Kamailio 3.1.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database

Author:
   Daniel-Constantin Mierla

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:

• Kamailio 3.0.x and Asterisk 1.6.2 Realtime Integration using Asterisk Database

• 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

Kamailio does authentication for registration. If successful, it notifies Asterisk with a new REGISTER that the phone is available at its IP.

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.

• 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:

sudo su -

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:

apt-get install mysql-server

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:

apt-get install libmysqlclient-dev

To install the UnixODBC devel libraries, run:

apt-get install unixodbc-dev

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:

apt-get install libmyodbc

Get Asterisk sources from http://www.asterisk.org. At this moment Asterisk 1.6.2.13 is the latest stable version.

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 

To enable ODBC storage for voicemail, run:

make menuselect

Then select option Voicemail Build Options, enable option ODBC_STORAGE. Save and exit.

Then compile and install:

make
make install

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:

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 

This database is required to store location records (phone contact addresses).

Use kamdbctl to create the database:

/usr/local/sbin/kamdbctl create

No other changes to Kamailio database structure are required. The SIP server will fetch the password from Asterisk database.

Execute next SQL script with mysql client:

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');

If you save it to asterisk.sql, then you can load it to MySQL server with:

mysql -u root -p <asterisk.sql

Before executing the SQL script, be sure you change the password for MySQL asterisk user, in line:

GRANT ALL ON asterisk.* to asterisk@localhost IDENTIFIED BY 'asterisk_password';

Edit /etc/odbcinst.ini and add:

[MySQL]
Description = MySQL driver
Driver = /usr/lib/odbc/libmyodbc.so
Setup = /usr/lib/odbc/libodbcmyS.so
CPTimeout =
CPReuse =
UsageCount = 1

Edit /etc/odbc.ini and add:

[MySQL-asterisk]
Description = MySQL Asterisk database
Trace = Off
TraceFile = stderr
Driver = MySQL
SERVER = localhost
USER = asterisk
PASSWORD = asterisk_password
PORT = 3306
DATABASE = asterisk 

Edit /etc/asterisk/res_odbc.conf and set:

[asterisk]
enabled => yes
dsn => MySQL-asterisk
username => asterisk
password => asterisk_password
pre-connect => yes

Edit /etc/asterisk/extconfig.conf and set:

sipusers => odbc,asterisk,sipusers
sippeers => odbc,asterisk,sipusers
sipregs => odbc,asterisk,sipregs
voicemail => odbc,asterisk,voiceboxes

In case you need to cache the realtime users, then edit /etc/asterisk/sip.conf and set:

rtcachefriends=yes 

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.

For testing purposes, here is a sample that can be plugged in /etc/asterisk/extensions.conf:

; 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

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:

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');

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:

#!define WITH_ASTERISK

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.

#!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();
	##}
}

• 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

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

• Install Kamailio 3.0.x from GIT
• Realtime Integration of Asterisk 1.4.x with Kamailio 1.5.x