The Guru Of Tech Gobind

[Command]# cat /etc/redhat-release

Keeping the default 22 SSH port is really not a good practice. I had seen many people do not bother about changing their SSH port even they configured many security features.

This article applies for Ubuntu/Debian/Centos/Redhat O/S

Note- Mess with others things on SSH config file is really harmful and it may lock your server from accessing via SSH. So you will need console access to reset SSH or contact server provider to do this. So do with your own risk!

Steps

Use NANO editor for edit files or vi editor. If you dont have NANO installed then you can use vi editor or you can install NANO by

apt-get install nano or yum install nano

1.You need to open SSHD config file using nano.

nano /etc/ssh/sshd_config

2.Thats your SSHD config file. On that there will be line called

Port 22

In sip.conf under [general] add a register definition:

Format:
register => user[:secret[:authuser]]@host[:port][/extension]

or
register => fromuser@fromdomain:secret@host

or
register => fromuser@fromdomain:secret:authuser@host:port/extension

Example1:
register => username:pwd:username@216.144.238.155/extension

; Register 2345@mysipprovider.com with authname 2345 at mysipprovider.com at sip provider as 1234 here:
register => 2345:password@mysipprovider.com/1234

; Register 1235@mysipprovider.com with authname 1235 and realm mysipprovider.com at proxyof.mysipprovider.com as s here.
; (as needed by budgetphone.nl):
register => 1235@mysipprovider.com:password:1235@proxyof.mysipprovider.com

Templates are a means to avoid repetitive sections in Asterisk configuration files
like extensions.conf, sip.conf or iax.conf. Templates are not to be confused with
macros in extensions.conf.

For template configurations, the syntax for defining a section is as follows:

[section](options)
label = value

The options field is used to define templates, refer to templates and hide
templates. Any object can be used as a template.

No whitespace is allowed between the closing "]" and the parenthesis "(".

Adding to an existing section
(this is actually not template-related, but still an interesting method)

[section]
label = value

[section](+)
label2 = value2

In this case, the plus sign indicates that the second section (with the
same name) is an addition to the first section. The second section can
be in another file (by using the #include statement). If the section
name referred to before the plus is missing, the configuration will fail
to load.

Defining a template-only section

[section](!)
label = value

The exclamation mark indicates to the config parser that this is a only
a template and should not itself be used by the Asterisk module for
configuration. The section can be inherited by other sections (see section
"Using templates" below) but is not used by itself.

Using templates (or other configuration sections)

[section](name[,name])
label = value

The name within the parenthesis refers to other sections, either
templates or standard sections. The referred sections are included
before the configuration engine parses the local settings within the
section as though their entire contents (and anything they were
previously based upon) were included in the new section.

Example 1

[tpl_ext](!)                ; a template
    dtmfmode=rfc2833
    context=from-internal
    type=friend
    disallow=all
    allow=gsm
    qualify=yes
    host=dynamic

[1000](tpl_ext)
secret=1000
callcounter=yes                  ; Enable call counter
busylevel=1

Example 2

[foo]
permit=192.168.0.2
host=asdf
deny=192.168.0.1

[bar]
permit=192.168.1.2
host=jkl
deny=192.168.1.1

[baz](foo,bar)
permit=192.168.3.1
host=bnm

The [baz] section will be processed as though it had been written in the
following way:

[baz]
permit=192.168.0.2
host=asdf
deny=192.168.0.1
permit=192.168.1.2
host=jkl
deny=192.168.1.1
permit=192.168.3.1
host=bnm

Example 3
In top-level sip.conf:

[defaults](!)
type=friend
nat=yes
qualify=on
dtmfmode=rfc2833
disallow=all
allow=alaw

include accounts/*/sip.conf

In accounts/customer1/sip.conf:

[def-customer1](!,defaults)
secret=this_is_not_secret
context=from-customer1
callerid=Customer 1 <300>
accountcode=0001

[phone1](def-customer1)
mailbox=phone1@customer1

[phone2](def-customer1)
mailbox=phone2@customer1

Change the Linux User Root Password

su to root
Type "passwd" (without quotes)

It will then prompt you to type your new pass, and then to confirm it.

If you are desperate because you have forgotten root password, don’t be. This tutorial will show you how to reset root.

Please note that you cannot perform this remotely, you must be physically at the machine.

1. Step : Reboot The Server

2. Step : Look for the centos blue screen saying booting press DOWN arrow key

3. Step : Select any Kernel and Press “ e ”

4. Step : Select Kernel /vmlinuz and Press “ e ”

5. Step : Type “ single “ and Press Enter

6. Step : You Will see the kernel /vmlinuz selected Press “ b ” Server will reboot

7. Step : After Rebooting the server you will get the Shell Prompt and Type Command “ passwd ”

8. Step : Type password “ newpassword “

9. Step : Reboot server using command “ init 6”

10. You Are Done!!

[root@123]#

mysqlcheck --auto-repair --databases database-name -uUser -pPass (for {INNODB} and MYSQL service should be running)

And

[root@123]#

myisamchk /var/lib/mysql/database-name/*.MYI ( for {MyISAM} and MYSQL service should be stopped)

HOW TO CHANGE NAME OF AUTHOR in Blogger

1.Log in to your dashboard--> template- -> Edit HTML

2.Click on "Expand Widget Templates"

3.Scroll down to where you see this:

<span class='post-author vcard'><b:if cond='data:top.showAuthor'><data:top.authorLabel/><span class='fn'><data:post.author/></span></b:if></span>

4.Replace <data:post.author/> of above code with your new name.

5.Click on "Save Templates" and Refresh your site.

This is a sample dialplan that will act as a plug-in replacement for AgentCallbackLogin

for Asterisk 1.6's.

It offers the following features:

- Single log-on and log-off, managing queue/agent associations centrally

- Agent pause (with pause reason) and unpause

- Managed device state - the queue will not try calling members that are busy

on other queues or because of other calls

- Tracking of the current hot-desk extension

- Fully compatible with QueueMetrics

- Very compact and easy to expand

It requires Asterisk 1.6.1 or newer to run.

SETTING UP

----------

First, make sure that the extension that you plan to use in sip.conf have

callconter=yes

busylevel=1

so that when one call is being made or received, the telephone state is BUSY.

Then define your queues in queues.conf (see file here for an example configuration).

For this example we define two queues aclled "q1" and "q2".

We leave the file agents.conf completely alone (agent/queue associations are managed

in the dialplan).

Import the file "agent16_queuemetrics.conf" in the main extension.conf file and create

"hooks" that your agents will dial (see examples in the extensions.conf file included).

Edit "agent16_queuemetrics.conf" to define which agents work on which queues (see below).

Included hooks:

- 820 -> Agent login. It will ask you the agent code and the current extension (as in SIP/ext)

- 821 -> Agent log off. It will ask you the current agent number.

- 823 -> Agent pause. It will autodetect the current agent based on the extension and will

ask you for a pause code. Enter # if none.

- 824 -> Agent unpause. It will autodetect the current agent.

Then at 827/828 you see an example of routing calls to each queue. Set up queues as you best

see fir in your dialplan.

Enjoy!

DEFINING WHICH AGENTS WORK ON WHICH QUEUES

------------------------------------------

Edit the file agent16_queuemetrics.conf where it says:

[macro-xagents-allqueues]

....

exten => s,n,Macro(xagents-associate,${ARG1},${ARG2},${ARG3},q1,100:101:103:105)

....

As you can see, here we say that for "q1" we have agents 100, 101, 103 and 105.

You must separate agents using a semicolon.

Add a line for each queue that you have defined in your dialplan.

SETTING UP QUEUEMETRICS

-----------------------

QueueMetrics requires no special setup:

1. make sure that default.rewriteLocalChannel=true (this should be the current default anyway)

2. manually assign agents to the queues

++++++++++++++++++++++++++++++++++++++++++++

QUEUS.CONF

++++++++++++++++++++++++++++++++++++++++++++

[general]

persistentmembers = yes

keepstats = no

autofill = yes

monitor-type = MixMonitor

shared_lastcall=no

[q1]

music=default

strategy=rrmemory

eventwhencalled=no

timeout=15

retry=1

wrapuptime=0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

autopause=no

autofill=yes

joinempty=yes

[q2]

music=default

strategy=rrmemory

eventwhencalled=no

timeout=15

retry=1

wrapuptime=0

maxlen=0

announce-frequency = 0

announce-holdtime=no

autopause=no

autofill=yes

joinempty=yes

++++++++++++++++++++++++++++++++++++++++++++

SIP.CONF

++++++++++++++++++++++++++++++++++++++++++++

[tpl_grandstream](!) ; a template

dtmfmode=rfc2833

context=from-internal

type=friend

disallow=all

allow=gsm

qualify=yes

host=dynamic

[999](tpl_grandstream)

secret=999

callcounter=yes ; Enable call counter

busylevel=1

++++++++++++++++++++++++++++++++++++++++++++

agents16_queuemetrics.conf

++++++++++++++++++++++++++++++++++++++++++++

[xagents]

exten => _.,1,Set(realchan=${DB(Xagent/${EXTEN})})

exten => _.,n,NoOP("Trying Xagent/${EXTEN} as ${realchan}")

exten => _.,n,Dial(${realchan},60,tT)

exten => h,1,NoOp

[macro-xagents-associate]

;

; ARG1 : operation: ADD or REM

; ARG2 : Agent ID

, ARG3 : Device to be used as state for agent ID

; ARG4 : queue name

, ARG5 : queue members, as 100:103:104

;

exten => s,1,NoOp("Macro2")

exten => s,n,Set(queue=${ARG4})

exten => s,n,Set(agentlist=:${ARG5}:)

exten => s,n,Set(agent=${ARG2})

exten => s,n,Set(oper=${ARG1})

exten => s,n,Set(devstate=${ARG3})

exten => s,n,Set(fnd=${REGEX(":${agent}:" ${agentlist})})

exten => s,n,GotoIf(${fnd}?found)

exten => s,n,NoOP( "NOTFOUND" )

exten => s,n,MacroExit

exten => s,n(found),Goto(op-${oper})

exten => s,n,MacroExit

exten => s,n(op-ADD),AddQueueMember(${queue},Local/${agent}@xagents,,,,${devstate})

exten => s,n,MacroExit

exten => s,n(op-REM),RemoveQueueMember(${queue},Local/${agent}@xagents)

exten => s,n,MacroExit

[macro-xagents-allqueues]

; AGR1: operation: ADD or REM

; ARG2: Agent ID

; ARG3: Device to be used as a source of state

exten => s,1,NoOP("")

exten => s,n,Macro(xagents-associate,${ARG1},${ARG2},${ARG3},q1,100:101:103:105)

exten => s,n,Macro(xagents-associate,${ARG1},${ARG2},${ARG3},q2,101:100:103:105)

exten => s,n,Macro(xagents-associate,${ARG1},${ARG2},${ARG3},q3,108:101:103:105)

exten => s,n,MacroExit

[xagents-logon]

exten => s,1,Answer

exten => s,n,Read(agent,agent-user)

exten => s,n,Read(phone,agent-newlocation)

exten => s,n,Set(DB(Xagent/${agent})=SIP/${phone})

exten => s,n,Set(DB(Xagent_byphone/${phone})=${agent})

exten => s,n,QueueLog(NONE,${UNIQUEID},Local/${agent}@xagents,AGENTCALLBACKLOGIN,${phone})

exten => s,n,Macro(xagents-allqueues,ADD,${agent},SIP/${phone})

exten => s,n,Playback(agent-loginok)

exten => s,n,Hangup

[xagents-logoff]

exten => s,1,Answer

exten => s,n,Read(agent,agent-user)

exten => s,n,Set(chan=${DB(Xagent/${agent})})

exten => s,n,Set(DB(Xagent/${agent})=)

exten => s,n,Macro(xagents-allqueues,REM,${agent},-)

exten => s,n,QueueLog(NONE,${UNIQUEID},Local/${agent}@xagents,AGENTCALLBACKLOGOFF,)

exten => s,n,Playback(agent-loggedoff)

exten => s,n,Hangup

[xagents-pause]

exten => s,1,Answer

exten => s,n,Set(agent=${DB(Xagent_byphone/${CALLERID(num)}})

exten => s,n,Read(code,agent-user)

exten => s,n,PauseQueueMember(,Local/${agent}@xagents,,${code})

exten => s,n,QueueLog(NONE,${UNIQUEID},Local/${agent}@xagents,PAUSEREASON,${code})

exten => s,n,Hangup

[xagents-unpause]

exten => s,1,Answer

exten => s,n,Set(agent=${DB(Xagent_byphone/${CALLERID(num)}})

exten => s,n,UnpauseQueueMember(,Local/${agent}@xagents)

exten => s,n,Hangup

++++++++++++++++++++++++++++++++++++++++++++

EXTENSIONS.CONF

++++++++++++++++++++++++++++++++++++++++++++

#include "agents16_queuemetrics.conf"

[from-internal]

exten => 820,1,Dial(Local/s@xagents-logon)

exten => 821,1,Dial(Local/s@xagents-logoff)

exten => 823,1,Dial(Local/s@xagents-pause)

exten => 824,1,Dial(Local/s@xagents-unpause)

exten => 827,1,Answer

exten => 827,n,Queue(q1)

exten => 828,1,Answer

exten => 828,n,Queue(q2)

----------------------

YOU ARE DONE!!!

The support for multi-WLM is not enabled by default, but can be easily turned on and enable with registry hack.

Run Registry Editor (regedit).
Navigate to the following registry key: KEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows Live\Messenge
For 64-bit (x64) OS, go to following registry branch instead:
HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Microsoft\Windows Live\Messenger
In the right pane, right click on any blank space, select New on context menu, and then click on DWORD (32-bit) Value.
Name the new registry value entry as MultipleInstances.
Modify the MultipleInstances and set its value data as 1.
Start as many Windows Live Messenger windows as you want from Start Menu.

Adding User Accounts in MYSQL

CREATE USER 'sam'@'localhost' IDENTIFIED BY 'some-password';

GRANT ALL PRIVILEGES ON *.* TO 'sam'@'localhost' WITH GRANT OPTION;

FLUSH PRIVILEGES; (telling the server to reload the grant tables using flush command)

The accounts created by these statements have the following properties:

Two of the accounts have a user name of sam and a password of some_pass. Both accounts are superuser accounts with full privileges to do anything. The 'sam'@'localhost' account can be used only when connecting from the local host. The 'sam'@'%' account uses the '%' wildcard for the host part, so it can be used to connect from any host.

SELECT * FROM table_name INTO OUTFILE 'C:/new_file.xls'

FIELDS TERMINATED BY ',' ENCLOSED BY '"'

LINES TERMINATED BY '\n';

If getting an error in asterisk cli (1.6.1.8)

-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected

then use the following command in asterisk cli

cli>core set verbose 0

Shows the CREATE TABLE statement that creates the given table. The statement requires the SELECT privilege for the table. As of MySQL 5.0.1, this statement also works with views.

mysql> SHOW CREATE TABLE tablename

#check partition sizes df -h

#check directory size du -s -h /var/log/

#check every directory and file sizes under a dir.

du -s -h /var/log/*

#check individual size size du -s -h /var/log/lastlog

mail -s .ANY SUBJECT. emailid@gmail.com => return Key

body message here => ctrl+d for CC return key

to check the logs:

cd /var/log
tail -100 maillog

Set the current date to May 7 2011 8:42:45pm.

#date -s "7 May 2011 20:42:45"

wget http://pkgs.fedoraproject.org/repo/pkgs/asterisk/asterisk-1.6.2.19.tar.gz/5dfb9352bacfa7f4dc2ef72cb83715c2/asterisk-1.6.2.19.tar.gz
tar -zxvf asterisk-1.6.2.19.tar.gz
cd asterisk-1.6.2.19
./configure
make
make install (Do not delete G729 module)
asterisk -vvvgc
asterisk -vr

http://www.blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/

Download from here: (Depend on 32bit or 64bit)["i686" means 32-bit. If it was 64-bit, it would be "x86_64"] Check by "uname -a" command
wget http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-pentium.so
(codec_g729-ast16-gcc4-glibc-pentium4.so)

Notice that we are in the modules directory

Rename the codec file
mv codec_g729-ast14-gcc4-glibc-pentium4.so codec_g729.so
Restart Asterisk
/etc/init.d/asterisk restart
Check your work
Connect to Asterisk

asterisk -r
Then show all of the codec translations available using the command below, and check g729 at the left, you should see that it can be translated to many other codecs on the top of the table

Machine*CLI> show translations

**********

After that install mysql module in a proper directory

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.2.3.tar.gz
tar -zxvf asterisk-addons-1.6.2.3.tar.gz
cd asterisk-addons-1.6.2.3
./configure
make clean
make
make install

First :

At starting of the file add this lines

#ifndef AST_MODULE
#define AST_MODULE "app_voicemail"
#endif

AFTER THAT COMPILE THE MODULE

cd /usr/src/asterisk-1.6.1.8/apps/

gcc -shared app_voicemail.c -o app_voicemail.so

cp app_voicemail.so /usr/lib/asterisk/modules/

CLI> restart now

tethereal -i eth -w ~/mycapture.pcap

go to /etc/my.cnf

If you are using MySQL 4.0 or newer, enter:

[mysqld]
# Allow packets up to 16M (change its value)
max_allowed_packet=16M

If you are using an older version of MySQL, enter:
[mysqld]
# Allow packets up to 16M
set-variable = max_allowed_packet=16M

To store 8 MB objects (default is 16M).

Go to Admin-> Sysconfig-> search for PostmasterMaxEmails

LoopProtection!!! Send no more emails to 'emailid@.com'! Max. count of 40 has been reached!

PostmasterMaxEmails 40 Default

Change its value from 40 to whatever you want

To add the option on NOTE for changing the state of ticket, go to:

Frontend::Agent::Ticket::ViewNote
Then --> Ticket::Frontend::AgentTicketNote###State: "YES" --> Update

After adding this, you will have the otion to change the state of a ticket
under notes.

Run the below command

SELECT table_schema "Data Base Name", sum( data_length + index_length) / 1024 / 1024
"Data Base Size in MB" FROM information_schema.TABLES GROUP BY table_schema ;

In /etc/my.cnf

server-id=2

replicate-same-server-id = 0

auto-increment-increment = 2

auto-increment-offset = 2

replicate-do-db = databasename

Mysql> Show slave status;

Above command will show the status of replication. we see two column -- value for both should be 'Yes'.

a. SlaveIORunning -- 'Yes'
b.SlaveSQLRunning --'Yes'

Incase of replication failure value for the columns will be 'No'.

If value is 'No' look for the error. Note the logfile and logposition being read.
Stop slave. First Diagnose error and resync replication from the log position it broke.

Deleting Binlogs of mysql. ----->

Log into Secondary Server

mysql> show slave status;
Above command will give the file name being read by slave machine.

Go to master and execute--->
mysql> purge binlogs to 'file_name_got_from_above_command'

In case of master to master replication we have to delete binlog from both machines. Do the same as above on both machines.

My.ini/My.cnf configuration for Master
-------------------------------

log-bin=mysql-bin
server-id=1

--------------------------------

INNODB Specific options
--------------------------------
innodb_flush_log_at_trx_commit=1
sync_binlog=1

--------------------------------

Restart MySql Service

FLUSH PRIVILEGES;
-----------------------------------------

mysql>USE exampledb;

mysql>FLUSH TABLES WITH READ LOCK;

Take a backup (or MYSQLDUMP) of the DB to be replicated

Restore DB on Slave

************************************************** *********************************

Step 2 – Configure the Slave Server(Windows XP)

Edit the my.ini/my.cnf

server-id=2
replicate-do-db = exampledb

Step 3 – Restart Mysql Service

Restart mysql Service

mysql > Stop slave;

mysql>CHANGE MASTER TO MASTER_HOST='master_IP', MASTER_USER='user', MASTER_PASSWORD='password',MASTER_LOG_FILE='mysql-bin.000006',MASTER_LOG_POS=183;

mysql > Start slave;

mysql > show slave_status; [This command will show whether your replication is running fine.]

To know if your replication is working check output of above command.
IN Out put of above command value of these two running status should be yes.
Slave_IO_Running | Slave_SQL_Running
Yes | Yes |

If any of these is NO than your replication is not working. To resynchronize DB and start replication again contact your MYSQL DBA.

Install And Maintain Kamailio (OpenSER) v3.0.x From GIT on CentOS 5.x

Main author:

Elena-Ramona Modroiu <ramona (at) asipto.com>

This is a step by step tutorial about how to install and maintain Kamailio (OpenSER) using the sources from GIT on CentOS 5.x.

Using the GIT, you get access to the latest code very quickly.

This document focuses on Kamailio (OpenSER) v3.0.x with MySQL support, using CentOS 5.5 system - should works more or less in the same way with CentOS 5.3 or 5.4.

Kamailio 3.0.0 was released on January 11, 2010:

release notes are available here
what is new:

http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x

Prerequisites

GIT

You need git 1.5.x at least and might not be available on your CentOS installation, if it is too old. In case git is not available on your yum repository, then install Fedora repository for enterprise release.

On CentOS i386, do:

rpm -Uvh http://download.fedora.redhat.com/pub/epel/5/i386/epel-release-5-3.noarch.rpm

On CentOS x86_64, do:

rpm -Uvh http://download.fedora.redhat.com/pub/epel/5/x86_64/epel-release-5-3.noarch.rpm

Then install suing yum:

yum install git

Of course, as alternative, you can install it from sources, download it from:

http://git-scm.com/

Then compile and make install.

Dependencies

gcc
make
flex
bison
mysql-server
mysql-client
mysql-devel

Download sources

Create the directory on the file system where to store the sources.

mkdir -p /usr/local/src/kamailio-3.0

cd /usr/local/src/kamailio-3.0

Download the sources from GIT using the following commands.

git clone --depth 1 git://git.sip-router.org/sip-router kamailio

cd kamailio

git checkout -b kamailio_3.0 origin/kamailio_3.0

Installation

Generate the config for Makefile system and include MySQL connector module for installation:

make include_modules="db_mysql" cfg

If you want other modules which are not enabled by default to be compiled, add them in include_modules, separated by whitespace.

Compile and install:

make all

make install

Installation details

The binaries and executable scripts were installed in:

/usr/local/sbin

These are:

kamailio - Kamailio (OpenSER) server
kamdbctl - script to create and manage the Databases
kamctl - script to manage and control Kamailio (OpenSER) server
sercmd - CLI - command line tool to interface with Kamailio (OpenSER) server

To be able to use the binaries from command line, make sure that '/usr/local/sbin' is set in PATH environment variable. You can check that with 'echo $PATH'. If not and you are using 'bash', open '/root/.bash_profile' and at the end add:

PATH=$PATH:/usr/local/sbin

export PATH

Kamailio (OpenSER) modules are installed in:

/usr/local/lib/kamailio/modules/

/usr/local/lib/kamailio/modules_k/

The documentation and readme files are installed in:

/usr/local/share/doc/kamailio/

The man pages are installed in:

/usr/local/share/man/man5/

/usr/local/share/man/man8/

The configuration file was installed in:

/usr/local/etc/kamailio/kamailio.cfg

MySQL database

To create the MySQL database, you have to use the database setup script:

/usr/local/sbin/kamdbctl create

In order to create the database you need to specify the wanted db type (DBENGINE=MYSQL) in the

/usr/local/etc/kamailio/kamctlrc

file. Call this script without any parameter to get some help for the usage. You will be asked for the domain name Kamailio (OpenSER) is going to serve (e.g., mysipserver.com) and the password of the 'root' MySQL user. The script will create a database named 'openser' containing the tables required by Kamailio (OpenSER). You can change the default settings in the kamctlrc file mentioned above.

The script will add two users in MySQL:

- openser - having the password 'openserrw', user which has full access rights to 'openser' database

- openserro - having the password 'openserro', user which has read-only access rights to 'openser' database

Do change the passwords for these two users immediately after the database is created.

Configuration file

To fit your requirements for the VoIP platform, you have to edit the configuration file.

/usr/local/etc/kamailio/kamailio.cfg

Follow the instruction in the comments to enable usage of MySQL. Basically you have to add several lines at the top of config file, like:

#!define WITH_MYSQL

#!define WITH_AUTH

#!define WITH_USRLOCDB

init.d script

The init.d script can be used to start/stop the Kamailio (OpenSER) server in a nicer way. A sample of init.d script for Kamailio (OpenSER) is provided at:

copy the init scripts :

#cp /usr/local/src/kamailio-3.1.2/pkg/kamailio/rpm/kamailio.init /etc/init.d/kamailio
#chmod 755 /etc/init.d/kamailio
#cp /usr/local/src/kamailio-3.1.2/pkg/kamailio/rpm/kamailio.default /etc/default/kamailio

#vi /etc/default/kamailio and set RUN_KAMAILIO=yes

#vi /etc/init.d/kamailio and update :
KAM=/usr/local/sbin/kamailio

/usr/local/etc/kamailio/kamailio.cfg edit this file to fit your requirements.

For WEB-Interface please install siremis http://techgobind.blogspot.com/2011/11/siremis-open-source-web-management.html

SIREMIS: Open Source Web Management Interface for SIP Routing Engines:

Kamailio (OpenSER)
SIP-Router.org

Download

Latest version of SIREMIS 2.x is available at:

http://siremis.asipto.com/pub/downloads/siremis/
it includes everything, so you do not need to download something extra

Untar the archive on the local file system (e.g., in your web server root folder):

cd /var/www
tar xvfz siremis-x.y.z.tgz

Web Server Setup

If you want to setup an alias for Apache 2.x, you can run 'make apache-conf' and you get the config snippet printed:

siremis-2.0# make apache-conf

# siremis apache conf snippet ...

Alias /siremis "/var/www/siremis-2.0/siremis"

Options Indexes FollowSymLinks MultiViews

AllowOverride All

Order allow,deny

Allow from all

Order deny,allow

Deny from all

</FilesMatch>

Order deny,allow

Deny from all

</FilesMatch>

</Directory>

You can copy&paste it to Apache conf file (e.g., for Apache2 on Debian/Ubuntu: /etc/apache2/sites-available/default).

This will make SIREMIS available under http://yourwebserver.com/siremis/. If you want to use a different alias, edit Makefile and change the value of URLBASE

Local Configuration

Next step is to create .htaccess and Siremis configuration files, you have to run 'make prepare':

siremis-2.0# make prepare

updating htaccess file...

updating app.inc file...

done

Make sure that following directories have write access for web server user:

siremis/log
siremis/session
siremis/files
siremis/themes/default/template/cpl

Do not forget to restart the web server after doing the changes in its configuration file.

Database Configuration

Database server to be used is MySQL.

You have to create a MySQL user that has access to Siremis database, for example:

GRANT ALL PRIVILEGES ON siremis.* TO siremis@localhost IDENTIFIED BY 'siremisrw';

This user will be needed later during the installation, of course, you can use an existing MySQL user that has enough privileges.

Also, you have to create Kamailio database with kamdbctl tool, if you don't have it already:

kamdbctl create

You don't need to do anything else to setup the database of Siremis 2.0, the rest of the process is down via web installation wizard.

Web Installation Wizard

Once you have done the steps above, access the web page:

http://yourwebserver.com/siremis/

Step 1

First step shows the status of required items. You have to update local configuration until all the items are checked ok.

Step 2

In the second step you have to introduce the details to access Kamailio and Siremis database.

When you install Siremis first time, you probably have to check:

Create Siremis Database - this create database 'siremis'
Import Default Data - this imports initial data into database 'siremis'
Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules
Replace DB Config - this updates Siremis configuration file with the details to access databases

Post Installation

remove the install directory: siremis/install

If you want to run again installation wizard:

restore directory: siremis/install
remove file: siremis/install.lock

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