#!KAMAILIO

#

# Kamailio (OpenSER) SIP Server v4.1 - 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/wiki/

# 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

#

#!define WITH_NAT

# *** To enable nat traversal execute:

#     - define WITH_NAT

#     - install RTPProxy: http://www.rtpproxy.org

#     - start RTPProxy:

#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722

#

# *** To enable PSTN gateway routing execute:

#     - define WITH_PSTN

#     - set the value of pstn.gw_ip

#     - check route[PSTN] for regexp routing condition

#

# *** To enable database aliases lookup execute:

#     - enable mysql

#     - define WITH_ALIASDB

#

# *** To enable speed dial lookup execute:

#     - enable mysql

#     - define WITH_SPEEDDIAL

#

# *** To enable multi-domain support execute:

#     - enable mysql

#     - define WITH_MULTIDOMAIN

#

# *** To enable TLS support execute:

#     - adjust CFGDIR/tls.cfg as needed

#     - define WITH_TLS

#

# *** To enable XMLRPC support execute:

#     - define WITH_XMLRPC

#     - adjust route[XMLRPC] for access policy

#

# *** To enable anti-flood detection execute:

#     - adjust pike and htable=>ipban settings as needed (default is

#       block if more than 16 requests in 2 seconds and ban for 300 seconds)

#     - define WITH_ANTIFLOOD

#

# *** To block 3XX redirect replies execute:

#     - define WITH_BLOCK3XX

#

# *** To enable VoiceMail routing execute:

#     - define WITH_VOICEMAIL

#     - set the value of voicemail.srv_ip

#     - adjust the value of voicemail.srv_port

#

# *** To enhance accounting execute:

#     - enable mysql

#     - define WITH_ACCDB

#     - add following columns to database

#!ifdef ACCDB_COMMENT

  ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';

  ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';

  ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';

  ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '                                                                             ';

  ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';

  ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';

  ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '                                                                             ';

#!endif

####### Include Local Config If Exists #########

import_file "kamailio-local.cfg"

####### 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.

#!ifndef DBURL

#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"

#!endif

#!endif

#!ifdef WITH_MULTIDOMAIN

# - the value for 'use_domain' parameters

#!define MULTIDOMAIN 1

#!else

#!define MULTIDOMAIN 0

#!endif

# - flags

#   FLT_ - per transaction (message) flags

#       FLB_ - per branch flags

#!define FLT_ACC 1

#!define FLT_ACCMISSED 2

#!define FLT_ACCFAILED 3

#!define FLT_NATS 5

#!define FLB_NATB 6

#!define FLB_NATSIPPING 7

####### Global Parameters #########

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR

#!ifdef WITH_DEBUG

debug=2

log_stderror=no

#!else

debug=3

log_stderror=no

#!endif

memdbg=1

memlog=1

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=7878

#!ifdef WITH_TLS

enable_tls=yes

#!endif

# life time of TCP connection when there is no traffic

# - a bit higher than registration expires to cope with UA behind NAT

tcp_connection_lifetime=3605

####### Custom Parameters #########

# These parameters can be modified runtime via RPC interface

# - see the documentation of 'cfg_rpc' module.

#

# Format: group.id = value 'desc' description

# Access: $sel(cfg_get.group.id) or @cfg_get.group.id

#

#!ifdef WITH_PSTN

# PSTN GW Routing

#

# - pstn.gw_ip: valid IP or hostname as string value, example:

# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"

#

# - by default is empty to avoid misrouting

pstn.gw_ip = "" desc "PSTN GW Address"

pstn.gw_port = "" desc "PSTN GW Port"

#!endif

#!ifdef WITH_VOICEMAIL

# VoiceMail Routing on offline, busy or no answer

#

# - by default Voicemail server IP is empty to avoid misrouting

voicemail.srv_ip = "" desc "VoiceMail IP Address"

voicemail.srv_port = "5060" desc "VoiceMail Port"

#!endif

####### Modules Section ########

# set paths to location of modules (to sources or installation folders)

#!ifdef WITH_SRCPATH

mpath="modules/"

#!else

mpath="/usr/local/lib/kamailio/modules/"

#!endif

#!ifdef WITH_MYSQL

loadmodule "db_mysql.so"

#!endif

loadmodule "mi_fifo.so"

loadmodule "kex.so"

loadmodule "corex.so"

loadmodule "tm.so"

loadmodule "tmx.so"

loadmodule "sl.so"

loadmodule "rr.so"

loadmodule "pv.so"

loadmodule "maxfwd.so"

loadmodule "usrloc.so"

loadmodule "registrar.so"

loadmodule "textops.so"

loadmodule "siputils.so"

loadmodule "xlog.so"

loadmodule "sanity.so"

loadmodule "ctl.so"

loadmodule "cfg_rpc.so"

loadmodule "mi_rpc.so"

loadmodule "acc.so"

#!ifdef WITH_AUTH

loadmodule "auth.so"

loadmodule "auth_db.so"

#!ifdef WITH_IPAUTH

loadmodule "permissions.so"

#!endif

#!endif

#!ifdef WITH_ALIASDB

loadmodule "alias_db.so"

#!endif

#!ifdef WITH_SPEEDDIAL

loadmodule "speeddial.so"

#!endif

#!ifdef WITH_MULTIDOMAIN

loadmodule "domain.so"

#!endif

#!ifdef WITH_PRESENCE

loadmodule "presence.so"

loadmodule "presence_xml.so"

#!endif

#!ifdef WITH_NAT

loadmodule "nathelper.so"

loadmodule "rtpproxy.so"

loadmodule "nat_traversal.so"

#!endif

#!ifdef WITH_TLS

loadmodule "tls.so"

#!endif

#!ifdef WITH_ANTIFLOOD

loadmodule "htable.so"

loadmodule "pike.so"

#!endif

#!ifdef WITH_XMLRPC

loadmodule "xmlrpc.so"

#!endif

#!ifdef WITH_DEBUG

loadmodule "debugger.so"

#!endif

# ----------------- 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)

# max value for expires of registrations

modparam("registrar", "max_expires", 3600)

# set it to 1 to enable GRUU

modparam("registrar", "gruu_enabled", 0)

# ----- acc params -----

/* what special events should be accounted ? */

modparam("acc", "early_media", 0)

modparam("acc", "report_ack", 0)

modparam("acc", "report_cancels", 0)

/* by default ww do not adjust the direct of the sequential requests.

   if you enable this parameter, be sure the enable "append_fromtag"

   in "rr" module */

modparam("acc", "detect_direction", 0)

/* account triggers (flags) */

modparam("acc", "log_flag", FLT_ACC)

modparam("acc", "log_missed_flag", FLT_ACCMISSED)

modparam("acc", "log_extra",

        "src_user=$fU;src_domain=$fd;src_ip=$si;"

        "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")

modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)

/* enhanced DB accounting */

#!ifdef WITH_ACCDB

modparam("acc", "db_flag", FLT_ACC)

modparam("acc", "db_missed_flag", FLT_ACCMISSED)

modparam("acc", "db_url", DBURL)

modparam("acc", "db_extra",

        "src_user=$fU;src_domain=$fd;src_ip=$si;"

        "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")

#!endif

# ----- usrloc params -----

/* enable DB persistency for location entries */

#!ifdef WITH_USRLOCDB

modparam("usrloc", "db_url", DBURL)

modparam("usrloc", "db_mode", 2)

modparam("usrloc", "use_domain", MULTIDOMAIN)

#!endif

# ----- auth_db params -----

#!ifdef WITH_AUTH

modparam("auth_db", "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

# ----- speeddial params -----

#!ifdef WITH_SPEEDDIAL

modparam("speeddial", "db_url", DBURL)

modparam("speeddial", "use_domain", MULTIDOMAIN)

#!endif

# ----- domain params -----

#!ifdef WITH_MULTIDOMAIN

modparam("domain", "db_url", DBURL)

# register callback to match myself condition with domains list

modparam("domain", "register_myself", 1)

#!endif

#!ifdef WITH_PRESENCE

# ----- presence params -----

modparam("presence", "db_url", DBURL)

# ----- presence_xml params -----

modparam("presence_xml", "db_url", DBURL)

modparam("presence_xml", "force_active", 1)

#!endif

#!ifdef WITH_NAT

# ----- rtpproxy params -----

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params -----

modparam("nathelper", "natping_interval", 30)

modparam("nathelper", "ping_nated_only", 1)

modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)

modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")

# params needed for NAT traversal in other modules

modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")

modparam("usrloc", "nat_bflag", FLB_NATB)

#!endif

#!ifdef WITH_TLS

# ----- tls params -----

modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")

#!endif

#!ifdef WITH_ANTIFLOOD

# ----- pike params -----

modparam("pike", "sampling_time_unit", 2)

modparam("pike", "reqs_density_per_unit", 16)

modparam("pike", "remove_latency", 4)

# ----- htable params -----

# ip ban htable with autoexpire after 5 minutes

modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")

#!endif

#!ifdef WITH_XMLRPC

# ----- xmlrpc params -----

modparam("xmlrpc", "route", "XMLRPC");

modparam("xmlrpc", "url_match", "^/RPC")

#!endif

#!ifdef WITH_DEBUG

# ----- debugger params -----

modparam("debugger", "cfgtrace", 7)

#!endif

####### Routing Logic ########

# Main SIP request routing logic

# - processing of any incoming SIP request starts with this route

# - note: this is the same as route { ... }

request_route {

        # per request initial checks

        route(REQINIT);

        # NAT detection

        route(NATDETECT);

        # CANCEL processing

        if (is_method("CANCEL"))

        {

                if (t_check_trans()) {

                        route(RELAY);

                }

                exit;

        }

        # handle requests within SIP dialogs

        route(WITHINDLG);

        ### only initial requests (no To tag)

        t_check_trans();

        # authentication

        # Below line commented by test-MATHUR as PC-2-PC calls are not getting                                                                              through

        #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"))

        {

                sl_send_reply("100", "Trying");

                setflag(FLT_ACC); # do accounting

        }

        # dispatch requests to foreign domains

        route(SIPOUT);

        ### requests for my local domains

        # handle presence related requests

        route(PRESENCE);

        # handle registrations

        route(REGISTRAR);

        if ($rU==$null)

        {

                # request with no Username in RURI

                sl_send_reply("484","Address Incomplete");

                exit;

        }

        # dispatch destinations to PSTN

        route(PSTN);

        # user location service

        route(LOCATION);

}

route[RELAY] {

        # enable additional event routes for forwarded requests

        # - serial forking, RTP relaying handling, a.s.o.

        if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {

                if(!t_is_set("branch_route")) {

                        t_on_branch("MANAGE_BRANCH");

                }

        }

        if (is_method("INVITE|SUBSCRIBE|UPDATE")) {

                if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");

        }

        if (is_method("INVITE")) {

                if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");

        }

        if (!t_relay()) {

                xlog("testM : Unable to relay !!!! :{ \n");

                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 $fu (IP:$s                                                                             i:$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()) {

                        route(DLGURI);

                        if (is_method("BYE")) {

                                #xlog("testM : Got BYE from some PSTN/UAC.\n");

                                setflag(FLT_ACC); # do accounting ...

                                #xlog("testM : Did accounting.\n");

                                setflag(FLT_ACCFAILED); # ... even if the transa                                                                             ction fails

                                #xlog("testM : Calling route{NATMANAGE} from WI                                                                             THINDLG.\n");

                                #t_newtran();

                                #t_reply("200", "OK");

                                #xlog("testM : Called route{NATMANAGE} for BYE                                                                              froom WITHINDLG.\n");

                                #exit;

                        }

                        else if ( is_method("ACK") ) {

                                # ACK is forwarded statelessy

                                route(NATMANAGE);

                        }

                        else if ( is_method("NOTIFY") ) {

                                # Add Record-Route for in-dialog NOTIFY as per R                                                                             FC 6665.

                                record_route();

                        }

                        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

                                        route(RELAY);

                                        exit;

                                } else {

                                        # ACK without matching transaction ... i                                                                             gnore 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_SPEEDDIAL

        # search for short dialing - 2-digit extension

        if($rU=~"^[0-9][0-9]$")

                if(sd_lookup("speed_dial"))

                        route(SIPOUT);

#!endif

#!ifdef WITH_ALIASDB

        # search in DB-based aliases

        if(alias_db_lookup("dbaliases"))

                route(SIPOUT);

#!endif

        $avp(oexten) = $rU;

        if (!lookup("location")) {

                $var(rc) = $rc;

                route(TOVOICEMAIL);

                t_newtran();

                switch ($var(rc)) {

                        case -1:

                        case -3:

                                send_reply("404", "Not Found");

                                exit;

                        case -2:

                                send_reply("405", "Method Not Allowed");

                                exit;

                }

        }

        # when routing via usrloc, log the missed calls also

        if (is_method("INVITE"))

        {

                setflag(FLT_ACCMISSED);

        }

        route(RELAY);

        exit;

}

# Presence server route

route[PRESENCE] {

        if(!is_method("PUBLISH|SUBSCRIBE"))

                return;

        if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") {

                route(TOVOICEMAIL);

                # returns here if no voicemail server is configured

                sl_send_reply("404", "No voicemail service");

                exit;

        }

#!ifdef WITH_PRESENCE

        if (!t_newtran())

        {

                sl_reply_error();

                exit;

        }

        if(is_method("PUBLISH"))

        {

                handle_publish();

                t_release();

        } else if(is_method("SUBSCRIBE")) {

                handle_subscribe();

                t_release();

        }

        exit;

#!endif

        # if presence enabled, this part will not be executed

        if (is_method("PUBLISH") || $rU==$null)

        {

                sl_send_reply("404", "Not here");

                exit;

        }

        return;

}

# Authentication route

route[AUTH] {

#!ifdef WITH_AUTH

#!ifdef WITH_IPAUTH

        if((!is_method("REGISTER")) && allow_source_address())

        {

                # source IP allowed

                return;

        }

#!endif

        if (is_method("REGISTER") || from_uri==myself)

        {

                # authenticate requests

                if (!auth_check("$fd", "subscriber", "1")) {

                        auth_challenge("$fd", "0");

                        exit;

                }

                # user authenticated - remove auth header

                if(!is_method("REGISTER|PUBLISH"))

                        consume_credentials();

        }

        # if caller is not local subscriber, then check if it calls

        # a local destination, otherwise deny, not an open relay here

        if (from_uri!=myself && uri!=myself)

        {

                sl_send_reply("403","Not relaying");

                exit;

        }

#!endif

        return;

}

# Caller NAT detection route

route[NATDETECT] {

#!ifdef WITH_NAT

        force_rport();

        if (nat_uac_test("19")) {

                if (is_method("REGISTER")) {

                        fix_nated_register();

                } else {

                        if(is_first_hop())

                                set_contact_alias();

                }

                setflag(FLT_NATS);

        }

#!endif

        return;

}

# RTPProxy control

route[NATMANAGE] {

#!ifdef WITH_NAT

        if (is_request()) {

                if(has_totag()) {

                        if(check_route_param("nat=yes")) {

                                setbflag(FLB_NATB);

                        }

                }

        }

#if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))

#{

#               rtpproxy_manage();

#               return;

#}

        rtpproxy_manage("z90");

        if (is_request()) {

                if (!has_totag()) {

                        if(t_is_branch_route()) {

                                if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)                                                                             )) {

                                add_rr_param(";nat=yes");

                                }

                        }

                }

        }

        if (is_reply()) {

                if(isbflagset(FLB_NATB)) {

                        if(is_first_hop())

                                set_contact_alias();

                }

        }

#!endif

        return;

}

# URI update for dialog requests

route[DLGURI] {

#!ifdef WITH_NAT

        if(!isdsturiset()) {

                handle_ruri_alias();

        }

#!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;

        }

        if (strempty($sel(cfg_get.pstn.gw_port))) {

                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);

        } else {

                $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"

                                        + $sel(cfg_get.pstn.gw_port);

        }

        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 bu                                                                             g in

                # xmlrpclib: it waits for EOF before interpreting the response).

                if ($hdr(User-Agent) =~ "xmlrpclib")

                        set_reply_close();

                set_reply_no_connect();

                dispatch_rpc();

                exit;

        }

        send_reply("403", "Forbidden");

        exit;

}

#!endif

# route to voicemail server

route[TOVOICEMAIL] {

#!ifdef WITH_VOICEMAIL

        if(!is_method("INVITE|SUBSCRIBE"))

                return;

        # check if VoiceMail server IP is defined

        if (strempty($sel(cfg_get.voicemail.srv_ip))) {

                xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");

                return;

        }

        if(is_method("INVITE")) {

                if($avp(oexten)==$null)

                        return;

                $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_i                                                                             p)

                                + ":" + $sel(cfg_get.voicemail.srv_port);

        } else {

                if($rU==$null)

                        return;

                $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)

                                + ":" + $sel(cfg_get.voicemail.srv_port);

        }

        route(RELAY);

        exit;

#!endif

        return;

}

# manage outgoing branches

branch_route[MANAGE_BRANCH] {

        xdbg("new branch [$T_branch_idx] to $ru\n");

        route(NATMANAGE);

}

# manage incoming replies

onreply_route[MANAGE_REPLY] {

        xdbg("incoming reply\n");

        if(status=~"[12][0-9][0-9]")

                route(NATMANAGE);

}

# manage failure routing cases

failure_route[MANAGE_FAILURE] {

        route(NATMANAGE);

        if (t_is_canceled()) {

                exit;

        }

#!ifdef WITH_BLOCK3XX

        # block call redirect based on 3xx replies.

        if (t_check_status("3[0-9][0-9]")) {

                t_reply("404","Not found");

                exit;

        }

#!endif

#!ifdef WITH_VOICEMAIL

        # serial forking

        # - route to voicemail on busy or no answer (timeout)

        if (t_check_status("486|408")) {

                $du = $null;

                route(TOVOICEMAIL);

                exit;

        }

#!endif

}

REGISTER sip:71.71.71.71 SIP/2.0

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW

To: "57778" <sip:test@71.71.71.71>

Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7

CSeq: 62147 REGISTER

User-Agent: pj_arubaslim-16/r2

Contact: "57778" <sip:test@192.168.1.4:58931;ob>

Expires: 900

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Content-Length:  0

P-hint: outbound

SIP/2.0 407 Proxy Authentication Required

CSeq: 62147 REGISTER

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE

From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW

Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7

To: "57778" <sip:test@71.71.71.71>;tag=170604141058

Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301389817101016806045824224"

Content-Length: 0

REGISTER sip:71.71.71.71 SIP/2.0

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW

To: "57778" <sip:test@71.71.71.71>

Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7

CSeq: 62148 REGISTER

User-Agent: pj_arubaslim-16/r2

Contact: "57778" <sip:test@192.168.1.4:58931;ob>

Expires: 900

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301389817101016806045824224", uri="sip:71.71.71.71", response="0ca765f266b6bfc8ea0ef00227272393"

Content-Length:  0

P-hint: outbound

SIP/2.0 200 OK

CSeq: 62148 REGISTER

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI

From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW

Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7

To: "57778" <sip:test@71.71.71.71>;tag=170604141059

Contact: "57778" <sip:test@192.168.1.4:58931;ob>;expires=600

Expires: 600

Content-Length: 0

SUBSCRIBE sip:test@71.71.71.71 SIP/2.0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

To: "57778" <sip:test@71.71.71.71>

Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

CSeq: 10987 SUBSCRIBE

Event: message-summary

Expires: 3600

Supported: replaces, 100rel, timer, norefersub

Accept: application/simple-message-summary

Allow-Events: presence, message-summary, refer

User-Agent: pj_arubaslim-16/r2

Content-Length:  0

P-hint: outbound

SIP/2.0 401 Unauthorised

CSeq: 10987 SUBSCRIBE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq

From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

To: "57778" <sip:test@71.71.71.71>;tag=170605141000

Content-Length: 0

WWW-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301390017101016806050024224"

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

SUBSCRIBE sip:test@71.71.71.71 SIP/2.0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

To: "57778" <sip:test@71.71.71.71>

Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

CSeq: 10988 SUBSCRIBE

Event: message-summary

Expires: 3600

Supported: replaces, 100rel, timer, norefersub

Accept: application/simple-message-summary

Allow-Events: presence, message-summary, refer

User-Agent: pj_arubaslim-16/r2

Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301390017101016806050024224", uri="sip:test@71.71.71.71", response="15ee3de757c18e084fca607438017093"

Content-Length:  0

P-hint: outbound

SIP/2.0 200 OK

CSeq: 10988 SUBSCRIBE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l

From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

To: "57778" <sip:test@71.71.71.71>;tag=170605141001

Expires: 3600

Content-Length: 0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

NOTIFY sip:test@71.71.71.71 SIP/2.0

CSeq: 100 NOTIFY

Via: SIP/2.0/UDP 71.71.71.71:5060

From: "57778" <sip:test@71.71.71.71>;tag=170605141001

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

Contact: <sip:71.71.71.71:5060;transport=udp>

Subscription-State: active

Event: message-summary

Content-Type: application/simple-message-summary

Content-Length: 65

Messages-Waiting: no

Message-Account: sip:test@71.71.71.71

SIP/2.0 404 Not here

CSeq: 100 NOTIFY

Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060

From: "57778" <sip:test@71.71.71.71>;tag=170605141001

Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB

To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC

Server: kamailio (4.1.3 (i386/linux))

Content-Length: 0

INVITE sip:919000000002@71.71.71.71 SIP/2.0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

To: <sip:919000000002@71.71.71.71>

Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

CSeq: 23382 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Session-Expires: 1800

Min-SE: 90

User-Agent: pj_arubaslim-16/r2

Content-Type: application/sdp

Content-Length:   356

P-hint: outbound

v=0

o=- 3612003234 3612003234 IN IP4 81.81.81.81

s=pjmedia

c=IN IP4 192.168.1.4

t=0 0

m=audio 43314 RTP/AVP 3 18 0 8 101

c=IN IP4 81.81.81.81

a=rtcp:43315

a=sendrecv

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=nortpproxy:yes

SIP/2.0 407 Proxy Authentication Required

CSeq: 23382 INVITE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Contact: <sip:71.71.71.71:5060;transport=udp>

Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301391017101016806051024224"

Content-Length: 0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

ACK sip:919000000002@71.71.71.71 SIP/2.0

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

CSeq: 23382 ACK

Content-Length: 0

INVITE sip:919000000002@71.71.71.71 SIP/2.0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

To: <sip:919000000002@71.71.71.71>

Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1>

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

CSeq: 23383 INVITE

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: replaces, 100rel, timer, norefersub

Session-Expires: 1800

Min-SE: 90

User-Agent: pj_arubaslim-16/r2

Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301391017101016806051024224", uri="sip:919000000002@71.71.71.71", response="4edf5e2efd1e8e03bdbe4a693fdfee4c"

Content-Type: application/sdp

Content-Length:   356

P-hint: outbound

v=0

o=- 3612003234 3612003234 IN IP4 81.81.81.81

s=pjmedia

c=IN IP4 192.168.1.4

t=0 0

m=audio 43314 RTP/AVP 3 18 0 8 101

c=IN IP4 81.81.81.81

a=rtcp:43315

a=sendrecv

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=nortpproxy:yes

SIP/2.0 183 Session Progress

CSeq: 23383 INVITE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Contact: <sip:71.71.71.71:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 252

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

v=0

o=SipSwitch 6248 7248 IN IP4 71.71.71.71

s=VoipSIP

i=Audio Session

c=IN IP4 71.71.71.71

t=0 0

m=audio 6248 RTP/AVP 18 101

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

SIP/2.0 180 Ringing

CSeq: 23383 INVITE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Contact: <sip:71.71.71.71:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 252

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

v=0

o=SipSwitch 6248 7248 IN IP4 71.71.71.71

s=VoipSIP

i=Audio Session

c=IN IP4 71.71.71.71

t=0 0

m=audio 6248 RTP/AVP 18 101

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

SIP/2.0 200 OK

CSeq: 23383 INVITE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0

Via: SIP/2.0/UDP 192.168.1.4:58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Contact: <sip:71.71.71.71:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 252

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

v=0

o=SipSwitch 6248 7248 IN IP4 71.71.71.71

s=VoipSIP

i=Audio Session

c=IN IP4 71.71.71.71

t=0 0

m=audio 6248 RTP/AVP 18 101

a=rtpmap:18 G729/8000/1

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

ACK sip:71.71.71.71:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

CSeq: 23383 ACK

Content-Length:  0

BYE sip:71.71.71.71:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU

Max-Forwards: 69

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

CSeq: 23384 BYE

User-Agent: pj_arubaslim-16/r2

Content-Length:  0

SIP/2.0 200 OK

CSeq: 23384 BYE

Via: SIP/2.0/UDP 81.81.81.81:7878;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0

Via: SIP/2.0/UDP 192.168.1.4:58931;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU

From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ

Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81

To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249

Contact: <sip:71.71.71.71:5060;transport=udp>

Content-Length: 0

Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>