Kamailio incoming call error

[JasurbekYuldashov]

kamailio.cfg

#!KAMAILIO

Kamailio SIP Server v5.5 - default configuration script

- web: https://www.kamailio.org

- git: https://github.com/kamailio/kamailio

Direct your questions about this file to: sr-users@lists.kamailio.org

Refer to the Core CookBook at https://www.kamailio.org/wiki/

for an explanation of possible statements, functions and parameters.

Note: the comments can be:

- lines starting with #, but not the pre-processor directives,

which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef,

#!subst, #!substdef, ...

- lines starting with //

- blocks enclosed in between /* */

Note: the config performs symmetric SIP signaling

- it sends the reply to the source address of the request

- remove the use of force_rport() for asymmetric SIP signaling

Several features can be enabled using '#!define WITH_FEATURE' directives:

*** To run in debug mode:

- define WITH_DEBUG

- debug level increased to 3, logs still sent to syslog

- debugger module loaded with cfgtrace endabled

*** To enable mysql:

- define WITH_MYSQL

*** To enable authentication execute:

- enable mysql

- define WITH_AUTH

- add users using 'kamctl' or 'kamcli'

*** 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

- if modified headers or body in config must be used by presence handling:

- define WITH_MSGREBUILD

*** To enable nat traversal execute:

- define WITH_NAT

- option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING

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

- start RTPProxy:

rtpproxy -l your_public_ip -s udp:localhost:7722

*** To use RTPEngine (instead of RTPProxy) for nat traversal execute:

- define WITH_RTPENGINE

- install RTPEngine: https://github.com/sipwise/rtpengine

- start RTPEngine:

rtpengine --listen-ng=127.0.0.1:2223 ...

*** 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 JSONRPC over HTTP(S) support execute:

- define WITH_JSONRPC

- adjust event_route[xhttp:request] 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 block 401 and 407 authentication replies execute:

- define WITH_BLOCK401407

*** 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 #########

#!substdef "!MY_IP_ADDR!public ip!g"
#!substdef "!MY_DOMAIN!test.sip.domain.com"
#!substdef "!MY_WS_PORT!5065!g"
#!substdef "!MY_WSS_PORT!443!g"
#!substdef "!MY_MSRP_PORT!9000!g"
#!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g"
#!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g"
#!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g"
#!substdef "!MSRP_MIN_EXPIRES!1800!g"
#!substdef "!MSRP_MAX_EXPIRES!3600!g"

#!define WITH_MYSQL

#!define WITH_AUTH

#!define WITH_USRLOCDB

#!define WITH_ACCDB
#!define WITH_WEBSOCKETS
#!define WITH_NAT
#!define WITH_NATSIPPING
#!define WITH_DEBUG

*** Value defines - IDs used later in config

#!ifdef WITH_DEBUG
#!define DBGLEVEL 2
#!else
#!define DBGLEVEL 2
#!endif

#!ifdef WITH_MYSQL
#loadmodule "db_mysql.so"

- database URL - used to connect to database server by modules such

as: auth_db, acc, usrloc, a.s.o.

#!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!endif

#!ifdef WITH_MULTIDOMAIN

- the value for 'use_domain' parameters

#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif

- flags

FLT_ - per transaction (message) flags

#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5

FLB_ - per branch flags

#!define FLB_NATB 6
#!define FLB_NATSIPPING 7

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

/* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */
debug=DBGLEVEL

/* set to 'yes' to print log messages to terminal or use '-E' cli option */
log_stderror=no

memdbg=5
memlog=5

log_facility=LOG_LOCAL0
log_prefix="{$mt $hdr(CSeq) $ci} "

/* number of SIP routing processes for each UDP socket

• • value inherited by tcp_children and sctp_children when not set explicitely */
    children=8

/* uncomment the next line to disable TCP (default on) */

disable_tcp=yes

/* number of SIP routing processes for all TCP/TLS sockets */

tcp_children=8

/* 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 - it can be set many times */

alias="sip.mydomain.com"

/* listen sockets - if none set, Kamailio binds to all local IP addresses

• • basic prototype (full prototype can be found in Wiki - Core Cookbook):

•  listen=[proto]:[localip]:[lport] advertise [publicip]:[pport]
  

• • it can be set many times to add more sockets to listen to */

listen=udp:10.0.0.10:5060

/* 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

/* upper limit for TCP connections (it includes the TLS connections) */
tcp_max_connections=2048

#!ifdef WITH_JSONRPC
tcp_accept_no_cl=yes
#!endif

#!ifdef WITH_WEBSOCKETS
tcp_accept_no_cl=yes
tcp_rd_buf_size=65535
enable_tls=yes
#!endif

#!ifdef WITH_TLS

/* upper limit for TLS connections */
tls_max_connections=2048
#!endif

/* set it to yes to enable sctp and load sctp.so module */
enable_sctp=no

####### 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 = "sip.domain.com"
    voicemail.srv_port = "5060"
    #!endif

listen=MY_IP_ADDR
#!ifdef WITH_WEBSOCKETS
listen=MY_WS_ADDR
#!ifdef WITH_TLS
listen=MY_WSS_ADDR
#!endif
#!endif
#!ifdef WITH_MSRP
listen=MY_MSRP_ADDR
#!endif

tcp_connection_lifetime=3604
tcp_accept_no_cl=yes
tcp_rd_buf_size=16384

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

/* set paths to location of modules */

mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/"

#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif

#!ifdef WITH_JSONRPC
loadmodule "xhttp.so"
#!endif
loadmodule "jsonrpcs.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 "textopsx.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.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"
#!ifdef WITH_RTPENGINE
loadmodule "rtpengine.so"
#!else
loadmodule "rtpproxy.so"
#!endif
#!endif

#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif

#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif

#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif

#!ifdef WITH_WEBSOCKETS
loadmodule "websocket.so"
loadmodule "xhttp.so"
#!endif

----------------- setting module-specific parameters ---------------

----- jsonrpcs params -----

modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */

modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo")

/* set the path to RPC unix socket control file */

modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock")

#!ifdef WITH_JSONRPC
modparam("jsonrpcs", "transport", 7)
#!endif

----- ctl params -----

/* set the path to RPC unix socket control file */

modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl")

----- sanity params -----

modparam("sanity", "autodrop", 0)

----- 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 -----

set next param to 1 to add value to ;lr param (helps with some UAs)

modparam("rr", "enable_full_lr", 0)

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)
/ set it to 0 to disable Path handling /
modparam("registrar", "use_path", 1)
/ save Path even if not listed in Supported header */
modparam("registrar", "path_mode", 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 -----

modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "timer_procs", 1)
modparam("usrloc", "use_domain", MULTIDOMAIN)
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
#!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", "load_backends", 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
#!ifdef WITH_RTPENGINE

----- rtpengine params -----

modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222")
#!else

----- rtpproxy params -----

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
#!endif

----- 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", "/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_DEBUG

----- debugger params -----

modparam("debugger", "cfgtrace", 1)
modparam("debugger", "log_level_name", "exec")
#!endif

#!ifdef WITH_WEBSOCKETS
modparam("websocket", "keepalive_mechanism", 0)
modparam("websocket", "keepalive_timeout", 100)
#!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 retransmissions

  if (!is_method("ACK")) {
  if(t_precheck_trans()) {
  t_check_trans();
  exit;
  }
  t_check_trans();
  }

  handle requests within SIP dialogs

  route(WITHINDLG);

  only initial requests (no To tag)

  # authentication

  route(AUTH);

  record routing for dialog forming requests (in case they are routed)

  - remove preloaded route headers

  remove_hf("Route");
  if (is_method("INVITE|SUBSCRIBE")) {
  record_route();
  }

  account only INVITEs

  if (is_method("INVITE")) {
  setflag(FLT_ACC); # do accounting
  }

  dispatch requests to foreign domains

  route(SIPOUT);

  requests for my local domains

  handle presence related requests

  route(PRESENCE);

  handle registrations

  route(REGISTRAR);

  if ($rU==$null) {
  # request with no Username in RURI
  sl_send_reply("484","Address Incomplete");
  exit;
  }

  dispatch destinations to PSTN

  route(PSTN);

  user location service

  route(LOCATION);

}

Wrapper for relaying requests

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()) {
	sl_reply_error();
}
exit;

}

Per SIP request initial checks

route[REQINIT] {
# no connect for sending replies
set_reply_no_connect();
# enforce symmetric signaling
# - send back replies to the source address of request
force_rport();

#!ifdef WITH_ANTIFLOOD
# flood detection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself) {
if($sht(ipban=>$si)!=$null) {
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
exit;
}
if (!pike_check_req()) {
xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif
if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") {
# silent drop for scanners - uncomment next line if want to reply
# sl_send_reply("200", "OK");
exit;
}

if (!mf_process_maxfwd_header("10")) {
	sl_send_reply("483","Too Many Hops");
	exit;
}

if(is_method("OPTIONS") && uri==myself && $rU==$null) {
	sl_send_reply("200","Keepalive");
	exit;
}

if(!sanity_check("17895", "7")) {
	xlog("Malformed SIP request from $si:$sp\n");
	exit;
}

}

Handle requests within SIP dialogs

route[WITHINDLG] {
if (!has_totag()) return;

# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
	route(DLGURI);
	if (is_method("BYE")) {
		setflag(FLT_ACC); # do accounting ...
		setflag(FLT_ACCFAILED); # ... even if the transaction fails
	} else if ( is_method("ACK") ) {
		# ACK is forwarded statelessly
		route(NATMANAGE);
	} else if ( is_method("NOTIFY") ) {
		# Add Record-Route for in-dialog NOTIFY as per RFC 6665.
		record_route();
	}
	route(RELAY);
	exit;
}

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 ... ignore and discard
		exit;
	}
}
sl_send_reply("404","Not here");
exit;

}

Handle SIP registrations

route[REGISTRAR] {
if (!is_method("REGISTER")) return;

if(isflagset(FLT_NATS)) {
	setbflag(FLB_NATB);

#!ifdef WITH_NATSIPPING
# do SIP NAT pinging
setbflag(FLB_NATSIPPING);
#!endif
}
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 processing

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
#!ifdef WITH_MSGREBUILD
# apply changes in case the request headers or body were modified
msg_apply_changes();
#!endif
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;

}

IP authorization and user authentication

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;
}

#!else

# authentication not enabled - do not relay at all to foreign networks
if(uri!=myself) {
	sl_send_reply("403","Not relaying");
	exit;
}

#!endif
return;
}

Caller NAT detection

route[NATDETECT] {
#!ifdef WITH_NAT
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 and signaling updates for NAT traversal

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))) return;

#!ifdef WITH_RTPENGINE
if(nat_uac_test("8")) {
rtpengine_manage("SIP-source-address replace-origin replace-session-connection");
} else {
rtpengine_manage("replace-origin replace-session-connection");
}
#!else
if(nat_uac_test("8")) {
rtpproxy_manage("co");
} else {
rtpproxy_manage("cor");
}
#!endif

if (is_request()) {
	if (!has_totag()) {
		if(t_is_branch_route()) {
			add_rr_param(";nat=yes");
		}
	}
}
if (is_reply()) {
	if(isbflagset(FLB_NATB)) {
		if(is_first_hop())
			set_contact_alias();
	}
}

if(isbflagset(FLB_NATB)) {
	# no connect message in a dialog involving NAT traversal
	if (is_request()) {
		if(has_totag()) {
			set_forward_no_connect();
		}
	}
}

#!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) return;

append_hf("P-Hint: outbound\r\n");
route(RELAY);
exit;

}

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 routing 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;
}

# normalize target number for pstn gateway
# - convert leading 00 to +
if (starts_with("$rU", "00")) {
	strip(2);
	prefix("+");
}

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;

}

JSONRPC over HTTP(S) routing

#!ifdef WITH_JSONRPC
event_route[xhttp:request] {
set_reply_close();
set_reply_no_connect();
if(src_ip!=127.0.0.1) {
xhttp_reply("403", "Forbidden", "text/html",
"Not allowed from $si");
exit;
}
if ($hu =~ "^/RPC") {
jsonrpc_dispatch();
exit;
}

xhttp_reply("200", "OK", "text/html",
			"<html><body>Wrong URL $hu</body></html>");
exit;

}
#!endif

#!ifdef WITH_WEBSOCKETS
event_route[xhttp:request] {
set_reply_close();
set_reply_no_connect();
if ($hdr(Upgrade)="websocket"
&& $hdr(Connection)="Upgrade"
&& $rm=~"GET") {

	# Validate Host - make sure the client is using the correct
	# alias for WebSockets
	if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) {
		xlog("L_WARN", "Bad host $hdr(Host)\n");
		xhttp_reply("403", "Forbidden", "", "");
		exit;
	}
	if (ws_handle_handshake()) {
		# Optional... cache some information about the
		# successful connection
		exit;
	}
}
xhttp_reply("404", "Not Found", "", "");

}
#!endif

Routing to voicemail server

route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE")) return;

# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
	xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n");
	return;
}
if(is_method("INVITE")) {
	if($avp(oexten)==$null) return;

	$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
			+ ":" + $sel(cfg_get.voicemail.srv_port);
} else {
	if($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

reply_route {
if(!sanity_check("17604", "6")) {
xlog("Malformed SIP response from $si:$sp\n");
drop;
}
}

Manage incoming replies in transaction context

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_BLOCK401407
# block call redirect based on 401, 407 replies.
if (t_check_status("401|407")) {
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
}

• Operating System:

Ubuntu 22.04

SIP/2.0 500 Service Unavailable
Via: SIP/2.0/WS 5h7t17u4ne29.invalid;branch=z9hG4bK1035487;rport=45366;received=178.218.200.29
To: sip:107@sip.domain.com;tag=ae7c22ff3e5ea89a75cf77ccdbc930c9-148e0000
From: sip:107@sip.domain.com;tag=i6hcmqvl9p
Call-ID: of4dmvvehi37ai051i83
CSeq: 1146 INVITE
Server: kamailio (5.5.4 (x86_64/linux))
Content-Length: 0

!g

[linuxmaniac]

This tracker is used for reporting bugs in the C code. For asking questions about using Kamailio and its configuration file, use sr-users@lists.kamailio.org mailing list.