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TOPOH is generating new Call-ID for BYE
by Marrold 24 Mar '22

24 Mar '22

Hi, I have enabled the TOPOH module with mask_callid enabled, but it's generating a new Call-ID for a BYE which gets rejected with a 481 downstream. If I disable the TOPOH module it works as expected. Does anyone know where I should start looking? SIP-wise the initial BYE seems compliant. This is Kamailio 5.5.3 on Debian 11 Thanks Matthew

2 2

core parsing of SIP message failed
by Seyyed Soroosh Hosseinalipour 24 Mar '22

24 Mar '22

ITNOA Description I try to connect Kamailio and Freeswitch together, and initiate call from Zoiper5 soft phone through Kamailio My Kamailio config is below #!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_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - 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 # 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, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=yes 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:172.28.72.99:5060 advertise 172.28.72.99:5060 #listen=tcp:172.28.72.99:5060 /* 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_TLS enable_tls=no /* 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 = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### 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 # ----------------- 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", "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 #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!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 ####### 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(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); # handle requests within SIP dialogs # route(WITHINDLG); ### only initial requests (no To tag) # authentication # route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); #if (is_method("INVITE|SUBSCRIBE")) { # record_route(); #} # account only INVITEs #if (is_method("INVITE")) { # setflag(FLT_ACC); # do accounting #} # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests #route(PRESENCE); # handle registrations # route(REGISTRAR); #if ($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("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address 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", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($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) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$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 } As you can see I just one thing in request_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(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); The 172.28.69.52 is ip of my Freeswitch After I try to call from Zoiper5, I get below error in Kamailio log and Zoiper does not work. 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) Troubleshooting Reproduction * pull official Kamailio docker image kamailio/kamailio 5.4.6-buster * then run below command docker run --net=host --name kamailio \ -v /home/ssoroosh/kamailio:/etc/kamailio \ kamailio/kamailio:5.4.6-buster Debugging Data I do not have any debug data Log Messages Listening on udp: 172.28.72.99 [172.28.72.99]:5060 advertise 172.28.72.99:5060 Aliases: 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2919]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 13(19) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19 15(21) INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) 2(8) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) SIP Traffic My pcap is attached sip.zip<https://github.com/kamailio/kamailio/files/8326814/sip.zip> Possible Solutions I have not any solution yet. Additional Information * Kamailio Version - output of kamailio -v version: kamailio 5.4.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 * Operating System: Linux master 5.4.0-104-generic #118-Ubuntu SMP Wed Mar 2 19:02:41 UTC 2022 x86_64 GNU/Linux root@master:/home# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster How to resolve this? thanks

2 1

carrierroute with non-numeric characters
by Arek Łazarczyk 24 Mar '22

24 Mar '22

Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek Łazarczyk

2 3

bug in tm module?
by Daniel Paape 23 Mar '22

23 Mar '22

Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan

2 1

SIPS Errors on Kamailio
by Christopher Vincent 23 Mar '22

23 Mar '22

Hi, Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to SDES SRTP conversion. This worked as expected Attempted to duplicate the setup on RHEL but errors were seen. These errors were present on both RHEL 7 / RHEL 8. The errors seen were as below kamailio -c loading modules under config path: /usr/lib64/kamailio/modules/ 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the memory allocation functions 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(9165) ERROR: <core> [core/sr_module.c:590]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(9165) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to load module 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(9165) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized The kamailio config was exactly the same as on the CentOS systems and started as below /* Server ports: */ #!substdef "!SIP_PORT!5060!g" #!substdef "!SIPS_PORT!5061!g" /* Listen addresses */ #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g" #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g" /* Server connections: */ #!ifndef MAX_CONNECTIONS #!define MAX_CONNECTIONS 8192 #!endif ##!define WITH_DEBUG /* Transaction and branch 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 FLT_OUT 8 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_TLS enable_tls=1 listen=tls:<ipaddr>:5062 ####### Global Parameters ######### ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1, INFO=2, DBG=3 #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* display memory usage on exit */ mem_summary=15 /* join free memory fragments */ mem_join=1 /* proxy will fork and run in daemon mode */ /* one process will be created for each network interface the proxy listens to and for each protocol (TCP/UDP), multiplied with the value of 'children' parameter */ fork=yes children=8 listen=TCP_LOCAL_ADDR listen=UDP_LOCAL_ADDR /* 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 /* sip over websockets may not specify a content length header */ tcp_accept_no_cl=yes /* buffer size used for tcp reads, limits the maximum message size (SIP, HTTP) that can be received over tcp */ tcp_rd_buf_size=65536 /* max number of tcp connections */ tcp_max_connections=MAX_CONNECTIONS ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib64/kamailio/modules/" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.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 "dispatcher.so" loadmodule "cfgutils.so" loadmodule "textopsx.so" loadmodule "nathelper.so" loadmodule "uac.so" loadmodule "ipops.so" loadmodule "debugger.so" loadmodule "exec.so" loadmodule "avpops.so" loadmodule "sqlops.so" loadmodule "rtpengine.so" loadmodule "sl.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- usrloc params ----- # store contacts in memory only modparam("usrloc", "db_mode", 0) # hash size of 16,384 modparam("usrloc", "hash_size", 14) # removes contact if ws disconnects modparam("usrloc", "handle_lost_tcp", 1) modparam("tm|usrloc", "xavp_contact", "ulattrs") # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") # ----- 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) # ----- uac params ----- modparam("uac", "restore_mode", "none") # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # disable 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 direction 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) # ----- dispatcher params ----- modparam("dispatcher", "db_url", KAMAILIODBURL1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:<address>.com") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 3) # ----- pv params ----- modparam("pv", "shvset", "maintenance=i:0") modparam("pv", "shvset", "virtualIP1=i:0") modparam("pv", "shvset", "virtualIP2=i:0") # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is only being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") #modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") #modparam("rtpengine", "force_send_interface", SERVER_IP_ADDR) #modparam("rtpengine", "setid_default", -1) #modparam("rtpengine", "rtp_inst_pvar", "$avp(RTPENGINE)") #modparam("rtpengine", "rtpengine_retr", 5) #modparam("rtpengine", "queried_nodes_limit", 5) #modparam("rtpengine", "rtpengine_allow_op", 1) #modparam("rtpengine", "hash_table_size", MAX_CONNECTIONS) #modparam("rtpengine", "hash_table_tout", 7200) modparam("tls", "private_key", "<cert path>") modparam("tls", "certificate", "<cert path>") modparam("tls", "ca_list", "<cert path>") # modparam("tls", "ca_list", "<cert path>") modparam("tls", "tls_method", "TLSv1+") ####### Routing Logic ######## If load module lines for TLS are move to near the top of the config file, config will parse and non-SIPS calls will work loadmodule "sl.so" loadmodule "tls.so" But logs will show WARNING: <core> [main.c:2985]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) WARNING: <core> [main.c:2987]: main(): disabling tls... Presumably loading the module before configuring it just gives it default values so the latter config is ignored Any advice on the matter would be appreciated Thanks in advance, Chris

3 4

{param.in, name[, delimiter]} error when delimiter is comma
by 韩小仿 23 Mar '22

23 Mar '22

Hi, all Kamailio5.5.4 $var(s) = "a=1;b=2;c=3"; $var(s1) = $(var(s){param.value,b}); // work well $var(s) = "a=1|b=2|c=3"; $var(s1) = $(var(s){param.value,b,|}); // work well // but $var(s) = "a=1,b=2,c=3"; $var(s1) = $(var(s){param.value,b,,}); ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(var(s){param.value,b,,})] at (4) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 13-38: Can't get from cache: $(var(s){param.value,b,,}) how to fix this? Thanks! hanxiaofang

2 1

bug in tm module?
by dpaape＠frontier.com 22 Mar '22

22 Mar '22

Hello, I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset.So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires and authorization. After the 401 is sent back, the callback via the module event route is called. All good except that the module event route script fragment is never executed again after the first call.I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route. thanks for any help,Dan

1 0

SERIAL FORKING : call to a group number
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear all, My use case is : I defined a username = 112 (for emergency call) This username doesn't exist in subscribers table and unavailable in location table. The idea is when Sopftphone1 call the group = 112 then Kamailio will forward the call to a group using SERIAL FORKING. => Softphone 1 -> Sends INVITE to 112 -> Kamailio -> FORWARD TO SERIAL FORKING (4x softphone all registered and in location table). The q priority could be hybrid that means 2x softphone could have same priority. My issues : - the q priority forced for each softphone is not taking into account. - the first branch created when send INVITE to 112 is still available. My setup Kamailio is behind a NAT. Kamailio version : sipsecure@kamailio:/$ kamailio -version version: kamailio 5.5.4 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 Kamailio.cfg # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 5sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 10sec modparam("tm", "fr_inv_timer", 10000) # This is the name of an XAVP that the t_load_contacts() function uses to store contacts of the destination set and that t_next_contacts() function uses to restore those contacts. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") # modparam("tm|usrloc", "xavp_contact", "tm_contacts") # ----- 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) route[INVITE]{ route(THREAD_CHECK); route(AUTH); route(RECORD_ROUTE); setflag(FLT_ACC); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(FORK_SERIAL); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(FORK_PARALLEL); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); break; } exit; } route[FORK_SERIAL]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); $var(uri1)="sip:9919995@192.168.1.41"; $var(uri2)="sip:9919994@192.168.1.41"; $var(uri3)="sip:9919991@192.168.1.41"; $var(uri4)="sip:9919997@192.168.1.41"; lookup("location","$var(uri4)"); append_branch("$ru","0.2"); lookup("location","$var(uri1)"); append_branch("$ru","0.8"); lookup("location","$var(uri2)"); append_branch("$ru","0.5"); lookup("location","$var(uri3)"); append_branch("$ru","0.5"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL"); #!ifdef WITH_SIPSECURE_DEBUG xlog("MY_SEVERITY"," ROUTE_SERIAL : Relaying \n"); #!endif #Relay (aka Forward) the request t_relay(); break; } failure_route[SERIAL]{ xlog("MY_SEVERITY","At failure route - Trying next destination"); if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } t_on_failure("SERIAL"); t_relay(); } Any highlight will be welcome. Best Regards, -- Youssef

1 0

$ru vs. $fu as r/w pseudo variables
by Juha Heinanen 22 Mar '22

22 Mar '22

If I set $ru in config file and then print $ru to log, the new value is printed. If I do the same with $fu, the old value is printed. I haven't found in Pseudo-Variables Wiki any documentation about this. Does it exist somewhere else? -- Juha

3 9

MESSAGE METHOD - Send message to a softphone - $uac_req
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, I am expecting to send a MESSAGE METHOD to a softphone supporting MESSAGE. route[SEND_MESSAGE_METHOD]{ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; $uac_req(callid)=$(mb{s.md5}); $uac_req(hdrs)="Subject: Emergency Alert\r\n"; $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n"; $uac_req(body)="Emergency Message"; $uac_req(evroute)=1; uac_req_send(); } event_route[uac:reply] { xlog("MY_SEVERITY"," -- MESSAGE PROCESSING - received reply code is: $uac_req(evcode)\n");} My use case is to send message from KAMAILIO -> SOFTPHONE kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:192.168.1.29:5066 (physical ip address of the softphone) $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; The sip capture : 2022/03/18 02:20:58.947603 192.168.1.29:5066 -> 192.168.1.41:5066 MESSAGE sip:192.168.1.41:5066@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611 Max-Forwards: 70 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41> Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE User-Agent: MicroSIP/3.20.7 Content-Type: text/plain Content-Length: 3 ;-) 2022/03/18 02:20:58.948822 192.168.1.41:5066 -> 192.168.1.29:5066 SIP/2.0 200 Stored Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611;rport=5066 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41>;tag=cbfe6543b9079987bce3b5b9a0d1195a.b63b503b Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 Now, if I want to use his username@domain kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:9919997@192.168.1.41:5066 The message is sent to kamailio itself not to the softphone 2022/03/18 02:30:39.285927 192.168.1.41:5060 -> 192.168.1.41:5066 MESSAGE sip:9919997@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0 To: <sip:9919997@192.168.1.41:5066> From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Max-Forwards: 70 Content-Length: 17 User-Agent: kamailio (5.5.4 (x86_64/linux)) Subject: Emergency Alert Content-Type: text/plain Emergency Message 2022/03/18 02:30:39.287590 192.168.1.41:5066 -> 192.168.1.41:5060 SIP/2.0 200 Stored Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0;rport=5060;received=192.168.1.41 To: <sip:9919997@192.168.1.41:5066>;tag=cbfe6543b9079987bce3b5b9a0d1195a.76a88393 From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 - finaly, how to relay the message from one softphone to another softphone through kamailio proxy? Best Regards -- Youssef

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