[root@kamailio log]# ngrep -d any -qt -W byline "sip" port 5060 interface: any filter: ( port 5060 ) and (ip || ip6) match: sip

U 2019/02/11 10:04:27.082271 214.22.198.51:5060 -> 38.149.172.163:5060 #1 INVITE sip:100@38.149.172.163:5060 SIP/2.0. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. Max-Forwards: 70. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Contact: sip:123456@214.22.198.51:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 14.6.2. Date: Mon, 11 Feb 2019 08:04:27 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 312. .. v=0. o=root 2139827131 2139827131 IN IP4 214.22.198.51. s=Asterisk PBX 14.6.2. c=IN IP4 214.22.198.51. t=0 0. m=audio 19844 RTP/AVP 18 8 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=maxptime:150. a=sendrecv. ................

U 2019/02/11 10:04:27.082792 38.149.172.163:5060 -> 214.22.198.51:5060 #2 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. Server: kamailio (5.2.1 (x86_64/linux)). Content-Length: 0. .. ................

U 2019/02/11 10:04:27.082887 38.149.172.163:5060 -> 38.149.172.164:5060 #3 INVITE sip:100@38.149.172.163:5060 SIP/2.0. Via: SIP/2.0/UDP 38.149.172.163;branch=z9hG4bK0e77.45dda2182eeb3d9621f578f5920132b5.0. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. Max-Forwards: 69. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Contact: sip:123456@214.22.198.51:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 14.6.2. Date: Mon, 11 Feb 2019 08:04:27 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 312. .. v=0. o=root 2139827131 2139827131 IN IP4 214.22.198.51. s=Asterisk PBX 14.6.2. c=IN IP4 214.22.198.51. t=0 0. m=audio 19844 RTP/AVP 18 8 0 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=maxptime:150. a=sendrecv. ................

U 2019/02/11 10:04:27.084992 38.149.172.164:5060 -> 38.149.172.163:5060 #4 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 38.149.172.163;branch=z9hG4bK0e77.45dda2182eeb3d9621f578f5920132b5.0;received=38.149.172.163. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:100@38.149.172.164:5060. Content-Length: 0. .. ................

U 2019/02/11 10:04:27.085173 38.149.172.163:5060 -> 214.22.198.51:5060 #5 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:100@38.149.172.164:5060. Content-Length: 0. .. ................

U 2019/02/11 10:04:27.130931 38.149.172.164:5060 -> 38.149.172.163:5060 #6 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 38.149.172.163;branch=z9hG4bK0e77.45dda2182eeb3d9621f578f5920132b5.0;received=38.149.172.163. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060;tag=as4281a87a. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:100@38.149.172.164:5060. Content-Length: 0. .. ................

U 2019/02/11 10:04:27.131137 38.149.172.163:5060 -> 214.22.198.51:5060 #7 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060;tag=as4281a87a. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:100@38.149.172.164:5060. Content-Length: 0. .. ................

U 2019/02/11 10:04:30.209844 214.22.198.51:5060 -> 38.149.172.163:5060 #8 CANCEL sip:100@38.149.172.163:5060 SIP/2.0. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. Max-Forwards: 70. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 CANCEL. User-Agent: Asterisk PBX 14.6.2. Content-Length: 0. .. ................

U 2019/02/11 10:04:30.710386 214.22.198.51:5060 -> 38.149.172.163:5060 #9 CANCEL sip:100@38.149.172.163:5060 SIP/2.0. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. Max-Forwards: 70. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 CANCEL. User-Agent: Asterisk PBX 14.6.2. Content-Length: 0. .. ................

U 2019/02/11 10:04:31.710612 214.22.198.51:5060 -> 38.149.172.163:5060 #10 CANCEL sip:100@38.149.172.163:5060 SIP/2.0. Via: SIP/2.0/UDP 214.22.198.51:5060;branch=z9hG4bK36045fbd. Max-Forwards: 70. From: "testaccount" sip:123456@214.22.198.51;tag=as5cd2705d. To: sip:100@38.149.172.163:5060. Call-ID: 049fb3dc6ef4901060de258b794fc916@214.22.198.51:5060. CSeq: 102 CANCEL. User-Agent: Asterisk PBX 14.6.2. Content-Length: 0. .. ................

This is my config. #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ACCDB # # Kamailio (OpenSER) SIP Server v5.2 - 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 /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 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

############################################################################## 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=4 log_stderror=yes #!else debug=3 log_stderror=yes #!endif

memdbg=5 memlog=5

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

/* number of SIP routing processes */ children=8

/* 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

#!ifdef WITH_TLS enable_tls=no #!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 */ # mpath="/usr/lib64/kamailio/modules/"

#!ifdef WITH_MYSQL loadmodule "db_mysql.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 "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "dispatcher.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" #!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 ---------------

# ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")

# ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")

# ----- 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)

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

modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio")

# ----- dispatcher params ----- #modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "ds_ping_interval", 30)

########################################################## 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); t_check_trans(); if ( method=="INVITE" ) { ds_select_dst("1","4"); sl_send_reply("100","Trying"); forward(); exit(); }

# 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] { #!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|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 message 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 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 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 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;

if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); }

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(); } } #!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; }

# XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif

# 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; }

# Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; }

failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

# 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_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 } route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); }

Closed #1841.

Thank you ! It work for me.