Tuesday, July 16, 2019

Capturing Network Traces From The CLI On CUCM

Here are the givens:
CUCM Publisher IP: 10.5.0.10  and Target ISR Gateway: 172.16.0.1
Step 1: Connect to the CUCM Host
Using an SSH client like putty/secureCRT connect to the  CUCM publisher.
—————-
lexter@lextertech.com /
$ ssh admin@10.5.0.10
admin@10.5.0.10’s password:
Last login: Tue Jul 16 19:53:00 2019 from 192.168.1.1
Welcome to the Platform Command Line Interface
admin:
————–
Step 2: Prepare a Capture Trace File
From the admin: prompt, you will use the capture network utility to setup a protocol trace.  You can do this live and watch it on the screen if you are simply trying to determine that a conversation is happening or you can capture the output to a file.  We are going to do the latter.
———-
admin: utils network capture file mycap count 100000 size all host all 172.16.0.1
Executing command with options:
size=all count=100000 interface=eth0
src= dest= port=
ip=172.16.0.1
———–
The trace continues until you press Ctrl-C.
Step 3: Download the Trace File
OK, so now you have a file somewhere on the system with the data that you would really like to view.  You have to download it to your machine.  You will need a running SFTP server on the target machine.  You can use openSSH on linux/unix, freeFTPd, copSSH, or something similar.
——-
admin:file get activelog platform/cli/mycap.cap
The size of the file and other information is sent to the output screen.
You are prompted for the host name: use DNS or IP address of SFTP server
You are prompted for the port: Defaults to 22 (SSH)
The CUCM checks its local keys and then asks for your user ID and password on the SFTP server.
Step 4: View Trace
After the file has been downloaded, you can view it in whatever protocol analyzer you prefer as long as it support the  tcpdump/libpcap format (.cap usual file extension).  A good tool is WireShark.
As previously noted you can look at packets live on the CUCM console screen. You can also specify port, source address, destination address, and protocol filters.  Enter the command “utils network capture ?” for a list of all options.

Tuesday, November 10, 2015

Resizing Disk Space on Ubuntu Server VMs Running on VMware ESXi 5

When you build a VM on VMware, you start with say 40GBs of hard disk space. You install the OS, setup the server, setup the database and you're online. But what happens when there's no more room on the server for your database and you need to add more disk space?
The process is quite simple:
a) Connect to the VMware ESXi 5 server using the vShpere Client. Edit the VM's properties to increase the hard disk size (VM has to be off). Now restart the VM. 
b) Login via SSH to the VM and follow this process.
- First list all partitions:
$ ls -al /dev/sda*
- Create new partition using fdisk:
$ fdisk /dev/sda
Then:
    type p - to list all your partitions
    type n - to create a new partition
    type l - for "logical"
    then give it a number (e.g. if you got 2 partitions listed as /dev/sda1 & /dev/sda2, for the new partition simply type "3" to create /dev/sda3)
    type t - to change the partition type to "Linux LVM"
    provide the partition number you previously created
    type 8e - for the "Linux LVM" type
    type p - to list the new partition table
    type w - to write changes and exit
- Reboot server:
$ reboot
- Assuming you created partition /dev/sda3, let's now create the physical volume in that partition:
$ pvcreate /dev/sda3 
- Now let's extend the server's Volume Group to that physical volume.
$ vgdisplay
This will give you the info on your current Volume Group. Note down the entry next to "VG Name". That's your Volume Group name.
$ vgextend EnterVolumeGroupNameHere /dev/sda3

Keep in mind

If you get a message saying /dev/sda3 could not be added to your Volume Group, you need to remove the physical volume and recreate it. Metadata might have gotten corrupt and thus the volume cannot be added to your Volume Group. So just do:
$ pvremove /dev/sda3
And then again:
$ pvcreate /dev/sda3 
- Since we're (essentially) extending the main logical volume, let's get the name of that:
$ lvdisplay
and note down the entry next to "LV Name". This is your logical volume's name (e.g. /dev/srv/root), which you'll now extend to the newly added partition/physical volume.

- Extend the logical volume by X GBs:
$ lvextend -L +XG yourLogicalVolumeNameMake sure you replace X above with the actual number of GBs you've added in your VM's settings. So if you increased your VM by 20GBs, the command becomes:
$ lvextend -L +20G yourLogicalVolumeName 
- Finally, let's resize the file system to the new allocated space:
$ resize2fs yourLogicalVolumeName
(this may take some time depending on number of GBs added to the file system.

- Check the new file system sizes:
$ df -hTYou should now see an increased disk space for your primary logical volume.

- Reboot and you're set :)

Tuesday, September 1, 2015

VMWare 5.5 Remove Locked File To Consolidate Snapshot


1. SSH into the ESXi host using Putty or another SSH client.
2. Drilled down to the Datastore where the VM in question resides: 
cd /vmfs/volumes/[datastore]/[VMDIR]
3. Locate the vmware.log and open with a text viewer.  I used cat vmware.log to view the log.  Also, used putty and set my session to log all screen output but you could extract that log to better review.
4. After you open the log, see if you see any errors regarding locked files, etc. (These errors existed in my case)
5. If errors exist such as "Failed to lock the file" there is possibly a active process that didn't stop correctly from a previous failed task.
6. Run # lsof | grep [name_of_locked_file]
7. Next run Kill [PID] .  The should be at the beginning from the output in step 6 which has been underlined in the following example. 
(EX:  5303   vpxa-worker  12   29  /vmfs/volumes/4360c6cf-2fee4d90-2404-5ef3fc344abb/VM/[name_of_locked_file])
8. The host may appear to have went offline for a few seconds if you happen to have a vSphere client open but should reconnect.
9. Next, select the VM which should still have a message of "Virtual machine disks consolidation is needed" and select Snapshot-->Consolidate.
10. The file should begin to consolidate and the error message should disappear.

Monday, January 19, 2015

How To Install Mediaproxy 2.5.2 on CentOS 6 64-bit


Mediaproxy 2.5.2 is a Python application from AG-Projects which is available as a free download as well as being available as a commercial product from AG-Projects. It is used in combination with the Mediaproxy module of OpenSIPS.
Mediaproxy 2 has several dependencies and can be quite tricky to install. The INSTALL instructions that come with the package are very helpful, but unfortunately they are aimed primarily at installers who either have Debian or Ubuntu Linux distributions. If you are using CentOS (or Red Hat), especially the 64 bit versions, then getting Mediaproxy 2 to run can be difficult.
This article is essentially an update for an earlier article showing how to install Mediaproxy 2.3.8 on CentOS 5. The installation process on CentOS 6 is quite different so I decided to leave the older article as it was and write up the process for CentOS 6 as a new article. Readers can choose whichever description is most suitable.
Here are the sequential instructions for installing the libraries needed for Mediaproxy 2.5.2. A number of packages that previously had to be installed using source tarballs are now installed using yum, making the process much easier.

Install some development and system packages using YUM

Development tools
yum groupinstall “Development Tools”
Development libraries and headers for some existing packages
yum install iptables-devel.x86_64
yum install libgpg-error-devel.x86_64
yum install python-devel.x86_64
libnfnetlink and libnetfilter_conntrack
yum install libnfnetlink-devel.x86_64
yum install libnetfilter_conntrack-devel.x86_64
libgcrypt
yum install libgcrypt-devel

Install some Python packages

python-zopeinterface
yum install python-zope-interface
python-cjson
This module can be installed using YUM, but first you will have to activate the rpmforge repository. The best way to do this is using the instructions here:
http://wiki.centos.org/AdditionalResources/Repositories/RPMForge
It is also possible to install the required python-cjson package from a source tarball as follows:
Suggested version: python-cjson-1.0.5.tar.gz
Locating a download site: http://pypi.python.org/pypi/python-cjson/1.0.5
Unzip: tar -xf python-cjson-1.0.5.tar.gz
Change to the sub-directory created when you unzipped the tarball
Build and install the library:
./setup.py build
./setup.py install
The following python packages are best installed from source tarballs. Details are given below:
python-application
Suggested version: python-application-1.3.0.tar.gz
Locating a download site: http://pypi.python.org/pypi/python-application/1.3.0
Unzip: tar -xf python-application-1.3.0.tar.gz
Change to the sub-directory created when you unzipped the tarball
Build and install the library:
./setup.py build
./setup.py install
python-gnutls
Suggested version: python-gnutls-1.2.4.tar.gz
Locating a download site: http://pypi.python.org/pypi/python-gnutls/1.2.4
Unzip: tar -xf python-gnutls-1.2.4.tar.gz
Change to the sub-directory created when you unzipped the tarball
Build and install the library:
./setup.py build
./setup.py install
Install the Twisted Python package
Install the core Twisted package from Twisted Matrix Labs
Suggested version: Twisted-11.0.0.tar.bz2
Locating a download site: http://twistedmatrix.com/trac
Unzip: tar -xf Twisted-11.0.0.tar.bz2
Change to the sub-directory created when you unzipped the tarball
Build and install the library:
./setup.py build
./setup.py install
Note also that version 8.2.0 of Twisted is available in the default YUM repositories. It is easier to install using YUM, but I prefer to install the later version from the source tarball because it matches the version used in the Debian install packages provided by AG-Projects.

A Checklist of dependencies and versions

On the system used for testing, the list of packages (and their versions) required by Mediaproxy looked like this prior to installation of the mediaproxy source code:
PackageVersion
gcc4.4.6
gcc-c++4.4.6
iptables and iptables-devel1.4.7
libnfnetlink and libnfnetlink-devel1.0.0
libnetfilter_conntrack and libnetfilter_conntrack-devel0.0.100
libgcrypt and libgcrypt-devel1.4.5
libgpg-error and libgpg-error-devel1.7
gnutls2.8.5
python and python-devel2.6.6
python-zope-interface3.5.2
python-twisted11.0.0
python-gnutls1.2.4
python-application1.3.0
python-cjson1.0.5

Install Mediaproxy 2

Mediaproxy 2
Suggested version: mediaproxy-2.5.2.tar.gz
Locating a download site: http://download.ag-projects.com/MediaProxy
Suggested location to copy and then unzip the tarball: /usr/local
Unzip: tar -xf mediaproxy-2.5.2.tar.gz
Change to the sub-directory created when you unzipped the tarball
I recommend that you build and install the library (standalone in the local directory) using the following command:
./build_inplace
In theory, an alternative approach to building and installing the library (system wide) is possible using the following commands, but I have had problems with this method:
./setup.py build
./setup.py install

And finally…

Make sure iptables is installed, running and correctly configured
Mediaproxy depends on iptables to relay IP packets through your server. Check that it is installed and running.
You can see the list of rules using the following command:
iptables -L -v -n
Note: In CentOS 6 you may well see a default rule preventing packet forwarding. If you leave this rule in place, the mediaproxy relay will start but it will not be able to relay any RTP. You will therefore need to change or delete this rule.
Set ip forwarding with immediate effect
echo “1″ > /proc/sys/net/ipv4/ip_forward
Set ip forwarding permanently
Edit the file /etc/sysctl.conf
Change the line “net.ipv4.ip_forward = 0″
to “net.ipv4.ip_forward = 1″
config.ini
Create a config.ini file from the sample template provided (config.ini.sample) and copy it to the appropriate directory. Using the system wide version, put it in /etc/mediaproxy. For standalone build you can also put it in the local directory where media-relay and media-dispatcher are located. If a file exists in both locations, the local one takes precedence

Using and troubleshooting Mediaproxy

When the media-relay program starts, it connects to one or more dispatchers using a TCP socket connection. Such connections can traverse a network which means dispatcher and relay do not have to run on the same server. It also means you should ideally start media-dispatcher before you start media-relay.
As far as I know, media-dispatcher has to run on the same server as OpenSIPS and they always have a 1-to-1 relationship. They connect via a local Unix socket. However, a relay can connect to more than one dispatcher and each dispatcher can accept connections from more than one relay, if needed. This many-to-many relationship is useful in terms of resilience and scalability.
By default, media-dispatcher listens for TCP connections from relays on port 25060. You can use the Linux command “netstat -ltnp” to see a list of listening ports and the name of the associated program. The command “netstat -tnp” will show established connections. Both media-dispatcher and media-relay appear as program name “python”, so the output of “netstat -tnp” on a server with both installed and connected might appear like this:
[root@sip1 log]# netstat -tnp
Active Internet connections (w/o servers)
Proto Recv-Q Send-Q Local Address       Foreign Address     State       PID/Program name
tcp        0      0 12.34.56.78:25060   12.34.56.78:43708   ESTABLISHED 1584/python
tcp        0      0 12.34.56.78:43708   12.34.56.78:25060   ESTABLISHED 1676/python
You can check if dispatcher and relay are running with the following Linux command:
ps ax | grep media-
The output should look somewhat like this:
[root@sip1 mediaproxy]# ps ax | grep media-
 1584 ?       SL   0:00 python /usr/local/mediaproxy/media-dispatcher
 1676 ?       SL   0:00 python /usr/local/mediaproxy/media-relay
15059 pts/0   S+   0:00 grep media-
If you are having problem running the programs, you should check for errors in the file /var/log/messages. Here is what the log looks like when dispatcher starts:
Sep 26 12:39 sip1 media-dispatcher[1584]: Starting MediaProxy Dispatcher 2.5.2
Sep 26 12:39 sip1 media-dispatcher[1584]: using set_wakeup_fd
Sep 26 12:39 sip1 media-dispatcher[1584]: Twisted is using epollreactor
Sep 26 12:39 sip1 media-dispatcher[1584]: mediaproxy.dispatcher.RelayFactory starting on 25060
Sep 26 12:39 sip1 media-dispatcher[1584]: mediaproxy.dispatcher.OpenSIPSControlFactory starting on "'/var/run/mediaproxy.sock'"
Sep 26 12:39 sip1 media-dispatcher[1584]: mediaproxy.dispatcher.ManagementControlFactory starting on 25061
…and here are the log messages for relay when it starts ok:
Sep 26 12:43 sip1 media-relay[1676]: Starting MediaProxy Relay 2.5.2
Sep 26 12:43 sip1 media-relay[1676]: using set_wakeup_fd
Sep 26 12:43 sip1 media-relay[1676]: Set resource limit for maximum open file descriptors to 11000

Saturday, January 17, 2015

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.