- sr-users - kamailio.org

get ip status from dispatcher
by ale Gural 10 Jun '25

10 Jun '25

Hello, There is a way to get the status of specific ip of dispatcher in kamailio.cfg? not using kamcmd dispatcher.list, thanks!

3 2

How to detect originating and terminating?
by James Morrison 09 Jun '25

09 Jun '25

I'm trying to find a way to differentiate between originating and terminating legs. Currently in kex module, in core_stats.c I'm adding new counters for more response codes, I want to have two sets of counters, one for originating and one for terminating. In ims_getters.c file, I found a function called cscf_has_originating, which by description "Finds if the message contains the orig parameter in the first Route header" I'm calling it like the following: static int km_cb_rpl_stats_by_method(struct sip_msg *msg, unsigned int flags, void *param) { int originating = cscf_has_originating(msg); .... However it always returns CSCF_RETURN_FALSE (or -1). I added a log message and increased log level for core module: DEBUG: <core> [ims_getters.c:561]: cscf_has_originating(): I_originating: Header Route not found ERROR: kex [core_stats.c:424]: km_cb_rpl_stats_by_method(): In kex module received rpls, the return value of cscf_has_originating is: -1 That's because cscf_has_originating function is calling parse_headers function, trying to get msg->route but that route property is not set correctly in the previous step: if(parse_headers(msg, HDR_ROUTE_F, 0) < 0) { LM_DBG("I_originating: error parsing headers\n"); return CSCF_RETURN_FALSE; } so it returns -1 after checking msg->route value. Do you know any alternative solution? or do you know why msg->route value is 0 or NULL?

3 6

siptrace module not respecting send_sock after kamailio 5.6.6
by Daniel Donoghue 09 Jun '25

09 Jun '25

Hello, I have been using the siptrace module on a multihomed kamailio setup for some time now, with kamailio 5.5.7 and 5.6.6, without incident. I am listening to my external interface using something like: listen=tls:external_ip:5061 listen=tcp:external_ip:5060 Listen=tcp:internal_ip:5000 name "homer" And for siptrace... modparam("siptrace", "duplicate_uri", "sip:homer_ip:9060;transport=tcp") modparam("siptrace", "send_sock_name", "homer") I have also used send_sock_addr and force_send... This works as expected. I recently deployed this same setup using kamailio 6.0.1 and discovered that the send_sock is not being respected, even though the logs say it has been selected. Instead, tcpdump and careful examination of all the file descriptors for the kamailio processes reveal it is sending with the source set to my external_ip instead of the internal_ip. I have verified this on the same server, ruling out os level issues routing issues and have tried manually connecting using nc with the internal ip address for the source, which works. I tried downgrading, and found that the same is true for kamailio 5.7.6, 5.8.6 and 6.0.1 This issue only appears to affect siptrace as other sockets appear to use the correct source ip to, for example, connect to a database. I had a look at the change logs and documentation and cant see anything obvious that I need to change from a configuration perspective. Could anybody help? I would prefer to be running 6.0+ in my new production environment but wont be able to if this cannot be solved quickly. Many thanks for your help. With every blessing, -- Daniel Donoghue

2 2

TCP pings from dispatcher
by Alex Balashov 07 Jun '25

07 Jun '25

Does anyone know offhand if dispatcher has access to existing TCP connections from the core for its OPTIONS pings to TCP dispatcher gateways, or whether it generates its own proprietary TCP connections for this specific purpose? -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com, https://www.csrpswitch.com Tel: +1-706-510-6800

2 1

Kamailio Deployment in AWS
by gcampos＠momenta.network 06 Jun '25

06 Jun '25

Hi community, I'm not sure if anyone has ever tried that before, but I'm trying to run Kamailio in AWS with RTPengine. I successfully installed both Kamailio and RTPengine on my EC2 instance and I'm able to make my devices register to it. However, when I make a call from one of my devices, the call is muted. As I analyze my kamailio.cfg file, I see that the "route[NATMANAGE]" is likely the cause of the mute call. I think that none of the conditions below are met, therefore, the flags are not set. Because of that, the last condition is met, thus triggering the "return" and exiting from the route[NATMANAGE] prematurely without even reaching the section where I define the RTPengine modifications. Not only that, but if I remove/comment out the "return", then the RTPengine is triggered, but there call doesn't connect. I wonder why this is happening and if there's any known workaround. # RTP relaying management and signaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO", "NATMANAGE path was followed\n"); #!ifdef WITH_NAT xlog("L_INFO", "WITH_NAT Triggered\n"); if (is_request()) { xlog("L_INFO", "is_request path was followed\n"); if (has_totag()) { xlog("L_INFO", "has_totag path was followed\n"); if (check_route_param("nat=yes")) { xlog("L_INFO", "nat=yes param detected, setting FLB_NATB\n"); setbflag(FLB_NATB); } } if (nat_uac_test("3")) { xlog("L_INFO", "nat_uac_test(3) matched, setting FLT_NATS + FLB_NATB\n"); setflag(FLT_NATS); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xlog("L_INFO", "No NAT flags set, skipping NAT handling\n"); return; } #!ifdef WITH_RTPENGINE xlog("L_INFO", "WITH_RTPENGINE Path followed\n"); if (has_body("application/sdp")) { if (is_method("INVITE")) { xlog("L_INFO", "IS_METHOD_INVITE Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } else if (is_reply()) { xlog("L_INFO", "IS_Reply Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } } #!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; }

3 4

Kamailio Asterisk Realtime integration
by cedric＠mrtn.ovh 06 Jun '25

06 Jun '25

Hello, I'm trying to use Kamailio as proxy for Asterisk. I follow this tutorial https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I maked some updates to replace old libs and old elements related to db table and fields to use the pjsip table (ps_auths) and fields (username, password). I configured Asterisk Realtime by following the Asterisk documentation (extconfig.conf, sorcery.conf, rec_odbc.conf etc...). I'm lock at this step : The registering is handle correctly by Kamailio and forwarded to Asterisk, but asterisk respond with a "401" error and want to get a digest auth... Bellow, my kamailio config file: ``` #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DEBUG # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### 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. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asterisk_password@localhost/asterisk" #!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 ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.1.2:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!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 #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.2" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.1.2" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.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 "jsonrpcs.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 #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs fifo params ----- modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- 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", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- 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 # ----- speedial 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:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!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); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # 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); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # 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 #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!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); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # 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; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "ps_auths", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif ``` Also, I'm not sure about the configuration to add into asterisk, like a kamailio registration block or endpoint block, and... probably others... Thanks to help. Flywall.

2 1

Disable IPSec in Kamailio PCSCF
by Navjyoty Singh 05 Jun '25

05 Jun '25

Hi, I'm working on a custom IMS deployment using the pcscf module in Kamailio. For our setup, we do not want Kamailio to handle IPsec or generate Security-Server headers in 401/407 responses. Is there a clean way to disable IPsec functionality entirely in the pcscf module? Specifically, I would like to: Prevent the module from requiring IPsec setup during UE registration Avoid any modification of SIP messages related to IPsec (e.g., removing Security-Verify, Security-Client, or Security-Server headers) Use only TLS or UDP without IPsec involvement Has anyone achieved this via configuration or by disabling specific features? Any guidance or examples would be greatly appreciated. *Regards,Navjyoty*

2 1

Replace ip address of source
by disa_369＠msn.com 05 Jun '25

05 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR IVR number is 3000. Avaya Prefix for routing to Kamailio is 1234. When we dial number 12343000 the call goes to Video system with Header "TO" 12343000@ip_of_PBXAvaya and call drops. How to replace ip-address of Avaya to ip-address of Video_System? What to add in kamailio.cfg? ip video_system - *.*.60.3 ip avaya *.*.20.10 My route config ####### 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 } if ($rU==$null) { sl_send_reply("484", "Address Incomplete"); exit; } $du = "sip:*.*.60.3:5060"; route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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|pplsip") { # 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", "<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); return; } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } return; } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } return; } # 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 return;

5 5

Why won't this work
by Fernando Lopes 05 Jun '25

05 Jun '25

Kamailio isn’t working on my Azure VM, even though it’s set up to use port 5060. The network rules on Azure have both inbound and outbound traffic open for all protocols. I also tried using advertised_address, but that didn’t help either. It looks like no traffic is even hitting Kamailio — none of the routing blocks are triggered. Kamailio.cfg alias=<PUBLIC-IP> listen=udp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 listen=tcp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 Sip Message: 2025/06/04 15:50:37.870679 64.16.250.10:5060 -> 10.0.0.4:5060 INVITE sip:+351200600020@<PUBLIC-IP>:5060 SIP/2.0 Record-Route: <sip:64.16.250.10;r2=on;lr;ftag=DpHvv09240egQ> Record-Route: <sip:10.255.0.2;r2=on;lr;ftag=DpHvv09240egQ> Via: SIP/2.0/UDP 64.16.250.10;branch=z9hG4bKd59a.fb67bf395ea20488a7a5ebf76b9b5685.0 v:SIP/2.0/UDP 10.239.196.24:6000;received=10.239.196.24;rport=6000;branch=z9hG4bKy9ZvBZcgretvr Max-Forwards:48 f:"+351908020050"<sip:+351908020050@sip.telnyx.com>;tag=DpHvv09240egQ t:<sip:+351200600020@<PUBLIC-IP>:5060> i:122d71bb-4f3d-473f-8853-9137023f1f14 CSeq:99972326 INVITE m:<sip:mod_sofia@10.239.196.24:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:202 P-Asserted-Identity:"+351908020050"<sip:+351908020050@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1749019739 1749019740 IN IP4 50.114.147.4 s=Telnyx c=IN IP4 50.114.147.4 t=0 0 m=audio 32498 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:32499 IN IP4 50.114.147.4 a=ptime:20

6 15

Kamailio Deployment in AWS
by gcampos＠momenta.network 04 Jun '25

04 Jun '25

Hi community, I'm not sure if anyone has ever tried that before, but I'm trying to run Kamailio in AWS with RTPengine. I successfully installed both Kamailio and RTPengine on my EC2 instance and I'm able to make my devices register to it. However, when I make a call from one of my devices, the call is muted. As I analyze my kamailio.cfg file, I see that the "route[NATMANAGE]" is likely the cause of the mute call. I think that none of the conditions below are met, therefore, the flags are not set. Because of that, the last condition is met, thus triggering the "return" and exiting from the route[NATMANAGE] prematurely without even reaching the section where I define the RTPengine modifications. Not only that, but if I remove/comment out the "return", then the RTPengine is triggered, but there call doesn't connect. I wonder why this is happening and if there's any known workaround. # RTP relaying management and signaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO", "NATMANAGE path was followed\n"); #!ifdef WITH_NAT xlog("L_INFO", "WITH_NAT Triggered\n"); if (is_request()) { xlog("L_INFO", "is_request path was followed\n"); if (has_totag()) { xlog("L_INFO", "has_totag path was followed\n"); if (check_route_param("nat=yes")) { xlog("L_INFO", "nat=yes param detected, setting FLB_NATB\n"); setbflag(FLB_NATB); } } if (nat_uac_test("3")) { xlog("L_INFO", "nat_uac_test(3) matched, setting FLT_NATS + FLB_NATB\n"); setflag(FLT_NATS); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xlog("L_INFO", "No NAT flags set, skipping NAT handling\n"); return; } #!ifdef WITH_RTPENGINE xlog("L_INFO", "WITH_RTPENGINE Path followed\n"); if (has_body("application/sdp")) { if (is_method("INVITE")) { xlog("L_INFO", "IS_METHOD_INVITE Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } else if (is_reply()) { xlog("L_INFO", "IS_Reply Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } } #!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; }

1 0

Will kamailio handle sip messages if Request contains public Ip
by Fernando Lopes 04 Jun '25

04 Jun '25

I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. The SIP message received looks like this: INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0 Will Kamailio handle this SIP message, or will it ignore it? Right now, Kamailio is not doing anything with these messages. If it’s supposed to handle them, can someone help me fix this issue?

2 6

Re: Will kamailio handle sip messages if Request contains public Ip
by airsay＠duck.com 04 Jun '25

04 Jun '25

It should. You need to add a listen and alias in the kamailio.cfg file listen=udp:1x5.2x6.1x8.1x3:5060 alias=1x5.2x6.1x8.1x3 BR, airsay > On 4 Jun 2025, at 10:46, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. > The SIP message received looks like this: > INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0

2 1

adding codecs to rtpengine
by Jeff Brower 03 Jun '25

03 Jun '25

Hi all, Is there a documented procedure for adding codecs to rtpengine ? As I understand it so far, we would adapt rtpengine encoder and decoder interfaces to the voplib API (documented at https://github.com/signalogic/SigSRF_SDK/blob/master/codecs_readme.md), allowing use of voplib shared object libs. I've been searching (e.g. adding codecs to rtpengine site:kamailio.org) but have not found a step-by-step procedure yet. Thanks for any advice / guidance. -Jeff

3 2

event_route[xhttp:request] is not executed
by Володимир Іванець 03 Jun '25

03 Jun '25

Hello! I'm doing some testing with http requests but got stuck at the beginning. event_route[xhttp:request] does not seem to be executed. I'm getting the following in the log with the most basic configuration. Am I missing something? Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1238]: tcpconn_new(): on port 46328, type 2, socket 23 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1630]: tcpconn_add(): hashes: 1391:1425:2615, 2 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:367]: io_watch_add(): processing io_watch_add(0xb22d80, 23, 2, 0x7f7a63877f30) - fd_no=17 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xb22d80, 23, -1, 0x0) fd_no=18 called Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4671]: handle_tcpconn_ev(): sending to child, events 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4335]: send2child(): selected tcp worker idx:1 proc:10 pid:9158 for activity on [tcp:127.0.0.1:8089], 0x7f7a63877f30 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1782]: handle_io(): received n=8 con=0x7f7a63877f30, fd=6 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:627]: tcp_read_headers(): no clen, p=A Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) ERROR: <core> [core/tcp_read.c:1555]: tcp_read_req(): bad request, state=7, error=4 buf: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: parsed: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1562]: tcp_read_req(): received from: port 46328 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): received from: ip127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1704]: release_tcpconn(): releasing con 0x7f7a63877f30, state -2, fd=6, id=2 ([127.0.0.1]:46328 -> [127.0.0.1]:8089) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1705]: release_tcpconn(): extra_data (nil) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3745]: handle_tcp_child(): reader response= 7f7a63877f30, -2 from 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3668]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3676]: tcp_emit_closed_event(): no callback registering for handling TCP closed event #!KAMAILIO children = 4 tcp_children = 4 debug=3 log_stderror=yes memdbg=5 memlog=5 fork=yes #!substdef "!LOCAL_UDP!udp:127.0.0.1:5060!g" #!substdef "!LOCAL_HTTP!tcp:127.0.0.1:8089!g" listen=LOCAL_UDP listen=LOCAL_HTTP loadmodule "xlog.so" loadmodule "pv.so" loadmodule "ctl.so" loadmodule "sl.so" loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } Thank you very much!

2 1

Kamailio Dispatcher for Docker Asterisk
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello everyone, I have Kamailio running on port 5060, and I'm running Asterisk inside a Docker container. I'm trying to connect to Asterisk via port 5061. Could someone please advise what I should add to my dispatcher list? Currently, I have the dispatcher.list entry: 1 sip:10.0.0.5:5061;transport=tcp 0 Here’s the network info for my Kamailio host (eth0 interface): inet 10.0.0.4/24 brd 10.0.0.255 scope global eth0 And here are the relevant port mappings from Docker: 5038/tcp -> 0.0.0.0:5038 5060/tcp -> 0.0.0.0:5061 5060/udp -> 0.0.0.0:5061 ... My main question: Is the dispatcher entry sip:10.0.0.5:5061;transport=tcp correct if Asterisk is listening on port 5061 in the container? Do I need to adjust anything on the Kamailio side or change the dispatcher entry format? Thanks in advance for any help!

3 5

Small Question
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello Everyone, I have a cloud vm with public ip: 135.2xx.1xx.1xx:5060 and locally 10.0.0.4 When I receive a SIP message the invite looks like this, it gets stuck here. SIP Invite: 2025/06/03 16:24:58.249284 192.76.120.10:5060 -> 10.0.0.4:5060 INVITE sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060 SIP/2.0 Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=765jrrKarBe3F> Record-Route: <sip:10.255.0.1;r2=on;lr;ftag=765jrrKarBe3F> Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bKbb37.28332dbc5901e729308403cd5e049713.0 v:SIP/2.0/UDP 10.239.44.224:6000;received=10.239.44.224;rport=6000;branch=z9hG4bKD4UtjN52Qrm9D Max-Forwards:55 f:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.com>;tag=765jrrKarBe3F t:<sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060> i:c64314db-75b9-4029-8147-be3f3d77da2a CSeq:99930157 INVITE m:<sip:mod_sofia@10.239.44.224:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:205 P-Asserted-Identity:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1748940128 1748940129 IN IP4 50.114.150.25 s=Telnyx c=IN IP4 50.114.150.25 t=0 0 m=audio 27770 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:27771 IN IP4 50.114.150.25 a=ptime:20 Kamailio Code: #!define DS_LIST "/etc/kamailio/dispatcher.list" #!define LISTEN_UDP_PUBLIC udp:135.2xx.1xx.1xx:5060 #!define LISTEN_UDP_PRIVATE udp:10.0.0.4:5060 #!define LISTEN_TCP_PUBLIC tcp:135.2xx.1xx.1xx:5060 #!define LISTEN_TCP_PRIVATE tcp:10.0.0.4:5060 #!define EXTERNAL_IP "10.0.0.4" #!define KUBERNETES_NETWORK "172.19." #!define FLAG_TELNYX_CALL 12 #set module path mpath = "/usr/lib/x86_64-linux-gnu/kamailio/modules" ####### Global Parameters ######### #mhomed=1 # doesn't work with virtual IPs #!ifdef WITH_DEBUG debug=4 #!else debug=2 #!endif log_stderror=no memdbg=5 memlog=5 #log_facility=LOG_LOCAL6 log_facility=LOG_LOCAL0 fork=yes children=4 auto_aliases=no #listen=LISTEN_UDP_PUBLIC listen=LISTEN_UDP_PRIVATE listen=LISTEN_TCP_PRIVATE ####### FLAGS ###### #!define FLAG_FROM_ASTERISK 10 #!define FLAG_FROM_PEER 11 sip_warning=no

1 0

Kamailio as Proxy only
by disa_369＠msn.com 03 Jun '25

03 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from Avaya PBX -------> Kamailio ----> Video System. We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. I got questions. 1. Do we need install maria_db for that? 2. Do we need create subscriber\peer on Kamailio? 3. Do we need add modules for that? And main question what to add in kamailio.cfg is? Thx!!!!!!!!!

2 2

Kamailio Regex
by Timothy Delin 03 Jun '25

03 Jun '25

Hi everyone, I need your advice. I am extracting a SIP Body message from a User Agent Client using the regex as such "$var(calling_party)= $(var(body){re.subst,/.*Calling-party:\s*sip:(\+[0-9]+)@.*/\1/});" And I tried to log it as "xlog("L_INFO", "Extracted number 1: $var(calling_party) <><><>\n");" However, the output in my logs is only showing as below, where I cannot see any value after my variables. "1(6) INFO: {1 10 SUBSCRIBE 5017ae21204c72a3} <script>: Extracted number 1: +6512345678" I suspect this is the reason why when I tried to use this variable in a redis_cmd, it returns as null as the variable may have impacted the command input to Redis. I tried to change the $var(calling_party)="+6512345678" and it is working perfectly fine Best regards, Timothy

3 5

Re: Kamailio as Proxy only
by airsay＠duck.com 30 May '25

30 May '25

I'm not very skilled with Kamailio config. But it sounds like dsiprouter (a graphical UI built ontop of Kamailio) would be your easiest solution. On Dsiprouter, You can set up Avaya as a Carrier Group and Video conference as an Endpoint Group. Then route accordingly. Sent from my iPhone > On 30 May 2025, at 12:19, disa_369--- via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello to everyone. > I deployed Kamailio on Debian system. Install was successfull. > I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. > I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. > Outgoing calls only from > Avaya PBX -------> Kamailio ----> Video System. > We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. > Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. > I got questions. > 1. Do we need install maria_db for that? > 2. Do we need create subscriber\peer on Kamailio? > 3. Do we need add modules for that? > > And main question what to add in kamailio.cfg is? > > Thx!!!!!!!!! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

1 0

Hyperthreading and high request volumes in modern times
by Brooks Bridges 30 May '25

30 May '25

So, I have a platform that handles in the order of around 2500 or more call attempts per second. There are 2 servers that are built that handle this traffic, "old" and "new". The "old" design is a bit heavy, has multiple redis calls to external IO, does a lot of "work" on each invite before passing it along, etc. The "new" design is extremely lightweight, only a couple of htable comparisons and a dispatcher check on each invite before passing it along. The 2 servers are running different versions of Kamailio, however they're both in the 5.x train and I've found nothing in the changelogs that I believe would explain a significant difference in how child process CPU scheduling is performed, especially a detrimental change. I'm more than happy to be proven wrong about that though! The "old" system is running 5.4.8: This system is running 4 x 8 core CPUs (32 cores) with no hyperthreading (4 x E5-4627 v2 to be exact) and has multiple external IO interactions with things like redis, lots of "work" being done on each invite checking various things before allowing it to proceed. This is currently running 32 child processes and again, has no issues handling the requests whatsoever and has been running at this level and higher for literal years. The "new" system is running 5.8.3: This system is running 2 x 8 core CPUs (16 cores) *with* hyperthreading (2 x E5-2667 v4 to be exact) and has no external IO interactions with anything, and only does a couple of checks against some hash table values and dispatcher before allowing it to proceed. This is also currently running 32 child processes, however I am experiencing spikes in context switching and odd memory access issues (e.g. backtraces show things like "<error: Cannot access memory at address 0x14944ec60>" in udp_rvc_loop) with similar loads. What I'm looking for here is empirical knowledge about things specifically like context switching and if hyperthreading is detrimental in cases where I'm doing very high volume, low latency (e.g. no waiting on external replies, etc) work. Is it possible there are additional issues with the way hyperthreading works and the concepts behind it if the individual calls for CPU resources are so "quick" as to overwhelm the scheduler and cause congestion in the CPU's thread scheduler itself? Have you run into anything like this? Have you discovered that very low latency call processing can incur more hyperthreading overhead than the benefit you get from it? Do you have additional data I may be missing to help me justify to the C suite to change out the system architecture to provide a higher physical core count? Can you share it? Thanks!

6 14

Sca in cluster/load balancer
by mauro.celli76＠gmail.com 29 May '25

29 May '25

We have two kamailio in load balancing, mysql is in cluster mode with galera. We need to use sca, but kamailio db_cluster is not supported (i need to specify connection string to sql not cluster name) and if phone1 is registered on kamailio1 and phone2 on kamailio2, sca not work. It is possibile to implement cluster?

3 5

How to count the first re-register request?
by James Morrison 26 May '25

26 May '25

I want to count the first REGISTER request during an initial register or re-register from UE. In my scscf instance, I have defined att_init_reg and att_rereg counters to count the number of initial registers and re-registers respectively. In my early tests, the following piece of logic worked as expected to filter the very first register request (one that is sent from UE before getting challenged). However, more tests showed that during re-registers, UE sends a REGISTER request, containing an authorization header and non-empty response, in that case the code is unable to distinguish a re-register attempt and the related counter fails to increment. Is there a way to fix this issue? #!ifdef WITH_XHTTP_PROM /* Only count the first register request */ if (!is_present_hf("Authorization") || search_hf("Authorization","response=\"\"","f")) { if ($var(impu_registered) != 1) { prom_counter_inc("att_init_reg", "1"); } else { prom_counter_inc("att_rereg", "1"); } } #!endif

3 5

5.7.7 release, tag and docker build
by Henning Westerholt 23 May '25

23 May '25

Hello, I have noticed that the git tag for release 5.7.7 was missing, I've just set it. The 5.7.7-bullseye docker image is also missing from https://github.com/kamailio/kamailio-docker/pkgs/container/kamailio Not sure if this is done automatically, if not - please can somebody please trigger the workflow to create it? Thank you, Henning

1 0

Help Adding WebRTC to Existing SIP Setup
by Fernando Lopes 22 May '25

22 May '25

Hi everyone, I currently have a working SIP setup, but I'm trying to get WebRTC working alongside it and haven’t had much success so far. Does anyone have tips or best practices for enabling both SIP and WebRTC in the same configuration? Is it possible to handle both within a single Kamailio config? Or would I need some kind of conditional logic (like if/else) to manage how the config handles different types of clients? My setup looks like this: Router → Kamailio → Kubernetes (Asterisk inside the cluster) I was having some issues with the Contact header, but I managed to resolve that. Now I'm stuck figuring out how to make WebRTC work cleanly alongside SIP. Any advice would be appreciated! Thanks!

2 1

Kamailio v5.7.7 Released
by Daniel-Constantin Mierla 21 May '25

21 May '25

Hello, Kamailio SIP Server v5.7.7 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.7. Note that 5.7 is now the third last stable branch, v5.7.7 being the last planned released in 5.7.x series. The latest two stable branch are 5.8 and 6.0, with v6.0.1 being released a while ago. For more details about version 5.7.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/05/kamailio-v5-7-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

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