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asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb [2012/08/05 13:04] (current)
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 +====== Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database ======
  
 +<code c>
 +Author:
 +   ​Daniel-Constantin Mierla
 +</​code>​
 +
 +This tutorial shows how to use Asterisk database to load the SIP user profile from within Kamailio configuration file. A typical use case is Kamailio as a SIP proxy router to scale Asterisk, by handling the user authentication and registration,​ letting one or a farm of Asterisks to deal with call handling (e.g., IVR, transconding,​ gatewaying, prepaid billing, a.s.o.).
 +
 +The authentication module in Kamailio can be configured to connect to any database and fetch the password from custom table and column, therefore creation of a database view is not really required, unless you want for other purposes.
 +
 +The document here presents the installation from sources, uses MySQL as database server and unixodbc for Asterisk realtime. The steps are given for Ubuntu/​Debian operating systems.
 +
 +Previous release of this tutorial is using Kamailio 3.0.x series and it is available at:
 +  * [[asterisk:​realtime:​kamailio-3.0.x-asterisk-1.6.2-astdb|Kamailio 3.0.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database]]
 +
 +<note important>​
 +**//The improvements added to Kamailio configuration comparing with previous version://​**
 +  * IP authentication can be enabled with define WITH_IPAUTH
 +  * TLS support can be enabled with define WITH_TLS
 +    * TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP
 +  * detection of DoS attacks, it can be enabled with define WITH_ANTIFLOOD
 +  * banning traffic from attacker IP addresses for a specific time interval
 +  * restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing
 +</​note>​
 +
 +<note tip>
 +If you look for the other kind of integration approach (use of Kamailio database and create views to be accessed by Asterisk), follow next link:
 +  * [[http://​www.voip-info.org/​wiki/​view/​Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x|Asterisk 1.4.x and Kamailio 1.5.x Realtime Integration]]
 +</​note>​
 +
 +===== Architecture =====
 +
 +  * reuse as much as possible the default Asterisk relatime configuration
 +  * handle authentication in Kamailio
 +  * handle user location in Kamailio
 +  * routing of calls between local phones is managed by Asterisk
 +  * media services are handled by Asterisk according to Asterisk dialplan
 +  * routing of other SIP messages not related to calls are handled by Kamailio directly
 +
 +==== Registration ====
 +
 +Kamailio does authentication for registration. If successful, it notifies Asterisk with a new REGISTER that the phone is available at its IP.
 +
 +<ditaa round noedgesep center>
 +
 +    +----------+ ​  ​REGISTER ​      ​+----------+ ​                 +----------+
 +    |          |----------------->​+ ​         |                  |          |
 +    |          |   ​contact:​ IP1   ​| ​         |                  |          |
 +    | Phone A  |                  | Kamailio |                  | Asterisk |
 +    |          |   401 Auth       ​| ​         |                  |          |
 +    |   ​IP1 ​   +<​-----------------+ ​  ​IP2 ​   |                  |   ​IP3 ​   |
 +    |          |                  |          |                  |          |
 +    |          |                  |          |                  |          |
 +    |          |   ​REGISTER ​      ​| ​         |                  |          |
 +    |          +----------------->​+ ​         |                  |          |
 +    |          |   ​contact:​ IP1   ​| ​         |   ​REGISTER ​      ​| ​         |
 +    |          |                  |          +----------------->​+ ​         |
 +    |          |                  | c6D6     ​| ​  ​contact:​ IP2   | cF90     |
 +    |          |   ​200ok ​         |          |                  |          |
 +    |          +<​-----------------+ ​         |   ​200ok ​         |          |
 +    |          |   ​contact:​ IP1   ​| ​         +<​-----------------+ ​         |
 +    |          |                  |          |   ​contact:​ IP2   ​| ​         |
 +    +----------+ ​                 +----------+ ​                 +----------+
 +</​ditaa>​
 +
 +==== Call Initiation ====
 +
 +Call authentication is handled by Kamailio. When a new calls arrives and it is authenticated,​ Kamailio forwards it to Asterisk. If destination number is online, Asterisk will send the call back to Kamailio since the contact of destination is Kamailio IP. Then Kamailio will do location lookup and send to destination phone IP.
 +
 +<ditaa round noedgesep center>
 +
 +    +----------+ ​  ​INVITE ​        ​+----------+ ​                 +----------+
 +    |          |----------------->​+ ​         |                  |          |
 +    |          |   To: B@IP2      |          |                  |          |
 +    |  Phone A |                  | Kamailio |                  | Asterisk |
 +    |          |   401 Auth       ​| ​         |                  |          |
 +    |   ​IP1 ​   +<​-----------------+ ​ IP2     ​| ​                 |  IP3     |
 +    |          |                  |          |                  |          |
 +    |          |                  | c6D6     ​| ​                 | cF90     |
 +    |          |   ​INVITE ​        ​| ​         |                  |          |
 +    |          |----------------->​+ ​         |                  |          |
 +    |          |   To: B@IP2      |          |   ​INVITE ​        ​| ​         |
 +    |          |                  |          +----------------->​+ ​         |
 +    |          |                  |          |   To: B@IP3      |          |
 +    |          |                  |          |                  |          |
 +    +----------+ ​                 |          |   ​INVITE ​        ​| ​         |
 +                                  |          +<​-----------------+ ​         |
 +    +----------+ ​  ​INVITE ​        ​| ​         |   To: B@IP2      |          |
 +    |          +<​-----------------+ ​         |                  |          |
 +    |          |   To: B@IP4      |          |                  |          |
 +    |  Phone B |                  |          |                  |          |
 +    |          |                  |          |                  |          |
 +    |          |   ​200ok ​         |          |                  |          |
 +    |   ​IP4 ​   +------> ​ ...      |          |                  |          |
 +    +----------+ ​                 +----------+ ​                 +----------+
 +
 +</​ditaa>​
 +
 +===== Requirements =====
 +
 +  * Linux OS (this tutorial is applied on Ubuntu 10.04)
 +  * Kamailio 3.1.x - http://​www.kamailio.org
 +  * Asterisk 1.6.2.x - http://​www.asterisk.org
 +  * UnixODBC - http://​www.unixodbc.org
 +  * MySQL Server and Client (recommended 5.1+, min 5.0) - http://​www.mysql.com
 +  * MySQL client library
 +  * ODBC MySQL connector
 +
 +Since many commands require root privileges, I assume you either know to use **sudo** to run the command or do **su** to root and run all commands as root:
 +
 +<​code>​
 +sudo su -
 +</​code>​
 +
 +===== MySQL Installation =====
 +
 +MySQL server and client are included in all major Linux distributions. So is in Ubuntu which has version 5.1.x. To install the server and client, open a terminal and do:
 +
 +<​code>​
 +apt-get install mysql-server
 +</​code>​
 +
 +For a more detailed tutorial about MySQL installation on Ubuntu 10.04, see:
 +  * http://​library.linode.com/​databases/​mysql/​ubuntu-10.04-lucid
 +
 +To install MySQL client library do:
 +
 +<​code>​
 +apt-get install libmysqlclient-dev
 +</​code>​
 +
 +
 +===== Install UnixODBC =====
 +
 +To install the UnixODBC devel libraries, run:
 +
 +<​code>​
 +apt-get install unixodbc-dev
 +</​code>​
 +
 +If your operating system does not provide a package for it, download the sources from http://​www.unixodbc.org/,​ compile and install. Then tune the Asterisk compilation system if the unixodbc is not detected automatically.
 +
 +To install the ODBC MySQL connector, run:
 +
 +<​code>​
 +apt-get install libmyodbc
 +</​code>​
 +
 +===== Asterisk Installation =====
 +
 +Get Asterisk sources from http://​www.asterisk.org. At this moment Asterisk 1.6.2.13 is the latest stable version.
 +
 +<​code>​
 +cd /​usr/​local/​src
 +wget http://​downloads.asterisk.org/​pub/​telephony/​asterisk/​releases/​asterisk-1.6.2.13.tar.gz
 +tar xvfz asterisk-1.6.2.13.tar.gz
 +cd asterisk-1.6.2.13
 +./​configure ​
 +</​code>​
 +
 +To enable ODBC storage for voicemail, run:
 +
 +<​code>​
 +make menuselect
 +</​code>​
 +
 +Then select option **Voicemail Build Options**, enable option **ODBC_STORAGE**. Save and exit.
 +
 +Then compile and install:
 +
 +<​code>​
 +make
 +make install
 +</​code>​
 +
 +===== Kamailio Installation =====
 +
 +A step by step tutorial to install latest Kamailio 3.1.x from git is available at:
 +  * http://​www.kamailio.org/​dokuwiki/​doku.php/​install:​kamailio-3.1.x-from-git
 +
 +If you want to install from sources tarball:
 +
 +<​code>​
 +cd /​usr/​local/​src
 +wget http://​www.kamailio.org/​pub/​kamailio/​3.0.1/​src/​kamailio-3.1.0_src.tar.gz
 +tar xvfz kamailio-3.1.0_src.tar.gz
 +cd kamailio-3.1.0
 +make include_modules="​db_mysql"​ cfg
 +make all
 +make install ​
 +</​code>​
 +
 +===== Kamailio Database =====
 +
 +This database is required to store location records (phone contact addresses).
 +
 +Use kamdbctl to create the database:
 +
 +<​code>​
 +/​usr/​local/​sbin/​kamdbctl create
 +</​code>​
 +
 +No other changes to Kamailio database structure are required. The SIP server will fetch the password from Asterisk database.
 +
 +===== Asterisk Database =====
 +
 +Execute next SQL script with **mysql** client:
 +
 +<code sql>
 +create database asterisk;
 +
 +use asterisk;
 +
 +GRANT ALL ON asterisk.* to asterisk@localhost IDENTIFIED BY '​asterisk_password';​
 +
 +CREATE TABLE `sipusers` (
 + `id` int(11) NOT NULL AUTO_INCREMENT,​
 + ​`name` varchar(80) NOT NULL DEFAULT '',​
 + ​`host` varchar(31) NOT NULL DEFAULT '',​
 + `nat` varchar(5) NOT NULL DEFAULT '​no',​
 + ​`type` enum('​user','​peer','​friend'​) NOT NULL DEFAULT '​friend',​
 + ​`accountcode` varchar(20) DEFAULT NULL,
 + ​`amaflags` varchar(13) DEFAULT NULL,
 + ​`call-limit` smallint(5) UNSIGNED DEFAULT NULL,
 + ​`callgroup` varchar(10) DEFAULT NULL,
 + ​`callerid` varchar(80) DEFAULT NULL,
 + ​`cancallforward` char(3) DEFAULT '​yes',​
 + ​`canreinvite` char(3) DEFAULT '​yes',​
 + ​`context` varchar(80) DEFAULT NULL,
 + ​`defaultip` varchar(15) DEFAULT NULL,
 + ​`dtmfmode` varchar(7) DEFAULT NULL,
 + ​`fromuser` varchar(80) DEFAULT NULL,
 + ​`fromdomain` varchar(80) DEFAULT NULL,
 + ​`insecure` varchar(4) DEFAULT NULL,
 + ​`language` char(2) DEFAULT NULL,
 + ​`mailbox` varchar(50) DEFAULT NULL,
 + ​`md5secret` varchar(80) DEFAULT NULL,
 + ​`deny` varchar(95) DEFAULT NULL,
 + ​`permit` varchar(95) DEFAULT NULL,
 + ​`mask` varchar(95) DEFAULT NULL,
 + ​`musiconhold` varchar(100) DEFAULT NULL,
 + ​`pickupgroup` varchar(10) DEFAULT NULL,
 + ​`qualify` char(3) DEFAULT NULL,
 + ​`regexten` varchar(80) DEFAULT NULL,
 + ​`restrictcid` char(3) DEFAULT NULL,
 + ​`rtptimeout` char(3) DEFAULT NULL,
 + ​`rtpholdtimeout` char(3) DEFAULT NULL,
 + ​`secret` varchar(80) DEFAULT NULL,
 + ​`setvar` varchar(100) DEFAULT NULL,
 + ​`disallow` varchar(100) DEFAULT NULL,
 + ​`allow` varchar(100) DEFAULT NULL,
 + ​`fullcontact` varchar(80) NOT NULL DEFAULT '',​
 + ​`ipaddr` varchar(15) NOT NULL DEFAULT '',​
 + ​`port` mediumint(5) UNSIGNED NOT NULL DEFAULT '​0',​
 + ​`regserver` varchar(100) DEFAULT NULL,
 + ​`regseconds` int(11) NOT NULL DEFAULT '​0',​
 + ​`lastms` int(11) NOT NULL DEFAULT '​0',​
 + ​`username` varchar(80) NOT NULL DEFAULT '',​
 + ​`defaultuser` varchar(80) NOT NULL DEFAULT '',​
 + ​`subscribecontext` varchar(80) DEFAULT NULL,
 + ​`useragent` varchar(20) DEFAULT NULL,
 + ​`sippasswd` varchar(80) DEFAULT NULL,
 + ​PRIMARY KEY  (`id`),
 + ​UNIQUE KEY `name_uk` (`name`)
 +);
 + 
 +CREATE TABLE `sipregs` (
 + `id` int(11) NOT NULL AUTO_INCREMENT,​
 + ​`name` varchar(80) NOT NULL DEFAULT '',​
 + ​`fullcontact` varchar(80) NOT NULL DEFAULT '',​
 + ​`ipaddr` varchar(15) NOT NULL DEFAULT '',​
 + ​`port` mediumint(5) UNSIGNED NOT NULL DEFAULT '​0',​
 + ​`username` varchar(80) NOT NULL DEFAULT '',​
 + ​`regserver` varchar(100) DEFAULT NULL,
 + ​`regseconds` int(11) NOT NULL DEFAULT '​0',​
 + ​PRIMARY KEY  (`id`),
 + ​UNIQUE KEY `name` (`name`)
 +);
 +
 +CREATE TABLE IF NOT EXISTS `voiceboxes` (
 + ​`uniqueid` int(4) NOT NULL AUTO_INCREMENT,​
 + ​`customer_id` varchar(10) DEFAULT NULL,
 + ​`context` varchar(10) NOT NULL,
 + ​`mailbox` varchar(10) NOT NULL,
 + ​`password` varchar(12) NOT NULL,
 + ​`fullname` varchar(150) DEFAULT NULL,
 + ​`email` varchar(50) DEFAULT NULL,
 + ​`pager` varchar(50) DEFAULT NULL,
 + `tz` varchar(10) DEFAULT '​central',​
 + ​`attach` enum('​yes','​no'​) NOT NULL DEFAULT '​yes',​
 + ​`saycid` enum('​yes','​no'​) NOT NULL DEFAULT '​yes',​
 + ​`dialout` varchar(10) DEFAULT NULL,
 + ​`callback` varchar(10) DEFAULT NULL,
 + ​`review` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`operator` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`envelope` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`sayduration` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`saydurationm` tinyint(4) NOT NULL DEFAULT '​1',​
 + ​`sendvoicemail` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`delete` enum('​yes','​no'​) NULL DEFAULT '​no',​
 + ​`nextaftercmd` enum('​yes','​no'​) NOT NULL DEFAULT '​yes',​
 + ​`forcename` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`forcegreetings` enum('​yes','​no'​) NOT NULL DEFAULT '​no',​
 + ​`hidefromdir` enum('​yes','​no'​) NOT NULL DEFAULT '​yes',​
 + ​`stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE CURRENT_TIMESTAMP,​
 + ​PRIMARY KEY  (`uniqueid`),​
 + KEY `mailbox_context` (`mailbox`,​`context`)
 +); 
 + 
 +CREATE TABLE `voicemessages` (
 + `id` int(11) NOT NULL AUTO_INCREMENT,​
 + ​`msgnum` int(11) NOT NULL DEFAULT '​0',​
 + `dir` varchar(80) DEFAULT '',​
 + ​`context` varchar(80) DEFAULT '',​
 + ​`macrocontext` varchar(80) DEFAULT '',​
 + ​`callerid` varchar(40) DEFAULT '',​
 + ​`origtime` varchar(40) DEFAULT '',​
 + ​`duration` varchar(20) DEFAULT '',​
 + ​`mailboxuser` varchar(80) DEFAULT '',​
 + ​`mailboxcontext` varchar(80) DEFAULT '',​
 + ​`recording` longblob,
 + ​`flag` varchar(128) DEFAULT '',​
 + ​PRIMARY KEY  (`id`),
 + KEY `dir` (`dir`)
 +);
 +
 +
 +CREATE TABLE version (
 +    table_name VARCHAR(32) NOT NULL,
 +    table_version INT UNSIGNED DEFAULT 0 NOT NULL
 +);
 +INSERT INTO version (table_name,​ table_version) values ('​sipusers','​6'​);​
 +
 +</​code>​
 +
 +If you save it to **asterisk.sql**,​ then you can load it to MySQL server with:
 +
 +<​code>​
 +mysql -u root -p <​asterisk.sql
 +</​code>​
 +
 +Before executing the SQL script, be sure you change the password for MySQL asterisk user, in line:
 +
 +<​code>​
 +GRANT ALL ON asterisk.* to asterisk@localhost IDENTIFIED BY '​asterisk_password';​
 +</​code>​
 +
 +<note important>​
 +**sipusers** is the standard table required by Asterisk to store SIP user profile, with one extra column **sippasswd** where will be stored the password for SIP authentication. By default, Asterisk uses the column **secret** for SIP user password, but if that is filled in, Asterisk will ask for authentication again, resulting in double-authentication which we want to avoid.
 +</​note>​
 +<note important>​
 +**sipregs** is used to store SIP registrations. Registrations can be stored in **sipusers** tables as well, in case you do not want a separate table. Just omit the appropriate entry in **/​etc/​asterisk/​extconfig.conf**.
 +</​note>​
 +<note important>​
 +**voiceboxes** is used to store voicemail box profiles and has the standard structure required by Asterisk. Storing voice box profiles in database allows to run several instances of Asterisk that can be load balanced or used in fail-over mode to store or listen to voice messages.
 +</​note>​
 +<note important>​
 +**voicemessages** is used to store voice messages and has the standard structure required by Asterisk. Storing voice messages in database allows to run several instances of Asterisk that can be load balanced or used in fail-over mode to store or listen to voice messages.
 +</​note>​
 +<note important>​
 +**version** is a table used by Kamailio to check database structure version. You can just forget about it once you create it.
 +</​note>​
 +
 +===== UnixODBC Configuration =====
 +
 +Edit **/​etc/​odbcinst.ini** and add:
 +
 +<​code>​
 +[MySQL]
 +Description = MySQL driver
 +Driver = /​usr/​lib/​odbc/​libmyodbc.so
 +Setup = /​usr/​lib/​odbc/​libodbcmyS.so
 +CPTimeout =
 +CPReuse =
 +UsageCount = 1
 +</​code>​
 +
 +Edit **/​etc/​odbc.ini** and add:
 +
 +<​code>​
 +[MySQL-asterisk]
 +Description = MySQL Asterisk database
 +Trace = Off
 +TraceFile = stderr
 +Driver = MySQL
 +SERVER = localhost
 +USER = asterisk
 +PASSWORD = asterisk_password
 +PORT = 3306
 +DATABASE = asterisk ​
 +</​code>​
 +
 +===== Asterisk UnixODBC Configuration =====
 +
 +Edit **/​etc/​asterisk/​res_odbc.conf** and set:
 +
 +<​code>​
 +[asterisk]
 +enabled => yes
 +dsn => MySQL-asterisk
 +username => asterisk
 +password => asterisk_password
 +pre-connect => yes
 +</​code>​
 +
 +Edit **/​etc/​asterisk/​extconfig.conf** and set:
 +
 +<​code>​
 +sipusers => odbc,​asterisk,​sipusers
 +sippeers => odbc,​asterisk,​sipusers
 +sipregs => odbc,​asterisk,​sipregs
 +voicemail => odbc,​asterisk,​voiceboxes
 +</​code>​
 +
 +===== Asterisk Configuration =====
 +
 +In case you need to cache the realtime users, then edit **/​etc/​asterisk/​sip.conf** and set:
 +
 +<​code>​
 +rtcachefriends=yes ​
 +</​code>​
 +
 +==== Dialplan Configuration ====
 +
 +It is up to you what dialplan you build in **/​etc/​asterisk/​extensions.conf**. Practically is nothing special for this configuration,​ as phones will appear in Asterisk with contact address pointing to Kamailio IP and port.
 +
 +==== Sample data ====
 +
 +For testing purposes, here is a sample that can be plugged in **/​etc/​asterisk/​extensions.conf**:​
 +
 +<​code>​
 +; our phones use 3 digit extensions, starting with 1
 +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
 +</​code>​
 +
 +It does the classic behaviour:
 +  * if phone is registered, route the call to it
 +  * if phone is unavailable,​ enter voicemail service
 +  * if phone is busy, enter voicemail service
 +
 +In the Asterisk database, you can insert following records to create SIP users 101, 102 and 103:
 +
 +<code sql>
 +insert into sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
 +  values ('​101',​ '​101',​ '​dynamic',​ '​101',​ '​101',​ '​yoursip.com',​ '​101'​);​
 +insert into sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
 +  values ('​102',​ '​102',​ '​dynamic',​ '​102',​ '​102',​ '​yoursip.com',​ '​102'​);​
 +insert into sipusers (name, username, host, sippasswd, fromuser, fromdomain, mailbox)
 +  values ('​103',​ '​103',​ '​dynamic',​ '​103',​ '​103',​ '​yoursip.com',​ '​103'​);​
 +
 +insert into sipregs(name) values('​101'​);​
 +insert into sipregs(name) values('​102'​);​
 +insert into sipregs(name) values('​103'​);​
 +
 +insert into voiceboxes(customer_id,​ context, mailbox, password) values ('​101',​ '​default',​ '​101',​ '​1234'​);​
 +insert into voiceboxes(customer_id,​ context, mailbox, password) values ('​101',​ '​default',​ '​102',​ '​1234'​);​
 +insert into voiceboxes(customer_id,​ context, mailbox, password) values ('​101',​ '​default',​ '​103',​ '​1234'​);​
 +</​code>​
 +
 +<​note>​In case you use **sipregs** you have to create a record for each extension where to set the '​name'​ to value of '​name'​ from **sipusers**. The rest is populated by Asterisk from registrations.</​note>​
 +
 +<note important>​Be sure you configure Asterisk **to not authenticate** SIP requests coming from Kamailio.</​note>​
 +
 +===== Kamailio Configuration =====
 +
 +This configuration file is an update of default Kamailio 3.1.x configuration file. It is easy to spot the changes with **diff** or following **#!define WITH_ASTERISK** (i.e., the parts within **#!ifdef WITH_ASTERISK ... #!endif**.
 +
 +Practically,​ if you want to disable the routing through Asterisk, remove the line:
 +
 +<​code>​
 +#!define WITH_ASTERISK
 +</​code>​
 +
 +<note important>​The **define** directives are supported only starting with version 3.0.0. Also, registering to Asterisk in behalf of phones setting the contact address to Kamailio IP and port is a feature introduced in Kamailio 1.5.x, don't try this config with other forks of SER, working variants are Kamailio 3.0.x+ or SER v3.0.x+.</​note>​
 +
 +==== Config File ====
 +
 +Entire config file is pasted below. Do not forget to change the listen IP, port for Kamailio and Asterisk. In this example, Kamailio listens on IP 192.168.178.23 port 5060 and Asterisk listens on IP 192.168.178.23 port 5080.
 +
 +Also, if you created Asterisk or Kamailio databases with different names than specified abover, or you changed the usernames and passwords to connect to MySQL server, do not forget to update **DBURL** and **DBASTURL** defines.
 +
 +<code c>
 +#!KAMAILIO
 +
 +#!define WITH_MYSQL
 +#!define WITH_AUTH
 +#!define WITH_USRLOCDB
 +#!define WITH_ASTERISK
 +
 +#
 +# Kamailio (OpenSER) SIP Server v3.1 - default configuration script
 +#     - web: http://​www.kamailio.org
 +#     - git: http://​sip-router.org
 +#
 +# Direct your questions about this file to: <​sr-users@lists.sip-router.org>​
 +#
 +# 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 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 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 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 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://​openser:​openserrw@localhost/​openser"​
 +#!define DBASTURL "​mysql://​asterisk:​asterisk_password@localhost/​asterisk"​
 +#!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
 +listen=192.168.178.23
 +
 +/* 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
 +
 +####### 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_ASTERISK
 +asterisk.bindip = "​192.168.178.23"​ desc "​Asterisk IP Address"​
 +asterisk.bindport = "​5080"​ desc "​Asterisk Port"
 +kamailio.bindip = "​192.168.178.23"​ desc "​Kamailio IP Address"​
 +kamailio.bindport = "​5060"​ desc "​Kamailio Port"
 +#!endif
 +
 +####### Modules Section ########
 +
 +# set paths to location of modules
 +#!ifdef LOCAL_TEST_RUN
 +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 "​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_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_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)
 +
 +
 +# ----- 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=$fd;​dst_ouser=$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;​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",​ "​use_domain",​ MULTIDOMAIN)
 +modparam("​auth_db",​ "​calculate_ha1",​ yes)
 +modparam("​auth_db",​ "​load_credentials",​ ""​)
 +
 +#!ifdef WITH_ASTERISK
 +modparam("​auth_db",​ "​user_column",​ "​username"​)
 +modparam("​auth_db",​ "​password_column",​ "​sippasswd"​)
 +modparam("​auth_db",​ "​db_url",​ DBASTURL)
 +#!else
 +modparam("​auth_db",​ "​db_url",​ DBURL)
 +modparam("​auth_db",​ "​password_column",​ "​password"​)
 +#!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
 +
 +
 +# ----- domain params -----
 +#!ifdef WITH_MULTIDOMAIN
 +modparam("​domain",​ "​db_url",​ DBURL)
 +# use caching
 +modparam("​domain",​ "​db_mode",​ 1)
 +# 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
 +
 +####### Routing Logic ########
 +
 +
 +# Main SIP request routing logic
 +# - processing of any incoming SIP request starts with this route
 +route {
 +
 + # per request initial checks
 + route(REQINIT);​
 +
 + # NAT detection
 + route(NAT);​
 +
 + # 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] {
 +#!ifdef WITH_NAT
 + if (check_route_param("​nat=yes"​)) {
 + setbflag(FLB_NATB);​
 + }
 + if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) {
 + route(RTPPROXY);​
 + }
 +#!endif
 +
 + /* example how to enable some additional event routes */
 + if (is_method("​INVITE"​)) {
 + #​t_on_branch("​BRANCH_ONE"​);​
 + t_on_reply("​REPLY_ONE"​);​
 + t_on_failure("​FAIL_ONE"​);​
 + }
 +
 + 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)
 + # - traffic from Asterisk excluded
 + if((src_ip!=myself) && (!route(FROMASTERISK)))
 + {
 + 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
 + }
 + 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_ALIASDB
 + # search in DB-based aliases
 + alias_db_lookup("​dbaliases"​);​
 +#!endif
 +
 +#!ifdef WITH_ASTERISK
 + if(!is_method("​INVITE"​)) {
 + # non-INVITE request are routed directly by Kamailio
 +#!endif
 + if (!lookup("​location"​)) {
 + switch ($rc) {
 + case -1:
 + case -3:
 + t_newtran();​
 + t_reply("​404",​ "Not Found"​);​
 + exit;
 + case -2:
 + sl_send_reply("​405",​ "​Method Not Allowed"​);​
 + exit;
 + }
 + }
 +#!ifdef WITH_ASTERISK
 + } /* end non-INVITE test */
 + # only INVITE from now on
 + if(route(FROMASTERISK))
 + {
 + # coming from Asterisk - do location lookup
 + if (!lookup("​location"​)) {
 + switch ($rc) {
 + case -1:
 + case -3:
 + t_newtran();​
 + t_reply("​404",​ "Not Found"​);​
 + exit;
 + case -2:
 + sl_send_reply("​405",​ "​Method Not Allowed"​);​
 + exit;
 + }
 + }
 + } else {
 + # new call - send to Asterisk
 + route(TOASTERISK);​
 + }
 +#!endif
 +
 + # 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] {
 +#!ifdef WITH_AUTH
 +
 +#!ifdef WITH_ASTERISK
 + # do not auth traffic from Asterisk - trusted!
 + if(route(FROMASTERISK))
 + return;
 +#!endif
 +
 + if (is_method("​REGISTER"​))
 + {
 + # authenticate the REGISTER requests (uncomment to enable auth)
 +#!ifdef WITH_ASTERISK
 + if (!www_authorize("​$td",​ "​sipusers"​))
 +#!else
 + if (!www_authorize("​$td",​ "​subscriber"​))
 +#!endif
 + {
 + www_challenge("​$td",​ "​0"​);​
 + exit;
 + }
 +
 + if ($au!=$tU)
 + {
 + sl_send_reply("​403","​Forbidden auth ID");
 + exit;
 + }
 + } else {
 +
 +#!ifdef WITH_IPAUTH
 + if(allow_source_address())
 + {
 + # source IP allowed
 + return;
 + }
 +#!endif
 +
 + # authenticate if from local subscriber
 + if (from_uri==myself)
 + {
 +#!ifdef WITH_ASTERISK
 + if (!proxy_authorize("​$fd",​ "​sipusers"​)) {
 +#!else
 + if (!proxy_authorize("​$fd",​ "​subscriber"​)) {
 +#!endif
 + proxy_challenge("​$fd",​ "​0"​);​
 + exit;
 + }
 + if (is_method("​PUBLISH"​))
 + {
 + if ($au!=$tU) {
 + sl_send_reply("​403","​Forbidden auth ID");
 + exit;
 + }
 + } else {
 + if ($au!=$fU) {
 + sl_send_reply("​403","​Forbidden auth ID");
 + exit;
 + }
 + }
 +
 + consume_credentials();​
 + # caller authenticated
 + } else {
 + # caller is not local subscriber, then check if it calls
 + # a local destination,​ otherwise deny, not an open relay here
 + if (!uri==myself)
 + {
 + sl_send_reply("​403","​Not relaying"​);​
 + exit;
 + }
 + }
 + }
 +#!endif
 + return;
 +}
 +
 +# Caller NAT detection route
 +route[NAT] {
 +#!ifdef WITH_NAT
 + force_rport();​
 + if (nat_uac_test("​19"​)) {
 + if (method=="​REGISTER"​) {
 + fix_nated_register();​
 + } else {
 + fix_nated_contact();​
 + }
 + setflag(FLT_NATS);​
 + }
 +#!endif
 + return;
 +}
 +
 +# RTPProxy control
 +route[RTPPROXY] {
 +#!ifdef WITH_NAT
 + if (is_method("​BYE"​)) {
 + unforce_rtp_proxy();​
 + } else if (is_method("​INVITE"​)){
 + force_rtp_proxy();​
 + }
 + if (!has_totag()) add_rr_param(";​nat=yes"​);​
 +#!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
 +
 +
 +#!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:​ <​sip:"​ + $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
 +
 +
 +# Sample branch router
 +branch_route[BRANCH_ONE] {
 + xdbg("​new branch at $ru\n"​);​
 +}
 +
 +# Sample onreply route
 +onreply_route[REPLY_ONE] {
 + xdbg("​incoming reply\n"​);​
 +#!ifdef WITH_NAT
 + if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))
 + &&​ status=~"​(183)|(2[0-9][0-9])"​) {
 + force_rtp_proxy();​
 + }
 + if (isbflagset("​6"​)) {
 + fix_nated_contact();​
 + }
 +#!endif
 +}
 +
 +# Sample failure route
 +failure_route[FAIL_ONE] {
 +#!ifdef WITH_NAT
 + if (is_method("​INVITE"​)
 + &&​ (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) {
 + unforce_rtp_proxy();​
 + }
 +#!endif
 +
 + if (t_is_canceled()) {
 + exit;
 + }
 +
 + # uncomment the following lines if you want to block client ​
 + # redirect based on 3xx replies.
 + ##if (t_check_status("​3[0-9][0-9]"​)) {
 + ##​t_reply("​404","​Not found"​);​
 + ## exit;
 + ##}
 +
 + # uncomment the following lines if you want to redirect the failed ​
 + # calls to a different new destination
 + ##if (t_check_status("​486|408"​)) {
 + ##​ sethostport("​192.168.2.100:​5060"​);​
 + ##​ append_branch();​
 + ## # do not set the missed call flag again
 + ##​ t_relay();​
 + ##}
 +}
 +</​code>​
 +
 +==== Config Remarks ====
 +
 +  * REGISTER request sent to Asterisk is triggered by a REGISTER coming from phone, but is built from scratch and sent with **uac_req_send()**.
 +  * IP and pot for kamailio set via custom global parameters **kamailio.bindip** and **kamailio.bindport** are used to build the contact for REGISTER request sent to Asterisk
 +  * any INVITE received from phones (not coming from Asterisk) is authenticated and then sent to Asterisk
 +  * Asterisk will automatically send back to Kamailio the INVITEs for online SIP phones (as the contact in Asterisk **sipregs** points to Kamailio IP and port)
 +  * any INVITE received from Asterisk is handled via user location and then sent to destination phone
 +
 +===== Other Benefits =====
 +
 +With such architecture,​ several other benefits can be achieved quickly:
 +  * increase of security - Kamailio handling SIP signaling only, can absorb easier the flooding attacks, protecting Asterisk
 +  * transport layer gatewaying - Kamailio has mature implementations for UDP, TCP, TLS and SCTP, therefore you can use it in front of Asterisk to translate between these protocols
 +  * load balancing - you can use several instances of Asterisk, Kamailio can do load balancing among them
 +  * high availability - Kamailio can be configured to re-route the call if selected Asterisk box does not react in a given period of time, e.g., if one Asterisk is not responsive in 2 sec, sent the call to another Asterisk
 +
 +===== See also =====
 +
 +  * [[http://​www.kamailio.org/​dokuwiki/​doku.php/​install:​kamailio-3.0.x-from-git|Install Kamailio 3.0.x from GIT]]
 +  * [[http://​www.voip-info.org/​wiki/​view/​Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x|Realtime Integration of Asterisk 1.4.x with Kamailio 1.5.x]]
 +
 +{{tag>​kamailio asterisk}}
 +
 +<box 100% round red|Kamailio (OpenSER) Advanced Course: **Jan 24-26, 2011**, Irvine, CA, USA>
 +<​html><​a href="​http://​www.asipto.com/"><​span style="​color:#​0000ff;">​Learn all about Load Balancing, Least Cost Routing, High Availability and more!<br />~~~ Click Here for More Details and Registration ~~~</​span></​a></​html>​
 +</​box>​


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