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freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc

Kamailio 3.1.x and FreeSWITCH 1.0.6+ for Media Services and SBC

Author:
    Daniel-Constantin Mierla

Overview

The scope of this tutorial is to show how you can use Kamailio (former OpenSER) and FreeSWITCH to build a complete SIP/VoIP platform for large number of subscribers.

Kamailio is an open source SIP server, forked from SIP Express Router (SER) in 2005 under the name OpenSER. In July 2008, OpenSER was renamed to Kamailio because of trademark issues. Starting with November 2008, Kamailio and SER teams restarted development collaboration, resulting in version 3.0.0 being the first that allow to run Kamailio and SER modules (extensions) in same SIP server instance - practically it is the same source code, the differences are the database structure used to store SIP user profiles and default enabled modules. The latest stable release is 3.1.0, out on October 6, 2010.

One of outstanding features of Kamailio is ability of hosting large number of active users in a single instance (depending of hardware it can be 100 000+). Long development life ensures the stability required in real-time tele-communications and a broad set of features in handling SIP signaling.

FreeSWITCH is an open source multi-protocol softswitch, supporting SIP as well. It is a very attractive project from features and extensibility point of view. Its media processing capabilities makes FreeSWITCH a perfect fit for providing media services to Kamailio based platforms.

Any Kamailio version 3.1.x can be used (right now last released is 3.1.0, v3.1.1 is planed for release in few days). For FreeSWITCH I used the development version from GIT after release 1.0.6, but before any other official release (no 1.0.7 or what is going to be next).

Following services are handled in the scenario built within document:

  • kamailio
    • user authentication
    • user registration
    • user location
    • call routing
    • instant messaging and presence
  • freeswitch
    • voicemail
    • conference
    • SBC - this can be used for topology hiding, transcoding, prepaid or playing audio messages within calls
    • other media services (announcement, ivr, a.s.o)

Local users have 3 digit IDs (we will use users 101 102, and 103 for testing). Voice box ID is the same as user ID. Extensions for media services start with 4.

Kamailio and FreeSWITCH are installed on the same physical server (ip 192.168.178.23), using different ports:

  • kamailio: port 5060
  • freeswitch: port 5090 for internal profile and 5092 for external profile

Updates

This is the second release of this tutorial, first one was using previous major stable release of Kamailio, v3.0.x. You can read it at:

Besides upgrade to Kamailio v3.1.0, this version of tutorial includes new features:

  • in Kamailio
    • IP authentication - can be enabled via define WITH_IPAUTH
    • TLS support - can be enabled via define WITH_TLS - TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure FreeSwitch only on UDP
    • detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD - banning automatically 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
    • after authentication, calls are routed to FreeSwitch and then back to Kamailio
  • in Freeswitch
    • calls coming with dialed extension starting with kb- are routed back to Kamailio. If the calls fail, they are sent to voicemail box

Call Initiation

Call authentication is handled by Kamailio. When a new call arrives and it is authenticated, then:

  • if the destination user is not online, it is sent to FreeSwitch directly to voicemail box
  • if the destination user is online, Kamailio forwards it to FreeSwitch. There you can apply FreeSwitch specific functionality, like playing audio during early session, set a maximum call duration, enforce specific codecs, etc.
  • FreeSwitch sends the call back to Kamailio, which based on location service, will deliver the call to callee
  • if the callee does not answer, FreeSwitch will redirect it to voicemail box

Kamailio Configuration

Installation

A step by step installation tutorial for Kamailio 3.1.x is available at:

It is no special tunning needed for this tutorial, just install it as usual. You can use MySQL or your preferred database server, has no relevance here. But if you choose another database type, be sure you update the Kamailio config file, the one provided here uses MySQL.

We try to keep FreeSWITCH and Kamailio installations as much as possible independent one from the other.

Config File

Kamailio has one config file by default, named kamailio.cfg. Starting with 3.0.0, you can split it in several files and use include_file to merge the pieces in main config file.

Another important features brought since 3.0.0 are 'define' directives, making easy to enable/disable features. This concept is used here as well, therefore you can see the changes done for FreeSWITCH integration by following define WITH_FREESWITCH (i.e., config parts in between #!ifdef WITH_FREESWITCH … #!endif).

#!KAMAILIO
 
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_FREESWITCH
 
#
# 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"
#!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
 
####### 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_FREESWITCH
freeswitch.bindip = "192.168.178.23" desc "FreeSWITCH IP Address"
freeswitch.bindport = "5090" desc "FreeSWITCH 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
 
# ----------------- 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)
modparam("rr", "append_fromtag", 0)
 
 
# ----- 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", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
 
# ----- 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);
 
	#!ifdef WITH_FREESWITCH
	# save callee ID
	$avp(callee) = $rU;
	route(FSDISPATCH);
	#!endif
 
	# 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
 
#!ifdef WITH_CFGSAMPLES
	/* 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");
	}
#!endif
 
	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
			}
			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();
 
		exit;
	}
}
 
# USER location service
route[LOCATION] {
 
#!ifdef WITH_ALIASDB
	# search in DB-based aliases
	alias_db_lookup("dbaliases");
#!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;
		}
	}
 
	# 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
	if (is_method("REGISTER"))
	{
		# authenticate the REGISTER requests (uncomment to enable auth)
		if (!www_authorize("$td", "subscriber"))
		{
			www_challenge("$td", "0");
			exit;
		}
 
		if ($au!=$tU)
		{
			sl_send_reply("403","Forbidden auth ID");
			exit;
		}
	} else {
 
#!ifdef WITH_FREESWITCH
	if(route(FSINBOUND))
		return;
#!endif
 
#!ifdef WITH_IPAUTH
		if(allow_source_address())
		{
			# source IP allowed
			return;
		}
#!endif
 
		# authenticate if from local subscriber
		if (from_uri==myself)
		{
			if (!proxy_authorize("$fd", "subscriber")) {
				proxy_challenge("$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_FREESWITCH
# FreeSWITCH routing blocks
route[FSINBOUND] {
	if($si== $sel(cfg_get.freeswitch.bindip)
			&& $sp==$sel(cfg_get.freeswitch.bindport))
		return 1;
	return -1;
}
 
route[FSDISPATCH] {
	if(!is_method("INVITE"))
		return;
	if(route(FSINBOUND))
		return;
 
	# dial number selection
	switch($rU) {
		case /"^41$":
			# 41 - voicebox menu
			# allow only authenticated users
			if($au==$null)
			{
				sl_send_reply("403", "Not allowed");
				exit;
			}
			$rU = "vm-" + $au;
		break;
		case /"^441[0-9][0-9]$":
			# starting with 44 folowed by 1XY - direct call to voice box
			strip(2);
			route(FSVBOX);
		break;
		case /"^433[01][0-9][0-9]$":
			# starting with 433 folowed by (0|1)XY - conference
			strip(2);
		break;
		case /"^45[0-9]+$":
			strip(2);
		break;
		default:
			# offline - send to voicebox
			if (!registered("location"))
			{
				route(FSVBOX);
				exit;
			}
			# online - do bridging
			prefix("kb-");
			if(is_method("INVITE"))
			{
				# in case of failure - re-route to FreeSWITCH VoiceMail
				t_on_failure("FAIL_FSVBOX");
			}
	}
	route(FSRELAY);
	exit;
}
 
route[FSVBOX] {
	if(!($rU=~"^1[0-9][0-9]+$"))
		return;
	prefix("vb-");
	route(FSRELAY);
}
 
# Send to FreeSWITCH
route[FSRELAY] {
	$du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":"
			+ $sel(cfg_get.freeswitch.bindport);
	if($var(newbranch)==1)
	{
		append_branch();
		$var(newbranch) = 0;
	}
	route(RELAY);
	exit;
}
 
#!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();
	##}
}
 
#!ifdef WITH_FREESWITCH
failure_route[FAIL_FSVBOX] {
#!ifdef WITH_NAT
	if (is_method("INVITE")
			&& (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) {
		unforce_rtp_proxy();
	}
#!endif
 
	if (t_is_canceled()) {
		exit;
	}
 
	if (t_check_status("486|408")) {
		# re-route to FreeSWITCH VoiceMail
		$rU = $avp(callee);
		$var(newbranch) = 1;
		route(FSVBOX);
	}
}
#!endif

SIP Users

Be sure you updated /usr/local/etc/kamailio/kamctlrc and you set properly SIP_DOMAIN and DBENGINE:

...
SIP_DOMAIN=192.168.178.23      # IP address or hostname for Kamailio
...
DBENGINE=MYSQL
...

Create some test users in Kamailio with kamctl:

kamctl add userid password email
kamctl add 101 101 101@mysipserver.com
kamctl add 102 102 102@mysipserver.com
kamctl add 103 103 103@mysipserver.com

Config Remarks

Routing to FreeSWITCH:

  • dial 41 to listen to your voice mail service
  • dial 44XYZ (e.g., 44101) to leave a message for user XYZ
  • dial 4330XY or 4331XY to dial in 30XY or 31XY conference rooms on FreeSWITCH
  • dial 45EXTEN to call extension EXTEN on FreeSWITCH (e.g., 459995 calls to 9995 on FreeSWITCH, which is echo service in default dialplan)
  • if dialed user is offline, call is sent to callee's voice mail box on FreeSWITCH
  • if dialed user is online, its extensions is prefixed with kb- and then sent to FreeSwitch, which will send it back to Kamailio after acting as SBC and media relay
  • calls coming from FreeSwitch are not authenticated again, they are handled via location service and forwarded
  • if callee does not answer or sends back busy, call is sent to callee's voice mail box on FreeSWITCH

To define the IP address and port, update the custom global parameters in class freeswitch:

#!ifdef WITH_FREESWITCH
freeswitch.bindip = "192.168.178.23" desc "FreeSWITCH IP Address"
freeswitch.bindport = "5090" desc "FreeSWITCH Port"
#!endif

FreeSWITCH Configuration

Installation

Just do the standard FreeSWITCH installation, a tutorial is available at:

Config Files

SIP

Since this tutorial uses same IP but different ports, update your SIP profiles setting, setting the port accordingly.

Internal profiles:

<param name="sip-port" value="5090"/>

External profile:

<param name="sip-port" value="5092"/>

Also, for internal profile, disable call authentication. SIP traffic from Kamailio will be trusted and verified by ACL.

<param name="auth-calls" value="false"/>

ACL

Update ACL to allow traffic from Kamailio IP, edit conf/autoload_configs/acl.conf.xml and add to domains list:

       <node type="allow" cidr="192.168.178.23/32"/>

After that should look like:

     <list name="domains" default="deny">
       <node type="allow" domain="$${domain}"/>
       <node type="allow" cidr="192.168.178.23/32"/>
     </list>

Dialplan

First direct the SIP traffic sent by Kamailio from public to default, edit conf/dialplan/public.xml and add:

    <extension name="from_kamailio">
      <condition field="network_addr" expression="^192\.168\.178\.23$" />
      <condition field="destination_number" expression="^(.+)$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

Then in conf/dialplan/default.xml, add new extensions for handling calls to Voice Box services and routing calls back to Kamailio:

    <extension name="vbox">
      <condition field="destination_number" expression="^vb-(1[0-9][0-9])$">
	    <action application="answer"/>
	    <action application="voicemail" data="default ${domain_name} $1"/>
      </condition>
    </extension>
 
    <extension name="vmenu">
      <condition field="destination_number" expression="^vm-(1[0-9][0-9])$">
	    <action application="voicemail" data="check default ${domain_name} $1"/>
      </condition>
    </extension>
 
    <extension name="kbridge">
      <condition field="destination_number" expression="^kb-(.+)$">
		  <action application="set" data="proxy_media=true"/>
		  <action application="set" data="call_timeout=50"/>
		  <action application="set" data="continue_on_fail=true"/>
		  <action application="set" data="hangup_after_bridge=true"/>
		  <action application="set" data="sip_invite_domain=192.168.178.23"/>
		  <action application="export" data="sip_contact_user=ufs"/>
		  <action application="bridge" data="sofia/$${domain}/$1@192.168.178.23"/>
		  <action application="answer"/>
		  <action application="voicemail" data="default ${domain_name} $1"/>
      </condition>
    </extension>

First extension is to leave a voice message to callee. Second is to listen to caller's voice messages.

The third is to loop back calls to Kamailio. This is important at least for NAT traversal over symmetric NAT routes. If the call is not answered, will be directed to voicemail after 50 seconds. This is the part where you can add more SBC-like functions, e.g., enforce specific codecs, call prepaid application, play audio during early session.

User Directory

You have to create some file that hold the user profiles to provide voicemail services. Here we put the XML files on the local file system.

However, with freeswitch is easy to get them dynamically, i.e., via HTTP (invoking PHP, CGI, etc.) or calling an application (written in Lua for example) that goes to database (can be Kamailio's database) and return the user profiles - I let that for a future article: Kamailio and FreeSWITCH realtime integration.

For now, add three files in directory/default:

  • 101.xml
<include>
  <user id="101">
    <params>
      <param name="vm-password" value="1001"/>
    </params>
    <variables>
      <variable name="accountcode" value="101"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 101"/>
      <variable name="effective_caller_id_number" value="101"/>
    </variables>
  </user>
</include>
  • 102.xml
<include>
  <user id="102">
    <params>
      <param name="vm-password" value="1002"/>
    </params>
    <variables>
      <variable name="accountcode" value="102"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 102"/>
      <variable name="effective_caller_id_number" value="102"/>
    </variables>
  </user>
</include>
  • 103.xml
<include>
  <user id="103">
    <params>
      <param name="vm-password" value="1003"/>
    </params>
    <variables>
      <variable name="accountcode" value="103"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 103"/>
      <variable name="effective_caller_id_number" value="103"/>
    </variables>
  </user>
</include>

Since we use them for voicemail, the important field is vm-password which represents the voicemail PIN. You can omit other files if you don't need them.

Testing

Once you have installed and configured kamailio and freeswitch, configure some phones with users 101, 102 and 103 to register with Kamailio.

The start calling between them:

  • if you answer, call is connected, you should have voice
  • if you reject, caller should get to callee's voice box
  • if you don't answer, caller should get to callee's voice bos

Call into a conference: dial 433001 from your phones.

Leave a voice message to user 101: dial 44101

Listen your voice messages: dial 41

See Also



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