-
sip.conf tos_sip cs3
sip.conf tos_audio ef
sip.conf tos_video af41
——————————————-
iax.conf tos ef
——————————————-
iaxprov.conf tos ef
===========================================
Asterisk SIP Response Codes
No commentsa list.
1xx: Provisional — request received, continuing to process the request;
Provisional responses, also known as informational responses, indicate that the server contacted is performing some further action and does not yet have a definitive response. A server sends a 1xx response if it expects to take more than 200 ms to obtain a final response. Note that 1xx responses are not transmitted reliably. They never cause the client to send an ACK. Provisional (1xx) responses MAY contain message bodies, including session descriptions.
2xx: Success — the action was successfully received, understood, and accepted;
3xx: Redirection — further action needs to be taken in order to complete the request;
4xx: Client Error — the request contains bad syntax or cannot be fulfilled at this server;
5xx: Server Error — the server failed to fulfill an apparently valid request;
6xx: Global Failure — the request cannot be fulfilled at any server.
Nokia N95 connect SIP to asterisk
No commentsFast and easy you can connect your Nokia N95 gsm to asterisk PBX.
( it’s work with almost all Nokia cell phones )
Go to Tools / Settings / Connection / SIP settings
Create new profile …
Profile name: stolitel
Enter name for your SIP profile.
Service Profile: IETF Default
Do Not change this.
Default Access Point: ( your WIFI AP name -> find it in Tools / Settings / Connections / Wireless LAN )
Choose the access point name.
Public User Name: 1000@sip.stolitel.net
Fill your username and @your registar server address.
Use Compression: No
Do Not Change this.
Registration: “Always on ”
Do Not Change this.
Use Security: No
Do Not Change this.
Proxy Server: none
Do Not Enter in this setting menu.
Registrar Server .-> Enter to edit this setting menu.
Registrar Server address: sip.stolitel.net
Enter your Server address.
Realm: asterisk
Do Not Change This unless there is realm: in server’s sip.conf
User Name: 1000
Enter your SIP username.
Password: password
Enter your SIP password.
Transport Type: UDP
Do not change or change it to UDP.
Port: 5060
Do not change OR enter your registar server SIP connection port.
Now your gsm phone is a gsm and VoIP phone in the same time.
When you want to call via VoIP , just enter the calling number, hit Options -> Internet Call
Asterisk on FreeBSD: install, configure and run. command line examples.
No commentsFirst we get and install the asterisk package file.
#cd /usr/porta/net/asterisk
#make install asterisk clean clean-depends
…………………………………………………………………………………..
INSTALLING…….( , If you exit with an error, just fix it and continue. )
………………………………………………………………………………..
and we love pkg worlds! 
LibClamAV Error: Can’t load /var/db/clamav/main.cvd: MD5 verification error - SOLUTION!!!
No commentsThe Problem:
ClamAV, refuse to start with error:
LibClamAV Warning: ***********************************************************
LibClamAV Warning: *** This version of the ClamAV engine is outdated. ***
LibClamAV Warning: *** DON’T PANIC! Read http://www.clamav.net/support/faq ***
LibClamAV Warning: ***********************************************************
LibClamAV Error: Can’t load /var/db/clamav/main.cvd: MD5 verification error
ERROR: MD5 verification error
Solution:
1. first do update on clamav package
smtp# portupgrade clamav
—> Upgrading ‘clamav-0.94.2′ to ‘clamav-0.95.2′ (security/clamav)
—> Building ‘/usr/ports/security/clamav’
===> Cleaning for clamav-0.95.2
===> Vulnerability check disabled, database not found
===> Found saved configuration for clamav-0.94.2
=> clamav-0.95.2.tar.gz doesn’t seem to exist in /usr/ports/distfiles/.
=> Attempting to fetch from http://superb-east.dl.sourceforge.net/sourceforge/clamav/.
fetch: http://superb-east.dl.sourceforge.net/sourceforge/clamav/clamav-0.95.2.tar.gz: Moved Temporarily
=> Attempting to fetch from http://nchc.dl.sourceforge.net/sourceforge/clamav/……..
Now when starting clamav it’s says:
LibClamAV Error: Can’t load /var/db/clamav/main.cvd: Can’t verify database integrity
ERROR: Can’t verify database integrity
2. second move main.cvd database from /var/db/clamav
smtp# mv main.cvd main.cvd.bak
and do new database update, so freshclam can create the right main.cvd
smtp# freshclam
Current working dir is /var/db/clamav
Max retries == 3
ClamAV update process started at Tue Aug 11 11:22:47 2009
Using IPv6 aware code
Querying current.cvd.clamav.net
TTL: 700
Software version from DNS: 0.95.2
THAT’S ALL!
smtp# /usr/local/etc/rc.d/clamav-clamd start
Starting clamav_clamd.
HOWTO install,config and run Asterisk on MacOSX in 3 easy steps
No commentsRealLife example
No Screenshots and media at all. Simpy line-by-line on commandline howto.
Step.1 Get Asterisk 4 MacOSX Leopard
#cd /; wget http://asterisk.vlessert.nl/installasterisk.tar.gz && tar zxvf installasterisk.tar.gz
#sudo ./installasterisk.sh
..In the end I receive this error:
launchctl: Couldn’t stat(”/Library/LaunchDaemons/org.asterisk.asterisk.plist”): No such file or directory
nothing found to load…This is becouse we need ot manual setup launchd startup script. I will not do that, but you can do it, follow steps explained in here -> Building+Asterisk+on+MacOSX
#sudo asterisk
Zaptel, Asterisk PBX, Nokia GSM - solution example on Debian 5.02
No comments
OK, this is an solution example with very simple configs. Everything is working propertly and clearly.
Just anouther line-by-line on command-line .
So.. We have fresh and new installed Debian 5.02 i386 .
#uname -a
Linux aster 2.6.26-2-686 #1 SMP Sun Jun 21 04:57:38 UTC 2009 i686 GNU/Linux
BUT That is all what we have + HIGHSPEED INTERNET CONNECION:)
So… this will be the same one like the others. commands on the commandline.
Step.1 Making our asterisk machine more stable. (Twiks , Tricks and all what you know, you do it now)
updating, upgrading and removing for not using components. In brief look you do it all the Debian way apt-get ,dpkg…. you know.
Step.2 Preparing the system with drivers and modules and tools needed for the components and drivers which we gonna use.
#apt-get install zlib1g-dev libssl-dev bison debhelpe
#apt-get install zaptel-source zaptel fxload libtonezone1
Step.3 Installing Zaptel first. Zaptel 1.4.11.
Whit installing I mean compiling drivers.
#module-assistant prepare
… there is a lot of work that Debian work here, but when ends, it’stime for zaptel
#module-assistant auto-install zaptel
…becouse we have some hardware installed , we run:
#genzaptelconf -svdM
...if dialog appers, just answer the questions..country code..etc. So we have /etc/zaptel.conf and we will edit it to look like this:
#vim /etc/zaptel.conf
# Span 1: WCTDM/0 “Wildcard TDM400P REV I Board 1″ (MASTER)
fxsks=1
fxsks=2
# Span 2: ZTDUMMY/1 “ZTDUMMY/1 (source: HRtimer) 1″
# Global data
loadzone = us
defaultzone = us
….And this is becouse I have one TDM with two fxsko modules linked to analog telecom line.
On more page: Installing Asterisk, Cofiguring SIP and calls, Connectiong Nokia with VoIP
Howto config Nokia N95 Sip to Asterisk 1.4
No commentsBefore starting the gsm phone SIP Client config settings, we assume that there is configured sip.conf and Astersik at all.
Start.
1. You have to link a.k.a. connect your phone to an internet. Mostly offcourse through Wifi AccessPoints.
2. Enter Menu->Tools->Settings->Sip Settings
3. Enter Settings:
Profile name : Name for the settings profile ( not connected in any way with Asterisk)
Service profile : IETF
Access point by default : YourAccessPoint_name (this one with the internet connection)
Public user name : sip:your_sip_username@your_sip_server_ip
Use the compression : No
Registration : I Always shoose ALways ACTIVE ( but there is no connection with Asterisk do you choose what you want )
Use security : No
Proxy Server -> NO CONFIGURATION HERE ( off course it’s completly compatible with ProxyServer if you want to, you can put your here)
Registrar server: -> From Here You Register Into Sip Server a.k.a. Asterisk
Server address registrar : sip:your_sip_server_ip
Realm : asterisk (ot else, you can specify it in sip peer configuration)
User name : your_sip_username
Password : your_secret_password (secret= into sip.conf)
Transport type : UDP
Port : 5060