Asterisk v11.7 Realtime Integration with Kamailio v4.1.X on Ubuntu

Asterisk Realtime Integration with Kamailio ( Asterisk v 11.7.X and Kamailio v 4.1.X )

I have been Googling this and the only link that i found was this :

http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb

Although what the author wrote is good enough to give u a slight perspective on how to do things, Still its written with lot of ambiguous-ness, leaving things very unclear and with lot of issues in Kamailio config file.
Therefore, i decided that i will write a blog myself for all the folks out there who are having issues when they want to use the existing asterisk database or a farm of asterisk server to be configured with kamalio as a PROXY server.

KEY TO USING THIS GUIDE :

Italic - indicate file editing.
Bold Italic - indicate CLI commands.

Starting off - This is written over Ubuntu 12.04 LTS with i386 Arch. - for 64bit Arch the command and the directory structure might change so the person following this must have to adjust accordingly.

Installing Asterisk Server with RDBMS support : 

Install LAMP which is a necessity to start with the asterisk server deployment, the steps are listed as follows :
nash@ubuntu:~$ sudo -i
nash@ubuntu:~#

From now on i will assume you being a root user as everything done here is from root perspective otherwise just keep adding sudo before every command.

# apt-get update && upgrade -y
# apt-get install apache2 -y
# apt-get install mysql-server libapache2-mod-auth-mysql php5-mysql -y

# mysql_install_db ( Initialize the Database )

# /usr/bin/mysql_secure_installation
# apt-get install php5 libapache2-mod-php5 php5-mcrypt
# nano -w /etc/apache2/apache2.conf

Enter anywhere within the file :

ServerName localhost

# service apache2 restart

Congrats LAMP is up and running verify it by placing your IP in your browser to check Apache and also by enter # mysql -u root -p into your CLI to verify the database installation.

mysql > show databases;
mysql > quit;

Now install the dependencies for asterisk server :
# apt-get install -y build-essential linux-headers-`uname -r` openssh-server apache2 mysql-server mysql-client bison flex php5 php5-curl php5-cli php5-mysql php-pear php-db php5-gd curl sox libncurses5-dev libssl-dev libmysqlclient15-dev mpg123 libxml2-dev libnewt-dev sqlite3 libsqlite3-dev pkg-config automake libtool autoconf git subversion uuid uuid-dev -y
# apt-get install libmysqlclient-dev
# apt-get install unixodbc-dev
# apt-get install libmyodbc

Once these are installed we now need to configure Asterisk to use ODBC/Mysql.

Configuring ODBC/Mysql support for Asterisk :

On Ubuntu, the /etc/odbcinst.ini file will be blank, so you’ll need to add the data to that configuration file. Add the following to the odbcinst.ini file:

# nano -w /etc/odbcinst.ini

[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
FileUsage = 1

Verify that the system is able to see the driver by running the following command. It should return the label name MySQL if all is well:

# odbcinst -q -d
[MySQL]

Next, configure the /etc/odbc.ini file, which is used to create an identifier that Asterisk will use to reference this configuration. If at any point in the future you need to change the database to something else, you simply need to reconfigure this file, allowing Asterisk to continue to point to the same place:

# nano -w /etc/odbc.ini

[asterisk-connector]
Description = MySQL connection to 'asterisk' database
Driver = MySQL
Database = asterisk
Server = localhost
UserName = root
Password = password
Port = 3306
Socket = /var/lib/mysql/mysql.sock

We will leave this part here, for now. We will touch this part again as few things are still left. Now we head towards the installation of the Asterisk server.

Installing Asterisk :

The steps are listed as follows:

# cd /usr/src
# wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
# tar xvfz asterisk-11-current.tar.gz
# cd asterisk-*
# ./configure
# make menuselect
# make
# make install
# make config
# make samples

This marks the end of Asterisk installation but the remaining work is to be done with lot of care. Now on with ARA with ODBC/Mysql .
 

Configuring Database to be used by Asterisk : 

Now we go ahead and prepare a database which will be used by asterisk to store user information and etc, interestingly only the database is to be defined whereas the predefined tables comes with every Asterisk Tar file which you will comprehend after looking at the commands i will give further. On we go:

Asterisk ODBC connections are configured in the res_odbc.conf file located in /etc/asterisk. The res_odbc.conf file sets the parameters that various Asterisk modules will use to connect to the database.
Modify the res_odbc.conf file so it looks like the following:

# nano -w /etc/asterisk/res_odbc.conf

[asterisk]
enabled => yes
dsn => asterisk-connector
username => root
password => password
pooling => no
limit => 1
pre-connect => yes

Now once this is done - back to CLI and start putting this :

# mysql -u root -p

mysql >  CREATE DATABASE asterisk;

mysql > quit;

# cd /usr/src/asterisk-11.*/contrib/realtime/mysql
# mysql -u root -p -h localhost asterisk < iaxfriends.sql
# mysql -u root -p -h localhost asterisk < meetme.sql
# mysql -u root -p -h localhost asterisk < musiconhold.sql
# mysql -u root -p -h localhost asterisk < queue_log.sql
# mysql -u root -p -h localhost asterisk < sippeers.sql
# mysql -u root -p -h localhost asterisk < voicemail_data.sql
# mysql -u root -p -h localhost asterisk < voicemail_messages.sql
# mysql -u root -p -h localhost asterisk < voicemail.sql


Verify that your ODBC is able to connect to the database by doing this.

# echo "select 1" | isql -v asterisk-connector

You shud get a response something like this to confirm that it is working correctly :

+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL> +------------+
| ?column?   |
+------------+
| 1          |
+------------+
SQLRowCount returns 1
1 rows fetched

Thats it for the configuration of the database now we move towards the next section that is the configuration of asterisk.

Configuring Asterisk : 

Now we will configure asterisk to make it useable, get ready as still lot of work needs to be done, so here we go :
Uncomment the following two line from extconfig.conf, remember we are working over a basic practical system - how you want to enhance asterisk capability is always up-to u. Hence we move on with our work.

# nano -w /etc/asterisk/extconfig.conf

It will start looking something like this :

sippeers => odbc,asterisk
voicemail => odbc,asterisk

# nano -w /etc/asterisk/sip.conf

Only add the following line to the file :

bindaddr=0.0.0.0

Just above the line :

udpbindaddr=0.0.0.0

Also uncomment the following lines :

tcpenable=yes
tcpbindaddr=0.0.0.0
rtcachefriends=yes

Please bear in mind here that the default port for listening and communication is 5060 and if tcpenable is set to NO turn it to YES. Save and exit , we are nearly there now, be patient, now on with work :

Now we figure out our own Dialplan Configuration:

It is up to you what dialplan you build in /etc/asterisk/extensions.conf. This is a simple and practical configuration, For testing purposes, here is a sample that can be plugged in /etc/asterisk/extensions.conf:

# nano -w /etc/asterisk/extensions.conf

[LocalSets]
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup

It does the classic behavior:

• if phone is registered, route the call to it.
• if phone is unavailable, enter voicemail service.
• if phone is busy, enter voicemail service.

Adding Users into the Database: 

Now we add some user into the database to test our asterisk server. We go back to CLI and type the following stuff :

# mysql -u root -p
mysql > use asterisk;
mysql > INSERT INTO sippeers (name, defaultuser, host, type, context, mailbox, fromdomain, fromuser) VALUES ('101', '101', 'dynamic', 'friend', 'LocalSets', '101', 'ubuntu.local', '101');

mysql > INSERT INTO sippeers (name, defaultuser, host, type, context, mailbox, fromdomain, fromuser) VALUES ('102', '102', 'dynamic', 'friend', 'LocalSets', '102', 'ubuntu.local', '102');

 mysql > INSERT INTO sippeers (name, defaultuser, host, type, context, mailbox, fromdomain, fromuser) VALUES ('103', '103', 'dynamic', 'friend', 'LocalSets', '103', 'ubuntu.local', '103');

Here we have added three users who can use asterisk server facilities, Please bear in mind i have not defined a password in this table, that is because i want to keep this setting as simple as possible, however, if you want to add a password for each user, just do the following:

mysql > UPDATE `sippeers` SET `secret` = '1234' WHERE `name` = '101';

This will set the user 101 password to 1234. I highlighted and pointed out the commands if u want to just use asterisk and not kamailio. This guide will serve as dual purpose. I'm taking the guide in such a manner that if you want to halt at asterisk you only need to follow till this part. As soon as i step into kamailio i will modify asterisk settings accordingly which i will keep highlighting in this tutorial that will follow. Anyways we move ahead with our work and make some voice mailboxes for the users just added earlier.

mysql > INSERT INTO voicemail (context, mailbox, password) VALUES ('default', '101', '1234');
mysql > INSERT INTO voicemail (context, mailbox, password) VALUES ('default', '102', '1234');
mysql > INSERT INTO voicemail (context, mailbox, password) VALUES ('default', '103', '1234');
mysql > quit;

Now we are done. now get ready to execute the Asterisk fully working Server.

Asterisk Execution :

Just hit the following commands and your server will be up and running :

# service mysql restart
# service asterisk restart

Congrats if u followed this tutorial to the dot your asterisk will be up and running, now we will control a bit of asterisk from CLI, this is how its done :

# asterisk -r
ubuntu*CLI> odbc show

This will verify our connection with the database.

ODBC DSN Settings
------------------------------
  Name:   asterisk
  DSN:    asterisk-connector
  Last connection attempt: 1970-01-01 05:00:00
  Pooled: No
  Connected: Yes

Now all you have to do is install any two or three softphone by xlite or ekiga on windows or linux based box and dial there extension, the call will be up and running.
Once two or three users are connected  run the following command to verify your ARA architecture:

# asterisk -r
ubuntu*CLI> sip show users

This will show the users connected to the asterisk server via ODBC/mysql.
This completes the tutorial for Asterisk Server.

Now we will move forward with our Kamailio installation and configuration and integration with Asterisk ARA via ODBC/mysql.

Installing Kamailio:

In order to install kamailio, we need a tar file placed in our /usr/local/src for that we can fetch the source file from its website or we can do it through CLI as follows:

# cd /usr/local/src
# wget http://www.kamailio.org/pub/kamailio/latest/src/kamailio-4.1.3_src.tar.gz
# tar xvfz kamailio-4.1.3_src.tar.gz
# cd kamailio-4.1.3
# make cfg
# gedit module.lst

Add db_mysql to the variable include_modules. It should look like this :

include_modules= db_mysql

Save and continue further :

# make all
# make install

The installation ends here, but please 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

Now once this is done we head towards its configuration.

Kamailio mysql support:

To create the MySQL database, you have to use the database setup script. First edit kamctlrc file to set the database server type:

# nano -w /usr/local/etc/kamailio/kamctlrc

Uncomment the line

SIP_DOMAIN=domain.com

and set it to your server's domain (in my case,it is ubuntu.local), and also most importantly uncomment this line as well :

DBENGINE=MYSQL

You can change other values in kamctlrc file, at least it is recommended to change the default passwords for the users to be created to connect to database.

Once you are done updating kamctlrc file, run the script to create the database used by Kamailio:

# /usr/local/sbin/kamdbctl create

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

The script will add two users in MySQL:

- kamailio - (with default password 'kamailiorw') - user which has full access rights to 'kamailio' database

- kamailioro - (with default password 'kamailioro') - user which has read-only access rights to 'kamailio' database

Adjusting Asterisk for Kamailio:

As highlighted earlier that before kamailio asterisk installation and configuration was done completely independently but now we will adjust asterisk to integrate with kamailio so that our initial target can be achieved.

Open CLI and start putting in these commands i will provide with explanations as why these steps are taken and there impact on our system.

# mysql -u root -p
mysql > use asterisk ; 
mysql > ALTER TABLE `sippeers` ADD `sippasswd` VARCHAR(80) NULL ;
mysql > UPDATE `asterisk.sippasswd` SET `sippasswd` = `secret`;
mysql > UPDATE `asterisk.secret` SET `secret` = NULL ;
mysql > UPDATE `asterisk.sippeers` SET `permit` = '192.168.1.22';
mysql > quit

The explanation is as follows :

- First we logged into the DB system.
- We selected the database to be used.
- We added a new column to sippeers table. This table will be used by kamailio for user authentication as kamailio uses "sippasswd" . It can be left empty if no password is required or filled in according to the administrators choice.
-  All the password that were in the "secret" column is the default column used by asterisk to authenticate users. They are all copied to the "sippasswd" column as now kamailio will take over this responsibility.
- We have emptied the "secret" as we want to avoid dual user authentication first by kamailio and then by asterisk. Making the system redundant and increasing overhead and extra load.
- We have filled the column "permit" with the IP address of Kamailio. YOU SHOULD FILL YOURS. just place the IP address of the machine on which your kamailio is installed.

Now some more tweaking to be done with asterisk:

# service asterisk stop 
# nano -w /etc/asterisk/sip.conf

Find the line

bindaddr=0.0.0.0

modify it like this :

bindaddr=0.0.0.0:5080

Also modify

udpbindaddr=0.0.0.0

to,

udpbindaddr=0.0.0.0:5080

and the final modification to be done is as follows,

tcpbindaddr=0.0.0.0

turn it like this,

tcpbindaddr=0.0.0.0:5080 

We're done with the modification of the asterisk. Now on ahead with our Kamailio modification.

Some Tweaking for Kamailio:

This is the heart of the modification that we will do with our Kamailio installation, I'm providing the entire modified kamailio.cfg all you have to do is discard the orignal one and replace it with this with proper rights to the file.

DONT FORGET TO UPDATE THE "DBURL" AND "DBUSTURL" INCASE YOU CHANGED THE PASSWORDS OR USERNAME OR ANYTHING ELSE. ALSO PLEASE REMEMBER TO CHANGE IP ADDRESSES IN THE FOLLOWING FILE ACCORDING TO YOUR SITUATION I.E. THE MACHINE ON WHICH ASTERISK AND KAMAILIO ARE INSTALLED RESPECTIVELY.

 - I'm highlighting the parts which needs the change,

# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#       as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"  /* change this accordingly */
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://root:password@localhost/asterisk"   /* change this accordingly */
#!endif
#!endif 

#!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.1.22" desc "Asterisk IP Address"                /* change this IP */
asterisk.bindport = "5080" desc "Asterisk Port"
kamailio.bindip = "192.168.1.22" desc "Kamailio IP Address"            /* change this IP */
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif

Explanation :
- the IP address "192.168.1.22" is my server address on which asterisk and kamailio is installed you must change it to your machine otherwise this will not work.

Whats there to be done actually on CLI goes on like this :

# cp /usr/local/etc/kamailio/kamailio.cfg /usr/local/etc/kamailio/kamailio.cfg_orig 
# nano -w /usr/local/etc/kamailio.cfg

Empty the file and paste this entire file that im giving below into it, the configuration file is as follows:

#!KAMAILIO
 
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_ASTERISK
 
#
# Kamailio (OpenSER) SIP Server v4.0 - default configuration script
#     - web: http://www.kamailio.org
#     - git: http://sip-router.org
#
# Direct your questions about this file to: 
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode: 
#     - define WITH_DEBUG
#
# *** To enable mysql: 
#     - define WITH_MYSQL
#
# *** To enable authentication execute:
#     - enable mysql
#     - define WITH_AUTH
#     - add users using 'kamctl'
#
# *** To enable IP authentication execute:
#     - enable mysql
#     - enable authentication
#     - define WITH_IPAUTH
#     - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
#     - enable mysql
#     - define WITH_USRLOCDB
#
# *** To enable presence server execute:
#     - enable mysql
#     - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
#     - define WITH_NAT
#     - install RTPProxy: http://www.rtpproxy.org
#     - start RTPProxy:
#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enable PSTN gateway routing execute:
#     - define WITH_PSTN
#     - set the value of pstn.gw_ip
#     - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
#     - enable mysql
#     - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
#     - enable mysql
#     - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
#     - enable mysql
#     - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
#     - adjust CFGDIR/tls.cfg as needed
#     - define WITH_TLS
#
# *** To enable XMLRPC support execute:
#     - define WITH_XMLRPC
#     - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
#     - adjust pike and htable=>ipban settings as needed (default is
#       block if more than 16 requests in 2 seconds and ban for 300 seconds)
#     - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
#     - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
#     - define WITH_VOICEMAIL
#     - set the value of voicemail.srv_ip
#     - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
#     - enable mysql
#     - define WITH_ACCDB
#     - add following columns to database
#!ifdef ACCDB_COMMENT
  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
 
####### Defined Values #########
 
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
#       as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://root:password@localhost/asterisk"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
 
# - flags
#   FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
 
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
 
####### Global Parameters #########
 
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
 
memdbg=5
memlog=5
 
log_facility=LOG_LOCAL0
 
fork=yes
children=4
 
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
 
/* uncomment the next line to disable the auto discovery of local aliases
   based on reverse DNS on IPs (default on) */
#auto_aliases=no
 
/* add local domain aliases */
#alias="sip.mydomain.com"
 
/* uncomment and configure the following line if you want Kamailio to 
   bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:10.0.0.10:5060
 
/* port to listen to
 * - can be specified more than once if needed to listen on many ports */
port=5060
 
#!ifdef WITH_TLS
enable_tls=yes
#!endif
 
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
 
####### Custom Parameters #########
 
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
 
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
#!endif
 
#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "" desc "VoiceMail IP Address"
voicemail.srv_port = "5060" desc "VoiceMail Port"
#!endif
 
 
#!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.178.25" desc "Asterisk IP Address"
asterisk.bindport = "5080" desc "Asterisk Port"
kamailio.bindip = "192.168.178.25" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
 
####### Modules Section ########
 
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules_k:modules"
#!else
mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"
#!endif
 
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
 
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
 
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
 
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
 
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
 
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
 
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
 
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
 
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
 
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
 
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
 
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
 
#!ifdef WITH_ASTERISK
loadmodule "uac.so"
#!endif
 
# ----------------- setting module-specific parameters ---------------
 
 
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
 
 
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
 
 
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#!ifdef WITH_ASTERISK
modparam("rr", "append_fromtag", 1)
#!else
modparam("rr", "append_fromtag", 0)
#!endif
 
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)
 
 
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra", 
 "src_user=$fU;src_domain=$fd;src_ip=$si;"
 "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
 "src_user=$fU;src_domain=$fd;src_ip=$si;"
 "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
 
 
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "load_credentials", "")
 
#!ifdef WITH_ASTERISK
modparam("auth_db", "user_column", "name")
modparam("auth_db", "password_column", "sippasswd")
modparam("auth_db", "db_url", DBASTURL)
modparam("auth_db", "version_table", 0)
#!else
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "use_domain", MULTIDOMAIN)
#!endif
 
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
 
#!endif
 
 
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- speedial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
 
 
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
 
 
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
 
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
 
 
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
 
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")
 
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
 
 
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
 
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
 
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
 
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
 
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
 
####### Routing Logic ########
 
 
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
 
 # per request initial checks
 route(REQINIT);
 
 # NAT detection
 route(NATDETECT);
 
 # handle requests within SIP dialogs
 route(WITHINDLG);
 
 ### only initial requests (no To tag)
 
 # CANCEL processing
 if (is_method("CANCEL"))
 {
  if (t_check_trans())
   t_relay();
  exit;
 }
 
 t_check_trans();
 
 # authentication
 route(AUTH);
 
 # record routing for dialog forming requests (in case they are routed)
 # - remove preloaded route headers
 remove_hf("Route");
 if (is_method("INVITE|SUBSCRIBE"))
  record_route();
 
 # account only INVITEs
 if (is_method("INVITE"))
 {
  setflag(FLT_ACC); # do accounting
 }
 
 # dispatch requests to foreign domains
 route(SIPOUT);
 
 ### requests for my local domains
 
 # handle presence related requests
 route(PRESENCE);
 
 # handle registrations
 route(REGISTRAR);
 
 if ($rU==$null)
 {
  # request with no Username in RURI
  sl_send_reply("484","Address Incomplete");
  exit;
 }
 
 # dispatch destinations to PSTN
 route(PSTN);
 
 # user location service
 route(LOCATION);
 
 route(RELAY);
}
 
 
route[RELAY] {
 
 # enable additional event routes for forwarded requests
 # - serial forking, RTP relaying handling, a.s.o.
 if (is_method("INVITE|SUBSCRIBE")) {
  t_on_branch("MANAGE_BRANCH");
  t_on_reply("MANAGE_REPLY");
 }
 if (is_method("INVITE")) {
  t_on_failure("MANAGE_FAILURE");
 }
 
 if (!t_relay()) {
  sl_reply_error();
 }
 exit;
}
 
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
 # flood dection from same IP and traffic ban for a while
 # be sure you exclude checking trusted peers, such as pstn gateways
 # - local host excluded (e.g., loop to self)
 if(src_ip!=myself)
 {
  if($sht(ipban=>$si)!=$null)
  {
   # ip is already blocked
   xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
   exit;
  }
  if (!pike_check_req())
  {
   xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
   $sht(ipban=>$si) = 1;
   exit;
  }
 }
#!endif
 
 if (!mf_process_maxfwd_header("10")) {
  sl_send_reply("483","Too Many Hops");
  exit;
 }
 
 if(!sanity_check("1511", "7"))
 {
  xlog("Malformed SIP message from $si:$sp\n");
  exit;
 }
}
 
# Handle requests within SIP dialogs
route[WITHINDLG] {
 if (has_totag()) {
  # sequential request withing a dialog should
  # take the path determined by record-routing
  if (loose_route()) {
   if (is_method("BYE")) {
    setflag(FLT_ACC); # do accounting ...
    setflag(FLT_ACCFAILED); # ... even if the transaction fails
   }
   if ( is_method("ACK") ) {
    # ACK is forwarded statelessy
    route(NATMANAGE);
   }
   route(RELAY);
  } else {
   if (is_method("SUBSCRIBE") && uri == myself) {
    # in-dialog subscribe requests
    route(PRESENCE);
    exit;
   }
   if ( is_method("ACK") ) {
    if ( t_check_trans() ) {
     # no loose-route, but stateful ACK;
     # must be an ACK after a 487
     # or e.g. 404 from upstream server
     t_relay();
     exit;
    } else {
     # ACK without matching transaction ... ignore and discard
     exit;
    }
   }
   sl_send_reply("404","Not here");
  }
  exit;
 }
}
 
# Handle SIP registrations
route[REGISTRAR] {
 if (is_method("REGISTER"))
 {
  if(isflagset(FLT_NATS))
  {
   setbflag(FLB_NATB);
   # uncomment next line to do SIP NAT pinging 
   ## setbflag(FLB_NATSIPPING);
  }
  if (!save("location"))
   sl_reply_error();
 
#!ifdef WITH_ASTERISK
  route(REGFWD);
#!endif
 
  exit;
 }
}
 
# USER location service
route[LOCATION] {
 
#!ifdef WITH_SPEEDIAL
 # search for short dialing - 2-digit extension
 if($rU=~"^[0-9][0-9]$")
  if(sd_lookup("speed_dial"))
   route(SIPOUT);
#!endif
 
#!ifdef WITH_ALIASDB
 # search in DB-based aliases
 if(alias_db_lookup("dbaliases"))
  route(SIPOUT);
#!endif
 
#!ifdef WITH_ASTERISK
 if(is_method("INVITE") && (!route(FROMASTERISK))) {
  # if new call from out there - send to Asterisk
  # - non-INVITE request are routed directly by Kamailio
  # - traffic from Asterisk is routed also directy by Kamailio
  route(TOASTERISK);
  exit;
 }
#!endif
 
 $avp(oexten) = $rU;
 if (!lookup("location")) {
  $var(rc) = $rc;
  route(TOVOICEMAIL);
  t_newtran();
  switch ($var(rc)) {
   case -1:
   case -3:
    send_reply("404", "Not Found");
    exit;
   case -2:
    send_reply("405", "Method Not Allowed");
    exit;
  }
 }
 
 # when routing via usrloc, log the missed calls also
 if (is_method("INVITE"))
 {
  setflag(FLT_ACCMISSED);
 }
}
 
# Presence server route
route[PRESENCE] {
 if(!is_method("PUBLISH|SUBSCRIBE"))
  return;
 
#!ifdef WITH_PRESENCE
 if (!t_newtran())
 {
  sl_reply_error();
  exit;
 };
 
 if(is_method("PUBLISH"))
 {
  handle_publish();
  t_release();
 }
 else
 if( is_method("SUBSCRIBE"))
 {
  handle_subscribe();
  t_release();
 }
 exit;
#!endif
 
 # if presence enabled, this part will not be executed
 if (is_method("PUBLISH") || $rU==$null)
 {
  sl_send_reply("404", "Not here");
  exit;
 }
 return;
}
 
# Authentication route
route[AUTH] {
 
 # if caller is not local subscriber, then check if it calls
 # a local destination, otherwise deny, not an open relay here
 if (from_uri!=myself && uri!=myself)
 {
  sl_send_reply("403","Not relaying");
  exit;
 }
 
#!ifdef WITH_AUTH
 
#!ifdef WITH_ASTERISK
 # do not auth traffic from Asterisk - trusted!
 if(route(FROMASTERISK))
  return;
#!endif
 
#!ifdef WITH_IPAUTH
 if((!is_method("REGISTER")) && allow_source_address())
 {
  # source IP allowed
  return;
 }
#!endif
 
 if (is_method("REGISTER") || from_uri==myself)
 {
  # authenticate requests
#!ifdef WITH_ASTERISK
  if (!auth_check("$fd", "sippeers", "1")) {
#!else
  if (!auth_check("$fd", "subscriber", "1")) {
#!endif
   auth_challenge("$fd", "0");
   exit;
  }
  # user authenticated - remove auth header
  if(!is_method("REGISTER|PUBLISH"))
   consume_credentials();
 }
#!endif
 return;
}
 
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
 force_rport();
 if (nat_uac_test("19")) {
  if (is_method("REGISTER")) {
   fix_nated_register();
  } else {
   fix_nated_contact();
  }
  setflag(FLT_NATS);
 }
#!endif
 return;
}
 
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
 if (is_request()) {
  if(has_totag()) {
   if(check_route_param("nat=yes")) {
    setbflag(FLB_NATB);
   }
  }
 }
 if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
  return;
 
 rtpproxy_manage();
 
 if (is_request()) {
  if (!has_totag()) {
   add_rr_param(";nat=yes");
  }
 }
 if (is_reply()) {
  if(isbflagset(FLB_NATB)) {
   fix_nated_contact();
  }
 }
#!endif
 return;
}
 
# Routing to foreign domains
route[SIPOUT] {
 if (!uri==myself)
 {
  append_hf("P-hint: outbound\r\n");
  route(RELAY);
 }
}
 
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
 # check if PSTN GW IP is defined
 if (strempty($sel(cfg_get.pstn.gw_ip))) {
  xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
  return;
 }
 
 # route to PSTN dialed numbers starting with '+' or '00'
 #     (international format)
 # - update the condition to match your dialing rules for PSTN routing
 if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
  return;
 
 # only local users allowed to call
 if(from_uri!=myself) {
  sl_send_reply("403", "Not Allowed");
  exit;
 }
 
 $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
 
 route(RELAY);
 exit;
#!endif
 
 return;
}
 
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
 # allow XMLRPC from localhost
 if ((method=="POST" || method=="GET")
   && (src_ip==127.0.0.1)) {
  # close connection only for xmlrpclib user agents (there is a bug in
  # xmlrpclib: it waits for EOF before interpreting the response).
  if ($hdr(User-Agent) =~ "xmlrpclib")
   set_reply_close();
  set_reply_no_connect();
  dispatch_rpc();
  exit;
 }
 send_reply("403", "Forbidden");
 exit;
}
#!endif
 
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
 if(!is_method("INVITE"))
  return;
 
 # check if VoiceMail server IP is defined
 if (strempty($sel(cfg_get.voicemail.srv_ip))) {
  xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
  return;
 }
 if($avp(oexten)==$null)
  return;
 
 $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
    + ":" + $sel(cfg_get.voicemail.srv_port);
 route(RELAY);
 exit;
#!endif
 
 return;
}
 
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
 xdbg("new branch [$T_branch_idx] to $ru\n");
 route(NATMANAGE);
}
 
# manage incoming replies
onreply_route[MANAGE_REPLY] {
 xdbg("incoming reply\n");
 if(status=~"[12][0-9][0-9]")
  route(NATMANAGE);
}
 
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
 route(NATMANAGE);
 
 if (t_is_canceled()) {
  exit;
 }
 
#!ifdef WITH_BLOCK3XX
 # block call redirect based on 3xx replies.
 if (t_check_status("3[0-9][0-9]")) {
  t_reply("404","Not found");
  exit;
 }
#!endif
 
#!ifdef WITH_VOICEMAIL
 # serial forking
 # - route to voicemail on busy or no answer (timeout)
 if (t_check_status("486|408")) {
  route(TOVOICEMAIL);
  exit;
 }
#!endif
}
 
#!ifdef WITH_ASTERISK
# Test if coming from Asterisk
route[FROMASTERISK] {
 if($si==$sel(cfg_get.asterisk.bindip)
   && $sp==$sel(cfg_get.asterisk.bindport))
  return 1;
 return -1;
}
 
# Send to Asterisk
route[TOASTERISK] {
 $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
   + $sel(cfg_get.asterisk.bindport);
 route(RELAY);
 exit;
}
 
# Forward REGISTER to Asterisk
route[REGFWD] {
 if(!is_method("REGISTER"))
 {
  return;
 }
 $var(rip) = $sel(cfg_get.asterisk.bindip);
 $uac_req(method)="REGISTER";
 $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport);
 $uac_req(furi)="sip:" + $au + "@" + $var(rip);
 $uac_req(turi)="sip:" + $au + "@" + $var(rip);
 $uac_req(hdrs)="Contact:  + $au + "@"
    + $sel(cfg_get.kamailio.bindip)
    + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
 if($sel(contact.expires) != $null)
  $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";
 else
  $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";
 uac_req_send();
}
#!endif

Save and Exit. The Job is nearly finished.

Final Execution:

Now to execute for whatever we have done just do the following on the CLI, as your asterisk-kamailio system will be up and running.

# service mysql restart
# service asterisk start
# kamctl start

Congrats, you have successfully finished this lengthy tutorial ! the REWARD = your initial goal is now live in action.
Use the same user and password set earlier in asterisk database to connect and achieve VOIP but this time all the sip handling is done by kamailio. Making your system more resilient to the threats of flooding, over loading etc.

Go ahead and type this in your CLI :

# kamctl moni

It will show the entire status of the system. The user adding to the database remains same. Always add the user in asterisk database and the kamailio will load the information all by itself.