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Documentation -> Tutorials -> OpenSIPS FreeSwitch IntegrationThis page has been visited 115351 times. Table of Content (hide)
1. Realtime OpenSIPS - FreeSWITCH IntegrationAuthor Giovanni Maruzzelli <gmaruzz at gmail dot com> This tutorial is made for OpenSIPS 1.8.x and FreeSWITCH 1.2.x 1.1 ScopeThis tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. This tutorial presents the concept and implementation of a realtime integration of OpenSIPS SIP server and FreeSWITCH media server. OpenSIPS is used a SIP server - users are registering with it, it routes calls, etc - while the purpose of FreeSWITCH is to provide a full set of media services - like voicemail, conference, announcements, etc. It is a realtime integration because both OpenSIPS and FreeSWITCH are provisioned in the same time when comes to user accounts - when creating a new OpenSIPS user, automatically FreeSWITCH will learn about it an provide and configure all necessary media services for it. Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts) via a shared mysql database. All FreeSWITCH functionalities will be available to OpenSIPS users by prefixing "*" (eg: star) to the extension dialed. *1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234 1.2 Setup presentationThis tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. The following services will be offered by FreeSWITCH by this integrated configuration:
2. Installation and First Configuration2.1 Install LinuxStart from a fresh Debian 6 (Squeeze) base install, bring in the latest updates and reboot apt-get clean apt-get update apt-get upgrade apt-get dist-upgrade reboot 2.2 Install the prerequisites
apt-get install build-essential \
subversion automake autoconf wget libtiff4-dev libtool \
libncurses5-dev git-core libcurl4-openssl-dev libjpeg-dev \
mysql-server libmysqlclient-dev mysql-client \
unixodbc-dev unixodbc libmyodbc flex bison libncurses-dev \
build-essential openssl bison flex perl libdbi-perl \
libdbd-mysql-perl libdbd-pg-perl libfrontier-rpc-perl \
libterm-readline-gnu-perl libberkeleydb-perl ncurses-dev \
libpcre3-dev libxml2-dev libxmlrpc-c-dev libpcre3 libxml2 \
perl libdbi-perl libdbd-mysql-perl libfrontier-rpc-perl \
libterm-readline-gnu-perl libberkeleydb-perl apache2 \
libapache2-mod-php5 php5 php5-cli php5-gd php5-mysql php-pear \
php5-xmlrpc
2.3 Compile and install OpenSIPScd /usr/src svn co https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.8 opensips_1_8 cd opensips_1_8 make menuconfig Go to 'Configure Compile Options' and then 'Configure Excluded Modules'. Select the db_mysql, db_unixodbc, dialplan, mi_xmlrpc, presence*, pua*, regex modules with the spacebar key, then hit the 'q' key. Go to 'Configure Install Prefix' and set the prefix to /usr/local/opensips . Hit 'Save Changes' and then compile and install OpenSIPS. 2.4 Create OpenSIPS Database and Configuration Filecd /usr/local/opensips/etc/opensips/ vi opensipsctlrc modify like the follow (you'll find in the next section the file ready to be copied and pasted). # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) DBNAME=opensips # database path used by dbtext or db_berkeley # DB_PATH="/usr/local/etc/opensips/dbtext" ## database read/write user DBRWUSER=opensips ## password for database read/write user DBRWPW="mydbpassword" ## database super user (for ORACLE this is 'scheme-creator' user) DBROOTUSER="root" then create the database cd ../.. cd sbin ./opensipsdbctl create you can then check database creation success with mysql -D opensips -p then create the OpenSIPS configuration file ./osipsconfig Go to 'Generate OpenSIPS Script' and then to 'Residential Script', 'Configure Residential Script'. Add ENABLE_TCP, USE_ALIASES, USE_AUTH, USE_DBACC, USE_DBUSRLOC, USE_DIALOG, USE_DIALPLAN, VM_DIVERSION Hit 'q' and then go to 'Generate Residential Script', which will generate a CFG file in your install path, in the /usr/local/opensips/etc/opensips/ folder. Change its name, and edit it (you'll find in the next section the file ready to be copied and pasted). cd ../etc/opensips mv opensips_residential_2012-12-19_8\:51\:27.cfg opensips_residential_01.cfg vi opensips_residential_01.cfg Follow this guidelines
#follow CUSTOMIZE ME
# modify db password
# modify
$du = "sip:127.0.0.2:5060"; # CUSTOMIZE ME
#to
$du = "sip:192.168.1.110:5090"; # CUSTOMIZE ME
#modify mpath to /usr/local/opensips/lib64/opensips/modules/
#modify from
#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
#to
#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
modparam("uri", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
(you'll find in the next section the file ready to be copied and pasted) 2.5 Install OpenSIPS init filesthen install and edit OpenSIPS init script cd /usr/src/opensips_1_8/ cd packaging/ cd debian cp opensips.init /etc/init.d/opensips chmod +x /etc/init.d/opensips vi /etc/init.d/opensips change the following lines, with the correct location of OpenSIPS executable, comment out those lines for debug, and edit the OPTIONS line as instructed DAEMON=/usr/local/opensips/sbin/opensips #CUSTOMIZE #comment out if [ "$1" != "debug" ]; then #comment out check_fork #comment out fi # edit OPTIONS adding -f config-file OPTIONS="-P $PIDFILE -m $S_MEMORY -M $P_MEMORY -u $USER -g $GROUP -f /usr/local/opensips/etc/opensips/opensips_residential_01.cfg" then edit the accessory files cp opensips.default /etc/default/opensips vi /etc/default/opensips edit it as in RUN_OPENSIPS=yes USER=root GROUP=root S_MEMORY=128 edit the system logger configuration file so to instruct it to convey OpenSIPS info to the right file vi /etc/rsyslog.conf add at file's end: local1.* -/var/log/opensips.log then restart the syslogger and OpenSIPS /etc/init.d/rsyslog restart /etc/init.d/opensips start 2.6 Install OpenSIPS Control PanelThis is a web configuration and management tool we will use to manage all our platform, both OpenSIPS and FreeSWITCH (eg we'll manage OpenSIPS, actually, but FreeSWITCH will source its user management data from the same database used by OpenSIPS and managed with OpenSIP-CP). cd /var/www svn co https://opensips-cp.svn.sourceforge.net/svnroot/opensips-cp/trunk opensips-cp pear install mdb2 pear install mdb2#mysql cd opensips-cp vi config/db.inc.php edit it like this: //database connection user $config->db_user = "opensips"; //CUSTOMIZE //database connection password $config->db_pass = "mydbpassword"; //CUSTOMIZE then edit the other config files vi config/boxes.global.inc.php make it like this: //## use fifo //$boxes[$box_id]['mi']['conn']="127.0.0.1:8000"; $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo"; //#comment out with double slashes monit stuff // monit host:port //$boxes[$box_id]['monit']['conn']="192.168.0.1:2812"; //$boxes[$box_id]['monit']['user']="admin"; //$boxes[$box_id]['monit']['pass']="pass"; //$boxes[$box_id]['monit']['has_ssl']=1; vi config/tools/system/dialog/local.inc.php make it like this: $box[1]['mi']['conn']="/tmp/opensips_fifo";; vi config/tools/system/dispatcher/local.inc.php make it like this: $box[1]['mi']['conn']="/tmp/opensips_fifo"; then add an alias to the apache configuration vi /etc/apache2/apache2.conf add at end of the file: Alias /cp /var/www/opensips-cp/web then change the ownership of the files, do the logging part and the CDR related editing
chown www-data.www-data /var/www/opensips-cp/config/access.log
pear install log
mysql -Dopensips -p < config/tools/admin/add_admin/ocp_admin_privileges.mysql
mysql -Dopensips -p -e "INSERT INTO ocp_admin_privileges (username,password,ha1,available_tools,permissions) values
('admin','admin',md5('admin:admin'),'all','all');"
mysql -Dopensips -p < config/tools/system/cdrviewer/cdrs.mysql
mysql -Dopensips -p < config/tools/system/cdrviewer/opensips_cdrs.mysql
mysql -Dopensips -p < config/tools/system/smonitor/tables.mysql
vi /var/www/opensips-cp/cron_job/generate-cdrs_mysql.sh
edit to look like: mysql -h $HOSTNAME -u $USER -p$PASS -e "call opensips_cdrs(); " $DATABASE then activate the CDR cronjob vi /etc/crontab add the following lines to it */3 * * * * root /var/www/opensips-cp/cron_job/generate-cdrs_mysql.sh * * * * * root php /var/www/opensips-cp/cron_job/get_opensips_stats.php > /dev/null and restart both apache and OpenSIPS /etc/init.d/apache2 restart /etc/init.d/opensips restart /etc/init.d/cron restart 2.7 FreeSWITCH compile and installLet's install FreeSWITCH and go for a coffee in the while (will take time) cd /usr/src git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git cd freeswitch/ ./bootstrap.sh && ./configure && make && make install && make hd-sounds-install && make hd-moh-install && make samples 2.8 FreeSWITCH listening portsEdit the SIP ports for FreeSWITCH vi /usr/local/freeswitch/conf/vars.xml to be <X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5091"/> 2.9 Install OpenSIPS and FreeSWITCH configs and database script, and ODBC configsBack up the original files for your reference cp /usr/local/opensips/etc/opensips/opensipsctlrc /usr/local/opensips/etc/opensips/opensipsctlrc.originale cp /usr/local/freeswitch/conf/dialplan/public.xml /usr/local/freeswitch/conf/dialplan/public.xml.original cp /usr/local/freeswitch/conf/vars.xml /usr/local/freeswitch/conf/vars.xml.original cp /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml.original cp /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml.original cp /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml.original cp /etc/odbcinst.ini /etc/odbcinst.ini.original cp /etc/odbc.ini /etc/odbc.ini.original Files' content follows, but first the files' list: /usr/local/opensips/etc/opensips/opensipsctlrc /usr/local/opensips/etc/opensips/opensips_residential_01.cfg /usr/local/freeswitch/conf/dialplan/public.xml /usr/local/freeswitch/scripts/config.lua /usr/local/freeswitch/scripts/xml_handler.lua /usr/local/freeswitch/conf/vars.xml /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml /etc/odbcinst.ini /etc/odbc.ini You will copy and paste each one the files, overwriting the existing ones you just backed-up. 3. Scripts and Configuration files you can Copy and Paste, with explanationsBelow the files' content, with some explanation, file by file. You copy and paste each one the following files, overwriting the existing ones (you just made a backup of them). Then you can change them to your like, eg: changing the "mydbpasswd" password and the IP addresses. The files as they are here compose a working installation, provided you followed this tutorial step by step, and you use the same IP addresses. All that has been changed from original installed files is marked as "CUSTOMIZE" Follow "CUSTOMIZE", Neo. 3.1 /usr/local/opensips/etc/opensips/opensipsctlrcIn this file, that configures the behavior of the opensipsctrl utilities, you modify the DB-related values. # $Id: opensipsctlrc 9049 2012-05-24 14:03:31Z osas $ # # The OpenSIPS configuration file for the control tools. # # Here you can set variables used in the opensipsctl and opensipsdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain # SIP_DOMAIN=opensips.org ## chrooted directory # $CHROOT_DIR="/path/to/chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, or DBTEXT, ## by default none is loaded # If you want to setup a database with opensipsdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host #CUSTOMIZE DBHOST=localhost ## database name (for ORACLE this is TNS name) #CUSTOMIZE DBNAME=opensips # database path used by dbtext or db_berkeley # DB_PATH="/usr/local/etc/opensips/dbtext" ## database read/write user #CUSTOMIZE DBRWUSER=opensips ## password for database read/write user #CUSTOMIZE DBRWPW="mydbpassword" ## database super user (for ORACLE this is 'scheme-creator' user) #CUSTOMIZE DBROOTUSER="root" # user name column # USERCOL="username" # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2020-05-28 21:32:15" # DEFAULT_ALIASES_EXPIRES=$FOREVER # DEFAULT_Q="1.0" # DEFAULT_CALLID="Default-Call-ID" # DEFAULT_CSEQ="13" # DEFAULT_LOCATION_EXPIRES=$FOREVER # Program to calculate a message-digest fingerprint # MD5="md5sum" # awk tool # AWK="awk" # grep tool # GREP="grep" # sed tool # SED="sed" # Describe what additional tables to install. Valid values for the variables # below are yes/no/ask. With ask (default) it will interactively ask the user # for an answer, while yes/no allow for automated, unassisted installs. # # If to install tables for the modules in the EXTRA_MODULES variable. # INSTALL_EXTRA_TABLES=ask # If to install presence related tables. # INSTALL_PRESENCE_TABLES=ask # Define what module tables should be installed. # If you use the postgres database and want to change the installed tables, # then you must also adjust the STANDARD_TABLES or EXTRA_TABLES variable # accordingly in the opensipsdbctl.base script. # opensips standard modules # STANDARD_MODULES="standard acc domain group permissions registrar usrloc # msilo alias_db uri_db speeddial avpops auth_db pdt dialog # dispatcher dialplan drouting nathelper load_balancer" # opensips extra modules # EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist b2b registrant" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none # ALIASES_TYPE="DB" ## control engine: FIFO or UNIXSOCK ## - default FIFO # CTLENGINE=xmlrpc ## path to FIFO file # OSIPS_FIFO="/tmp/opensips_fifo" ## MI_CONNECTOR control engine: FIFO, UNIXSOCK, UDP, XMLRPC # MI_CONNECTOR=FIFO:/tmp/opensips_fifo # MI_CONNECTOR=UNIXSOCK:/tmp/opensips.sock # MI_CONNECTOR=UDP:192.168.2.133:8000 # MI_CONNECTOR=XMLRPC:192.168.2.133:8000 ## check ACL names; default on (1); off (0) # VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## verbose - debug purposes - default '0' # VERBOSE=1 ## do (1) or don't (0) store plaintext passwords ## in the subscriber table - default '1' # STORE_PLAINTEXT_PW=0 ## do not display the output highlighted # NOHLPRINT=1 ## OPENSIPS START Options ## PID file path - default is: /var/run/opensips.pid # PID_FILE=/var/run/opensips.pid ## Extra start options - default is: not set # example: start opensips with 64MB share memory: STARTOPTIONS="-m 64" # STARTOPTIONS= 3.2 /usr/local/opensips/etc/opensips/opensips_residential_01.cfgWe created before using the osipsconfig utility a residential script, and we requested the following options: ENABLE_TCP, USE_ALIASES, USE_AUTH, USE_DBACC, USE_DBUSRLOC, USE_DIALOG, USE_DIALPLAN, VM_DIVERSION. So, we have tcp in addition to udp connectivity, aliases on login, authorization checks, db based accounting and location service, dialog tracking, dialplan transformations, and diversion (redirection) to voicemail server in case of no answer, busy, declined, etc. In this file you modify DB related values, inserting one block that was missed, IP addresses, syslog facility, modules path (follow "CUSTOMIZE"). Then you add a route( marked "2") that checks if the called number begins with "*" (star) in which case strips the first * and sends the call to FreeSWITCH (marked "freeswitch"). Then you put the correct FreeSWITCH IP address where it redirects to VM (marked "freeswitch").
#
# $Id: opensips_residential.m4 9042 2012-05-17 13:57:10Z vladut-paiu $
#
# OpenSIPS residential configuration script
# by OpenSIPS Solutions <team@opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
# the "Residential" scenario.
# You can enable / disable more features / functionalities by
# re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
# http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#
####### Global Parameters #########
debug=3
log_stderror=no
log_facility=LOG_LOCAL1 # CUSTOMIZE ME
fork=yes
children=4
/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes
/* uncomment the next line to enable the auto temporary blacklisting of
not available destinations (default disabled) */
#disable_dns_blacklist=no
/* uncomment the next line to enable IPv6 lookup after IPv4 dns
lookup failures (default disabled) */
#dns_try_ipv6=yes
/* comment the next line to enable the auto discovery of local aliases
based on revers DNS on IPs */
auto_aliases=no
listen=udp:192.168.1.110:5060 # CUSTOMIZE ME
disable_tcp=no
listen=tcp:192.168.1.110:5060 # CUSTOMIZE ME
disable_tls=yes
####### Modules Section ########
#set module path
mpath="/usr/local/opensips/lib64/opensips/modules/" # CUSTOMIZE ME
#### SIGNALING module
loadmodule "signaling.so"
#### StateLess module
loadmodule "sl.so"
#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)
#### MAX ForWarD module
loadmodule "maxfwd.so"
#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"
#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)
#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
modparam("uri", "db_url", # ADD and CUSTOMIZE ME
"mysql://opensips:mydbpassword@localhost/opensips") # ADD and CUSTOMIZE ME
#### MYSQL module
loadmodule "db_mysql.so"
#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", 10)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", 7)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we 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)
modparam("acc", "failed_transaction_flag", 3)
/* account triggers (flags) */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")
#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600) # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
#### DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
"mysql://opensips:mydbpassword@localhost/opensips") # CUSTOMIZE ME
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
# validate the sequential request against dialog
if ( $DLG_status!=NULL && !validate_dialog() ) {
xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
## exit;
}
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after
# a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
if ( !(is_method("REGISTER") ) ) {
if (from_uri==myself)
{
# authenticate if from local subscriber
# authenticate all initial non-REGISTER request that pretend to be
# generated by local subscriber (domain from FROM URI is local)
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!db_check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
consume_credentials();
# caller authenticated
} else {
# if caller is not local, then called number must be local
if (!uri==myself) {
send_reply("403","Rely forbidden");
exit;
}
}
}
# preloaded route checking
if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
# create dialog with timeout
if ( !create_dialog("B") ) {
send_reply("500","Internal Server Error");
exit;
}
setflag(1); # do accounting
}
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
}
# requests for my domain
if (is_method("PUBLISH|SUBSCRIBE"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if ( proto==TCP || 0 ) setflag(7);
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases
alias_db_lookup("dbaliases");
# apply transformations from dialplan table
dp_translate("0","$rU/$rU");
#freeswitch
route(2); # ADD and CUSTOMIZE ME
# do lookup with method filtering
if (!lookup("location","m")) {
if (!db_does_uri_exist()) {
send_reply("420","Bad Extension");
exit;
}
# redirect to a different VM system
$du = "sip:192.168.1.110:5090"; # CUSTOMIZE ME #freeswitch
route(1);
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
}
if (!t_relay()) {
send_reply("500","Internal Error");
};
exit;
}
# freeswitch
route[2] {# ADD and CUSTOMIZE ME
if (!is_method("INVITE")) {
return;
}
# if the called number begins with "star" (*) then strip it and redirect to freeswitch
# (if it begins with two stars, eg: **, then one will be passed to FS)
if ($rU=~"^\*") {
strip(1);
$du = "sip:192.168.1.110:5090"; # CUSTOMIZE ME
route(1);
}
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
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;
##}
# redirect the failed to a different VM system
if (t_check_status("486|408")) {
$du = "sip:192.168.1.110:5090"; # CUSTOMIZE ME
# do not set the missed call flag again
route(1);
}
}
local_route {
if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
acc_db_request("200 Dialog Timeout", "acc");
}
}
3.3 /usr/local/freeswitch/conf/dialplan/public.xmlStarting from the freshly installed default FreeSWITCH "public" (eg: from outside, not trusted to access services and features) dialplan, you insert as the first "extension", aptly named "from_opensips", an instruction that checks if the call is coming from the OpenSIPS server IP address, in which case the call is transferred to the corresponding destination_number in the "default" dialplan (eg: can access the services and features as an internal user). You must edit the OpenSIPS IP address. Follow "CUSTOMIZE".
<!--
NOTICE:
This context is usually accessed via the external sip profile listening on port 5080.
It is recommended to have separate inbound and outbound contexts. Not only for security
but clearing up why you would need to do such a thing. You don't want outside un-authenticated
callers hitting your default context which allows dialing calls thru your providers and results
in Toll Fraud.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="public">
<extension name="from_opensips">
<condition field="network_addr" expression="^192\.168\.1\.110$"> <!--CUSTOMIZE-->
<action application="transfer" data="${destination_number} XML default"/>
</condition>
</extension>
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!--
Tag anything pass thru here as an outside_call so you can make sure not
to create any routing loops based on the conditions that it came from
the outside of the switch.
-->
<extension name="outside_call" continue="true">
<condition>
<action application="set" data="outside_call=true"/>
<action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
</condition>
</extension>
<extension name="call_debug" continue="true">
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
</extension>
<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<!--
You can place files in the public directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="public/*.xml"/>
<!--
If you have made it this far lets challenge the caller and if they authenticate
lets try what they dialed in the default context. (commented out by default)
-->
<!--
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>
<extension name="transfer_to_default">
<condition>
<action application="transfer" data="${destination_number} XML default"/>
</condition>
</extension>
-->
</context>
</include>
3.4 /usr/local/freeswitch/scripts/config.luaThis file you create ex-nihilo (eg: is not installed by FreeSWITCH). It is read by the xml_handler script, and configure it with the correct values for directories and database name, user and password. Edit it all, or at least edit "mydbpassword". Follow "CUSTOMIZE".
--switch directories
sounds_dir = "/usr/local/freeswitch/sounds";
recordings_dir = "/usr/local/freeswitch/recordings";
--database connection info
db_type = "mysql";
db_name = "opensips";
dsn_name = "opensips";
dsn_username = "opensips";
dsn_password = "mydbpassword"; --CUSTOMIZE
--additional info
tmp_dir = "";
3.5 /usr/local/freeswitch/scripts/xml_handler.luaThis file you create ex-nihilo (eg: is not installed by FreeSWITCH). It is executed by the mod_lua Lua module, and fetch from OpenSIPS database the values with which it constructs an xml directory snippet that is passed in real time to FreeSWITCH when FS needs info about users. You don't need to edit this file, but can be interesting to read. This file comes originally from FusionPBX ( www.fusionpbx.com ), an opensource web interface to configure and manage FreeSWITCH, and has been cannibalized for our tutorial purposes. Particularly, you can find there almost all the fields that are supported in FS directory (eg user management), so it can be extended easyly sourcing values from the same or other databases.
-- HEAVILY MODIFIED AND POSSIBLY BUGIFIED BY Giovanni Maruzzelli (gmaruzz@gmail.com)
-- xml_handler.lua
-- Part of FusionPBX
-- Copyright (C) 2010 Mark J Crane <markjcrane@fusionpbx.com>
-- All rights reserved.
--
-- Redistribution and use in source and binary forms, with or without
-- modification, are permitted provided that the following conditions are met:
--
-- 1. Redistributions of source code must retain the above copyright notice,
-- this list of conditions and the following disclaimer.
--
-- 2. Redistributions in binary form must reproduce the above copyright
-- notice, this list of conditions and the following disclaimer in the
-- documentation and/or other materials provided with the distribution.
--
-- THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES,
-- INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
-- AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE
-- AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY,
-- OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
-- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
-- INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
-- CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
-- ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-- POSSIBILITY OF SUCH DAMAGE.
-- HEAVILY MODIFIED AND POSSIBLY BUGIFIED BY Giovanni Maruzzelli (gmaruzz@gmail.com)
--set the debug level
debug["params"] = true;
debug["sql"] = true;
debug["xml_request"] = true;
debug["xml_string"] = true;
--show param debug info
if (debug["params"]) then
freeswitch.consoleLog("notice", "[xml_handler] Params:\n" .. params:serialize() .. "\n");
end
--get the params and set them as variables
local domain_name = params:getHeader("domain");
local purpose = params:getHeader("purpose");
local profile = params:getHeader("profile");
local key = params:getHeader("key");
local user = params:getHeader("user");
local user_context = params:getHeader("variable_user_context");
local call_context = params:getHeader("Caller-Context");
local destination_number = params:getHeader("Caller-Destination-Number");
local caller_id_number = params:getHeader("Caller-Caller-ID-Number");
--include the lua script
scripts_dir = string.sub(debug.getinfo(1).source,2,string.len(debug.getinfo(1).source)-(string.len(argv[0])+1));
include = assert(loadfile(scripts_dir .. "/config.lua"));
include();
--connect to the database
--ODBC - data source name
if (dsn_name) then
dbh = freeswitch.Dbh(dsn_name,dsn_username,dsn_password);
end
--FreeSWITCH core db handler
if (db_type == "sqlite") then
dbh = freeswitch.Dbh("core:"..db_path.."/"..db_name);
end
--handle the directory
if (XML_REQUEST["section"] == "directory" and key and user and domain_name) then
--prevent processing for invalid user
continue = true;
if (user == "*97") then
continue = false;
end
--get the extension from the database
if (continue) then
sql = "SELECT * FROM subscriber WHERE domain = '" .. domain_name .. "' and username = '" .. user .. "' ";
if (debug["sql"]) then
freeswitch.consoleLog("notice", "[xml_handler] SQL: " .. sql .. "\n");
end
dbh:query(sql, function(row)
--general
domain_uuid = "";
extension = row.username;
cidr = "";
number_alias = "";
--params
password = row.password;
vm_enabled = "true";
vm_password = row.password;
vm_attach_file = "true";
vm_keep_local_after_email = "true";
vm_mailto = row.email_address;
mwi_account = "";
auth_acl = "";
--variables
sip_from_user = "";
call_group = "";
hold_music = "";
toll_allow = "";
accountcode = "";
user_context = "default";
effective_caller_id_name = "";
effective_caller_id_number = "";
outbound_caller_id_name = "";
outbound_caller_id_number = "";
emergency_caller_id_number = "";
directory_full_name = "";
directory_visible = "";
directory_exten_visible = "";
limit_max = "";
limit_destination = "";
sip_force_contact = "";
sip_force_expires = "";
nibble_account = "";
sip_bypass_media = "";
--set the dial_string
dial_string = "{sip_invite_domain=" .. domain_name .. ",presence_id=" .. user .. "@" ..
domain_name .. "}${sofia_contact(" .. user .. "@" .. domain_name .. ")}";
end);
end
--set the xml array and then concatenate the array to a string
if (password) then
local xml = {}
table.insert(xml, [[<?xml version="1.0" encoding="UTF-8" standalone="no"?>]]);
table.insert(xml, [[<document type="freeswitch/xml">]]);
table.insert(xml, [[ <section name="directory">]]);
table.insert(xml, [[ <domain name="]] .. domain_name .. [[">]]);
if (number_alias) then
if (cidr) then
table.insert(xml, [[ <user id="]] .. extension .. [["]] .. cidr .. number_alias .. [[>]]);
else
table.insert(xml, [[ <user id="]] .. extension .. [["]] .. number_alias .. [[>]]);
end
else
if (cidr) then
table.insert(xml, [[ <user id="]] .. extension .. [["]] .. cidr .. [[>]]);
else
table.insert(xml, [[ <user id="]] .. extension .. [[">]]);
end
end
table.insert(xml, [[ <params>]]);
table.insert(xml, [[ <param name="password" value="]] .. password .. [["/>]]);
table.insert(xml, [[ <param name="vm-enabled" value="]] .. vm_enabled .. [["/>]]);
if (string.len(vm_mailto) > 0) then
table.insert(xml, [[ <param name="vm-password" value="]] .. vm_password .. [["/>]]);
table.insert(xml, [[ <param name="vm-email-all-messages" value="]] .. vm_enabled ..[["/>]]);
table.insert(xml, [[ <param name="vm-attach-file" value="]] .. vm_attach_file .. [["/>]]);
table.insert(xml, [[ <param name="vm-keep-local-after-email" value="]] .. vm_keep_local_after_email .. [["/>]]);
table.insert(xml, [[ <param name="vm-mailto" value="]] .. vm_mailto .. [["/>]]);
end
if (string.len(mwi_account) > 0) then
table.insert(xml, [[ <param name="MWI-Account" value="]] .. mwi_account .. [["/>]]);
end
if (string.len(auth_acl) > 0) then
table.insert(xml, [[ <param name="auth-acl" value="]] .. auth_acl .. [["/>]]);
end
table.insert(xml, [[ <param name="dial-string" value="]] .. dial_string .. [["/>]]);
table.insert(xml, [[ </params>]]);
table.insert(xml, [[ <variables>]]);
table.insert(xml, [[ <variable name="domain_uuid" value="]] .. domain_uuid .. [["/>]]);
if (user_context ~= "default" and user_context ~= "public" and user_context ~= "features") then
table.insert(xml, [[ <variable name="domain_name" value="]] .. user_context .. [["/>]]);
end
table.insert(xml, [[ <variable name="caller_id_name" value="]] .. sip_from_user .. [["/>]]);
table.insert(xml, [[ <variable name="caller_id_number" value="]] .. sip_from_user .. [["/>]]);
if (string.len(call_group) > 0) then
table.insert(xml, [[ <variable name="call_group" value="]] .. call_group .. [["/>]]);
end
if (string.len(hold_music) > 0) then
table.insert(xml, [[ <variable name="hold_music" value="]] .. hold_music .. [["/>]]);
end
if (string.len(toll_allow) > 0) then
table.insert(xml, [[ <variable name="toll_allow" value="]] .. toll_allow .. [["/>]]);
end
if (string.len(accountcode) > 0) then
table.insert(xml, [[ <variable name="accountcode" value="]] .. accountcode .. [["/>]]);
end
table.insert(xml, [[ <variable name="user_context" value="]] .. user_context .. [["/>]]);
if (string.len(effective_caller_id_name) > 0) then
table.insert(xml, [[ <variable name="effective_caller_id_name" value="]] .. effective_caller_id_name.. [["/>]]);
end
if (string.len(effective_caller_id_number) > 0) then
table.insert(xml, [[ <variable name="effective_caller_id_number" value="]] .. effective_caller_id_number.. [["/>]]);
end
if (string.len(outbound_caller_id_name) > 0) then
table.insert(xml, [[ <variable name="outbound_caller_id_name" value="]] .. outbound_caller_id_name .. [["/>]]);
table.insert(xml, [[ <variable name="outbound_caller_id_name" value="]] .. outbound_caller_id_name .. [["/>]]);
end
if (string.len(outbound_caller_id_number) > 0) then
table.insert(xml, [[ <variable name="outbound_caller_id_number" value="]] .. outbound_caller_id_number .. [["/>]]);
end
if (string.len(emergency_caller_id_number) > 0) then
table.insert(xml, [[ <variable name="emergency_caller_id_number" value="]] .. emergency_caller_id_number .. [["/>]]);
end
if (string.len(directory_full_name) > 0) then
table.insert(xml, [[ <variable name="directory_full_name" value="]] .. directory_full_name .. [["/>]]);
end
if (string.len(directory_visible) > 0) then
table.insert(xml, [[ <variable name="directory-visible" value="]] .. directory_visible .. [["/>]]);
end
if (string.len(directory_exten_visible) > 0) then
table.insert(xml, [[ <variable name="directory-exten-visible" value="]] .. directory_exten_visible .. [["/>]]);
end
if (string.len(limit_max) > 0) then
table.insert(xml, [[ <variable name="limit_max" value="]] .. limit_max .. [["/>]]);
else
table.insert(xml, [[ <variable name="limit_max" value="5"/>]]);
end
if (string.len(limit_destination) > 0) then
table.insert(xml, [[ <variable name="limit_destination" value="]] .. limit_destination .. [["/>]]);
end
if (string.len(sip_force_contact) > 0) then
table.insert(xml, [[ <variable name="sip_force_contact" value="]] .. sip_force_contact .. [["/>]]);
end
if (string.len(sip_force_expires) > 0) then
table.insert(xml, [[ <variable name="sip-force-expires" value="]] .. sip_force_expires .. [["/>]]);
end
if (string.len(nibble_account) > 0) then
table.insert(xml, [[ <variable name="nibble_account" value="]] .. nibble_account .. [["/>]]);
end
if (sip_bypass_media == "bypass-media") then
table.insert(xml, [[ <variable name="bypass_media" value="true"/>]]);
end
if (sip_bypass_media == "bypass-media-after-bridge") then
table.insert(xml, [[ <variable name="bypass_media_after_bridge" value="true"/>]]);
end
if (sip_bypass_media == "proxy-media") then
table.insert(xml, [[ <variable name="proxy_media" value="true"/>]]);
end
table.insert(xml, [[ <variable name="record_stereo" value="true"/>]]);
table.insert(xml, [[ <variable name="transfer_fallback_extension" value="operator"/>]]);
table.insert(xml, [[ <variable name="export_vars" value="domain_name"/>]]);
table.insert(xml, [[ </variables>]]);
table.insert(xml, [[ </user>]]);
table.insert(xml, [[ </domain>]]);
table.insert(xml, [[ </section>]]);
table.insert(xml, [[</document>]]);
XML_STRING = table.concat(xml, "\n");
else
XML_STRING = "";
end
--send the xml to a file
if (debug["xml_string"]) then
local file = assert(io.open("/tmp/directory.xml", "w"));
file:write(XML_STRING);
file:close();
end
--send the xml to the console
if (debug["xml_string"]) then
freeswitch.consoleLog("notice", "[xml_handler] XML_STRING: \n" .. XML_STRING .. "\n");
end
end
if (debug["xml_request"]) then
freeswitch.consoleLog("notice", "[xml_handler] Section: " .. XML_REQUEST["section"] .. "\n");
freeswitch.consoleLog("notice", "[xml_handler] Tag Name: " .. XML_REQUEST["tag_name"] .. "\n");
freeswitch.consoleLog("notice", "[xml_handler] Key Name: " .. XML_REQUEST["key_name"] .. "\n");
freeswitch.consoleLog("notice", "[xml_handler] Key Value: " .. XML_REQUEST["key_value"] .. "\n");
end
3.6 /usr/local/freeswitch/conf/vars.xmlThis file contains the variables that are interpolated into FreeSWITCH configuration files when they are sourced the first time at FreeSWITCH startup (eg: NOT when you "reload" FreeSWITCH). Here you do not have to change nothing, we configured FreeSWITCH ports before. Anyway, they're marked "CUSTOMIZE".
<include>
<!-- Preprocessor Variables
These are introduced when configuration strings must be consistent across modules.
NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
toll fraud in the future. It's your responsibility to secure your own system.
This default config is used to demonstrate the feature set of FreeSWITCH.
WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
-->
<X-PRE-PROCESS cmd="set" data="default_password=1234"/>
<!-- Did you change it yet? -->
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
<!--
This setting is what sets the default domain FreeSWITCH will use if all else fails.
FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does
affect the sip authentication. Please review conf/directory/default.xml for more
information on this topic.
-->
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
<X-PRE-PROCESS cmd="set" data="use_profile=internal"/>
<!--
Enable ZRTP globally you can override this on a per channel basis
http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
-->
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
<!--
Examples of codec options: (module must be compiled and loaded)
codecname[@8000h|16000h|32000h[@XXi]]
XX is the frame size must be multples allowed for the codec
FreeSWITCH can support 10-120ms on some codecs.
We do not support exceeding the MTU of the RTP packet.
iLBC@30i - iLBC using mode=30 which will win in all cases.
DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
speex@8000h@20i - Speex 8kHz using 20ms ptime.
speex@16000h@20i - Speex 16kHz using 20ms ptime.
speex@32000h@20i - Speex 32kHz using 20ms ptime.
BV16 - BroadVoice 16kb/s narrowband, 8kHz
BV32 - BroadVoice 32kb/s wideband, 16kHz
G7221@16000h - G722.1 16kHz (aka Siren 7)
G7221@32000h - G722.1C 32kHz (aka Siren 14)
CELT@32000h - CELT 32kHz, only 10ms supported
CELT@48000h - CELT 48kHz, only 10ms supported
GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
These are the passthru audio codecs:
G729 - G729 in passthru mode. (mod_g729)
G723 - G723.1 in passthru mode. (mod_g723_1)
AMR - AMR in passthru mode. (mod_amr)
These are the passthru video codecs: (mod_h26x)
H261 - H.261 Video
H263 - H.263 Video
H263-1998 - H.263-1998 Video
H263-2000 - H.263-2000 Video
H264 - H.264 Video
RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.
96 - AMR
97 - iLBC (30)
98 - iLBC (20)
99 - Speex 8kHz, 16kHz, 32kHz
100 -
101 - telephone-event
102 -
103 -
104 -
105 -
106 - BV16
107 - G722.1 (16kHz)
108 -
109 -
110 -
111 -
112 -
113 -
114 - CELT 32kHz, 48kHz
115 - G722.1C (32kHz)
116 -
117 - SILK 8kHz
118 - SILK 12kHz
119 - SILK 16kHz
120 - SILK 24kHz
121 - AAL2-G726-40 && G726-40
122 - AAL2-G726-32 && G726-32
123 - AAL2-G726-24 && G726-24
124 - AAL2-G726-16 && G726-16
125 -
126 -
127 - BV32
-->
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
<!--
xmpp_client_profile and xmpp_server_profile
xmpp_client_profile can be any string.
xmpp_server_profile is appended to "dingaling_" to form the database name
containing the "subscriptions" table.
used by: dingaling.conf.xml enum.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
<!--
THIS IS ONLY USED FOR DINGALING
bind_server_ip
Can be an ip address, a dns name, or "auto".
This determines an ip address available on this host to bind.
If you are separating RTP and SIP traffic, you will want to have
use different addresses where this variable appears.
Used by: dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
If you're going to load test FreeSWITCH please input real IP addresses
for external_rtp_ip and external_sip_ip
-->
<!-- external_rtp_ip
Can be an one of:
ip address: "12.34.56.78"
a stun server lookup: "stun:stun.server.com"
a DNS name: "host:host.server.com"
where fs.mydomain.com is a DNS A record-useful when fs is on
a dynamic IP address, and uses a dynamic DNS updater.
If unspecified, the bind_server_ip value is used.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an one of:
ip address: "12.34.56.78"
a stun server lookup: "stun:stun.server.com"
a DNS name: "host:host.server.com"
where fs.mydomain.com is a DNS A record-useful when fs is on
a dynamic IP address, and uses a dynamic DNS updater.
If unspecified, the bind_server_ip value is used.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
<!-- unroll-loops
Used to turn on sip loopback unrolling.
-->
<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
<!-- outbound_caller_id and outbound_caller_name
The caller ID telephone number we should use when calling out.
Used by: conference.conf.xml and user directory for default
outbound callerid name and number.
-->
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
<!-- various debug and defaults -->
<X-PRE-PROCESS cmd="set" data="call_debug=false"/>
<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
<X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
<X-PRE-PROCESS cmd="set" data="default_country=US"/>
<!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
if true, the destination number is not included -->
<X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>
<X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
<X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
<X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
<X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
<X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
<X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
<X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
<X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
<X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
<X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
<X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
<X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
<X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
<!--
Setting up your default sip provider is easy.
Below are some values that should work in most cases.
These are for conf/directory/default/example.com.xml
-->
<X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
<X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
<X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
<!-- true or false -->
<X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
<!--
SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
-->
<X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>
<!-- Internal SIP Profile -->
<X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
<X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/> <!--CUSTOMIZE-->
<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
<X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
<!-- External SIP Profile -->
<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
<X-PRE-PROCESS cmd="set" data="external_sip_port=5091"/> <!--CUSTOMIZE-->
<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
<X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>
</include>
3.7 /usr/local/freeswitch/conf/autoload_configs/modules.conf.xmlThis file controls which FreeSWITCH modules are loaded at startup time. No changes from standard original FreeSWITCH configuration, just be sure mod_lua is not commented out. We use Lua for database user management.
<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<load module="mod_logfile"/>
<!-- <load module="mod_syslog"/> -->
<!--<load module="mod_yaml"/>-->
<!-- Multi-Faceted -->
<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
<load module="mod_enum"/>
<!-- XML Interfaces -->
<!-- <load module="mod_xml_rpc"/> -->
<!-- <load module="mod_xml_curl"/> -->
<!-- <load module="mod_xml_cdr"/> -->
<!-- <load module="mod_xml_scgi"/> -->
<!-- Event Handlers -->
<load module="mod_cdr_csv"/>
<!-- <load module="mod_cdr_sqlite"/> -->
<!-- <load module="mod_event_multicast"/> -->
<load module="mod_event_socket"/>
<!-- <load module="mod_event_zmq"/> -->
<!-- <load module="mod_zeroconf"/> -->
<!-- <load module="mod_erlang_event"/> -->
<!-- <load module="mod_snmp"/> -->
<!-- Directory Interfaces -->
<!-- <load module="mod_ldap"/> -->
<!-- Endpoints -->
<!-- <load module="mod_dingaling"/> -->
<!-- <load module="mod_portaudio"/> -->
<!-- <load module="mod_alsa"/> -->
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_freetdm"/> -->
<!-- <load module="mod_openzap"/> -->
<!-- <load module="mod_unicall"/> -->
<!-- <load module="mod_skinny"/> -->
<!-- <load module="mod_khomp"/> -->
<!-- <load module="mod_rtmp"/> -->
<!-- Applications -->
<load module="mod_commands"/>
<load module="mod_conference"/>
<load module="mod_db"/>
<load module="mod_dptools"/>
<load module="mod_expr"/>
<load module="mod_fifo"/>
<load module="mod_hash"/>
<load module="mod_voicemail"/>
<!--<load module="mod_directory"/>-->
<!--<load module="mod_distributor"/>-->
<!--<load module="mod_lcr"/>-->
<load module="mod_esf"/>
<load module="mod_fsv"/>
<load module="mod_cluechoo"/>
<load module="mod_valet_parking"/>
<!--<load module="mod_fsk"/>-->
<!--<load module="mod_spy"/>-->
<!--<load module="mod_random"/>-->
<load module="mod_httapi"/>
<!-- SNOM Module -->
<!--<load module="mod_snom"/>-->
<!-- This one only works on Linux for now -->
<!--<load module="mod_ladspa"/>-->
<!-- Dialplan Interfaces -->
<!-- <load module="mod_dialplan_directory"/> -->
<load module="mod_dialplan_xml"/>
<load module="mod_dialplan_asterisk"/>
<!-- Codec Interfaces -->
<load module="mod_spandsp"/>
<load module="mod_g723_1"/>
<load module="mod_g729"/>
<load module="mod_amr"/>
<!--<load module="mod_ilbc"/>-->
<load module="mod_speex"/>
<load module="mod_h26x"/>
<load module="mod_vp8"/>
<!--<load module="mod_siren"/>-->
<!--<load module="mod_isac"/>-->
<!--<load module="mod_celt"/>-->
<!--<load module="mod_opus"/>-->
<!-- File Format Interfaces -->
<load module="mod_sndfile"/>
<load module="mod_native_file"/>
<!-- <load module="mod_shell_stream"/> -->
<!--For icecast/mp3 streams/files-->
<!--<load module="mod_shout"/>-->
<!--For local streams (play all the files in a directory)-->
<load module="mod_local_stream"/>
<load module="mod_tone_stream"/>
<!-- Timers -->
<!-- <load module="mod_timerfd"/> -->
<!-- <load module="mod_posix_timer"/> -->
<!-- Languages -->
<load module="mod_spidermonkey"/>
<!-- <load module="mod_perl"/> -->
<!-- <load module="mod_python"/> -->
<!-- <load module="mod_java"/> -->
<load module="mod_lua"/>
<!-- ASR /TTS -->
<!-- <load module="mod_flite"/> -->
<!-- <load module="mod_pocketsphinx"/> -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_tts_commandline"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Say -->
<load module="mod_say_en"/>
<!-- <load module="mod_say_ru"/> -->
<!-- <load module="mod_say_zh"/> -->
<!-- Third party modules -->
<!--<load module="mod_nibblebill"/>-->
<!--<load module="mod_callcenter"/>-->
</modules>
</configuration>
3.8 /usr/local/freeswitch/conf/autoload_configs/lua.conf.xmlThis is the configuration file for mod_lua. Start with the standard installed file, and then add the name of the script that will provide the XML data (in our case "xml_handler.lua") for which part of FreeSWITCH (in our case just for the "directory" part). You don't need to customize this file, but the lines added are marked as "CUSTOMIZE".
<configuration name="lua.conf" description="LUA Configuration">
<settings>
<!--
Specify local directories that will be searched for LUA modules
These entries will be pre-pended to the LUA_CPATH environment variable
-->
<!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
<!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->
<!--
Specify local directories that will be searched for LUA scripts
These entries will be pre-pended to the LUA_PATH environment variable
-->
<!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
<!-- <param name="script-directory" value="$${base_dir}/scripts/?.lua"/> -->
<!--<param name="xml-handler-script" value="/dp.lua"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a lua script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.lua"/>-->
<!--<param name="startup-script" value="startup_script_2.lua"/>-->
<param name="xml-handler-script" value="xml_handler.lua"/> <!--CUSTOMIZE-->
<param name="xml-handler-bindings" value="directory"/> <!--CUSTOMIZE-->
</settings>
</configuration>
3.9 /usr/local/freeswitch/conf/autoload_configs/acl.conf.xmlThis file configure the Access Control List (ACL) for FreeSWITCH. Starting from the original installed file, you insert one line that allows (without further checking for authorization) any call coming from the OpenSIPS IP address. You need to customize that IP address, follow "CUSTOMIZE".
<configuration name="acl.conf" description="Network Lists">
<network-lists>
<!--
These ACL's are automatically created on startup.
rfc1918.auto - RFC1918 Space
nat.auto - RFC1918 Excluding your local lan.
localnet.auto - ACL for your local lan.
loopback.auto - ACL for your local lan.
-->
<list name="lan" default="allow">
<node type="deny" cidr="192.168.42.0/24"/>
<node type="allow" cidr="192.168.42.42/32"/>
</list>
<!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
-->
<list name="domains" default="deny">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="$${domain}"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
<node type="allow" cidr="192.168.1.110/24"/> <!--CUSTOMIZE-->
</list>
</network-lists>
</configuration>
3.10 /etc/odbcinst.iniThis is the ODBC configuration file that tells ODBC where to find drivers. You probably don't need to customize it. Just check that is like the following. [MySQL] Description = MySQL driver Driver = /usr/lib64/odbc/libmyodbc.so Setup = /usr/lib64/odbc/libodbcmyS.so UsageCount = 1 FileUsage = 1 Threading = 0 3.11 /etc/odbc.iniPAY ATTENTION TO THIS FILE !!! If you insert comments in it, it will not work (brain damaged, but true). You need at least to change "mydbpassword". DON'T INSERT COMMENTS [opensips] Driver = /usr/lib64/odbc/libmyodbc.so SERVER = 127.0.0.1 PORT = 3306 DATABASE = opensips OPTION = 67108864 USER = opensips PASSWORD = mydbpassword 4. PROFIT !Restart it all /etc/init.d/apache2 restart /etc/init.d/opensips restart /usr/local/freeswitch/bin/freeswitch -stop /usr/local/freeswitch/bin/freeswitch -nc -nonat Go with your browser to http://webserveripaddress/cp Use OpenSIPS-CP to create a domain, then a couple users in that domain. Verify that all is working. They can call each other, calls goes to voicemail when needed, FreeSWITCH features can be accessed prepending * (star) to the corresponding FreeSWITCH extension (eg: *9664 for Music). Get rich! |