- sr-users - kamailio.org

by Alexis Fidalgo

Hello all, little and fast context, we have 8 docker swarm clusters (4 nodes each) where there are deployed as global service 4 kamailio per cluster. Each cluster runs 8 replicas of a call processing service/app (for the sake of this question we’ll assume the performance of this app is ok, no problems on what it does and how much it consumes in cpu/memory). kamailio is configured to use http_client module to query the processing app (those queries are load balanced by the docker swarm cluster) in a very simple way, added below in this email. Everything goes well, there’s an average of ~450 caps on each kamailio on peak hours. Question and reason of this email is: Is this the best approach i can take? (the http_client) or maybe there’s a more efficient one for this scenario? maybe kemi, or other? The processing app is written in golang. We’re planning to add more traffic (significantly more) to this instances and i would like to know if there’s a better scenario, to explore it before this migration. Thanks, regards — request_route { route(AF_SL); } route[AF_SL] { if(is_method("ACK")){ exit; } if(is_method("REGISTER")){ sl_reply("200", "OK"); exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } if(is_method("INVITE")){ if ($hdr(X-Carrier-In)!="" && $hdr(Organization)!="" && $hdr(X-SBC-Type)!="") { sl_reply("100","Trying"); $var(retCode) = http_client_query("http://cbrcre:3001/cbrcre/v1/callHandler",'xsbct=$(hdr(X-SBC-Type){s.escape.param})&org=$(hdr(Organization){s.escape.param})&xci=$(hdr(X-Carrier-In){s.escape.param})&fu=$(fu{s.escape.param})&ru=$(ru{s.escape.param})&rU=$(rU{s.escape.param})&fU=$(fU{s.escape.param})&ci=$(ci{s.escape.param})&cin=$(hdr(Call-Info){s.escape.param})&ih=$hdr(Identity)&tag=$(ft{s.escape.param})&ct=$(ct{s.escape.param})&via=$(hdr(Via)[0]{s.escape.param})', "$var(result)"); xlog("L_INFO", "*************** RESULT: $var(result)\n\n"); jansson_get("response_code", "$var(result)", "$var(rc)"); switch($var(rc)){ case "200": if jansson_get("contact", "$var(result)", "$var(ct)") == true { append_branch("$var(ct)"); } if jansson_get("callinfo", "$var(result)", "$var(cinfo)") == true { append_to_reply("Call-Info: $var(cinfo)\r\n"); } if jansson_get("pai", "$var(result)", "$var(pai)") ==true { append_to_reply("P-Asserted-Identity: $var(pai)\r\n"); } if jansson_get("xfrom", "$var(result)", "$var(xfrom)") ==true { append_to_reply("X-From: $var(xfrom)\r\n"); } if jansson_get("identity", "$var(result)", "$var(identity)") ==true { append_to_reply("Identity: $var(identity)\r\n"); } append_to_reply("X-Handled: ServiceBroker\r\n"); sl_send_reply("302", "Redirect"); break; default: jansson_get("reason","$var(result)","$var(reason)"); jansson_get("sbReason","$var(result)","$var(sbreason)"); append_to_reply("X-SBReason: $var(sbreason)\r\n"); sl_send_reply("$var(rc)", "$var(reason)"); } exit; } xlog("L_ERR","missing xci or org or sbct"); sl_send_reply("608", "SBK Rejected"); } }

2 months

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ERROR: <core> [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif

by Tristan Ferreira

Hi, I am trying to learn Kamailio to set up a SIP/RTP proxy for WebRTC with FreeSWITCH, and I wanted to use this configuration file template as a starting point. (Source: https://github.com/havfo/WEBRTC-to-SIP) But i have the error : *ERROR: [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif* I don't understand because I checked and everything seems okay with the if/endif statements to me. |# grep '#!ifndef' kamailio.cfg | wc -l 2 # grep '#!ifdef' kamailio.cfg | wc -l 49 # grep '#!endif' kamailio.cfg | wc -l 51 | And it's ok for chat gpt too : |#!ifdef : 49 occurrences #!ifndef : 2 occurrences Total (#!ifdef and #!ifndef) : 51 occurrences #!endif : 51 occurrences | Is there a problem with the parser or a compatibility problem (i am on*5.6.3*) ? |#!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD #!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!XXXXX-XXXXX!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![XXXXXX-XXXXXX]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!XXXX-XXXX!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* 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", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!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("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@XXXX-XXXX") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!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) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication 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(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($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()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... 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); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri != myself && uri != myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else if (is_first_hop()) { set_contact_alias(); } setflag(FLT_NATS); } return; } # NAT handling route[NATMANAGE] { if (is_request()) { if (has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif ) ) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # SIP --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); 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] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif | Thank.

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Compatibility issue between proxy and registrar

by satyaprakash ch

Hi, There are two proxy servers and two registrar servers with DMQ enabled, and we have upgraded the proxy servers to 5.7.4 and the registrar servers to 5.5.5. Kamailio proxy version 5.7.4 Kamailio registrar version: 5.5.5 Following the upgrade of the proxy server to 5.7.4, calls are not working and no response is received. Once again, we have reverted the proxy server to 5.5.5, so that the proxy server and the registrar server version are now the same.The calls are now working as expected. We suspect it could be an DMQ issue. I would appreciate your suggestions on this matter of compatibility between proxy and registry servers. Thanks.

2 months

3
3
0 0

Incoming IPv6 is send to IPv4 of the PBX instead of IPv6

by Bernd Krueger-Knauber

Hi, I set up a transparent kamailo proxy for IPv4 and IPv6. The PBX which should be reached can handle both. Now I have the case that the phone comes in on IPv6, but kamailio sends it out via IPv4. If I ping the PBX on the kamailio server (debian) I get the IPv6 address, since it is preferred. Why does kamailio send to the IPv4 address of the PBX? And how can I change this? There are only A and AAAA records in the DNS dns_try_ipv6=yes is set. Debian 12.5 Kamailio 5.6.5 Any hint is welcome. Best regards Bernd

2 months, 1 week

3
4
0 0

Missing newline in dbtext tables causes crash and core dump

by Marrold

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine. I used ansible to template some files, and a newline was omitted at the end of the uacreg table due to a templating issue. This caused kamailio to crash on start up, with the error: 20(27) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 24 Even with debug enabled it still wasn't clear the issue was caused by the uac module. I had to do a gdb backtrace to figure it out. Whilst the mistake was on my part, I wondered if this should be handled more cleanly? Cheers Matthew

2 months, 1 week

2
1
0 0

Problems building with stirshaken

by Martin Nyström

I am trying to build Kamailio with repo tags/5.6.4 to get my hands on the stirshaken module that was not part of apt repositories, however it results in a build error: root@ip-fobby:/usr/src/kamailio# make modules modules=modules/stirshaken make -C src/ modules make[1]: Entering directory '/usr/src/kamailio/src' CC (gcc) [M stirshaken.so] stirshaken_mod.o stirshaken_mod.c:28:10: fatal error: stir_shaken.h: No such file or directory 28 | #include <stir_shaken.h> | ^~~~~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: stirshaken_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' make: *** [Makefile:34: modules] Error 2 Obviously there’s no stir_shaken.h file withib the directory src/modules/stirshaken, but I cannot see why, or how I proceed? Is this a likely symptom of faulty libstirshaken installation? /M

2 months, 1 week

2
1
0 0

lcr module, avp_subst to $avp(i:709)

by Ricardo Castillo

Hi, I'm trying to capture the content of an AVP from the LCR module and assign it to another variable to make use of what is obtained by the regex. For this, I have used the avp_subst. The content of $avp(i:709): 1|sip:|0|||2416027840||5061||;transport=udp|11|1 I need to extract the index that indicates the port "5061" at position 8. I haven't been able to achieve this by trying several regex patterns. Has anyone experienced something similar? Thanks!

2 months, 1 week

2
2
1 0

Kamailio crash due to UDP receiver child (kamailio 5.5.5)

by satyaprakash ch

Hi, The kamailio version we are using is "5.5.5", We are facing a kamailio crash issue recently and it has generated the core-dump. Based on the core-dump analysis we could see that "*si_desc = "udp receiver child=3" *getting crashed. We could see the below error in the core-dump trace, #22 0x0000562c33524750 in udp_rcv_loop () at core/udp_server.c:543 tmp = 0x1150955ff0 <error: Cannot access memory at address 0x1150955ff0> Please suggest how to fix this and your inputs will be appreciated. Note: Due to confidentiality, I'm not able to share the core-dump trace. Thanks.

2 months, 1 week

3
3
0 0

[--SPAM--] Designer Watches Starting at $250 - Shop Today!

by Tom

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2 months, 1 week

1
0
0 0

[--SPAM--] Mail account unusual sign-in activity

by Mail Administrator

2 months, 1 week

1
0
0 0

dynamic selection of authentication algorithm

by Juha Heinanen

Currently digest authentication hash algorithm (MD5 or SHA-256) is selected when auth module is loaded. How about allowing to select it also dynamically using the function flags? -- Juha

2 months, 1 week

1
1
0 0

Problem with logrotate in Ubuntu 22.04!

by christian.marinelli＠hotmail.it

Hi all, i set up a simple Kamailio SIP server with the latest stable versione. Than, by following the documntation, i tried to set up a file log instead of the standard syslog log, so i changed the syslog.conf by adding this lines: # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log Then, i created a kamailio file in /etc/logrotate.d with this content to manage the log rotation: /var/log/kamailio.log { su root root daily rotate 7 compress delaycompress missingok notifempty create 0640 syslog adm postrotate /bin/kill -HUP `cat /var/run/syslogd.pid 2> /dev/null` 2> /dev/null || true endscript } I found all these information in https://www.kamailio.org/wikidocs/tutorials/3.2.x/syslog/ Unfortunatly, in my Ubuntu 22.04 server there isn't a syslogd.pid file in that path so the logrotation doesn't work. Can anyone help me to understand the problem? Thank you so much in advance Christian

2 months, 1 week

2
2
1 0

[--SPAM--] Last Reminder: Urgent Two-Factor Authentication Activation

by MеtaMаsk

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2 months, 1 week

1
0
0 0

Newbie: accounting and billing

by Ghulam Mustafa

Hello, I'm new to Kamailio and considering its use for our inbound/outbound proxy, call rating, and billing system. I'd like to get your feedback on my current approach and understand the best practices for this scenario. Currently, I'm using SQL queries within my Kamailio configuration to handle authorization and rating. Here's a simplified example: # Call authorization route[DISPATCH] { if (is_method("INVITE") && isbflagset(FLB_SRC_CARRIER)) { if ($au == $null) { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount, cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_ip ip INNER JOIN customer_balance cb ON (ip.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (ip.customer_id = ct.customer_id) INNER JOIN customers cust ON (ip.customer_id = cust.customer_id) WHERE ip_address = '$si' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); } else { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount,cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_auth auth INNER JOIN customer_balance cb ON (auth.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (auth.customer_id = ct.customer_id) INNER JOIN customers cust ON (auth.customer_id = cust.customer_id) WHERE auth.subcriber = '$au' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); $dlg_var(auth) = "1"; } xalert("query is $var(acc_query)"); if (sql_xquery("dbh", "$var(acc_query)", "row") == 1) { $dlg_var(account) = $xavp(row[0]=>customer_id); $dlg_var(dispatcher_group) = $xavp(row[0]=>dispatcher_group); $dlg_var(retail_rate) = $xavp(row[0]=>retail_rate); $dlg_var(wholesale_rate) = $xavp(row[0]=>wholesale_rate); $dlg_var(cps) = $xavp(row[0]=>cps); $dlg_var(service_plan) = $xavp(row[0]=>tag); # $var(cash) = $xavp(row[0]=>cash); xalert("account: $xavp(row[0]=>customer_id) cash: $xavp(row[0]=>cash) retail_rate: $xavp(row[0]=>retail_rate) wholesale rate: $xavp(row[0]=>wholesale_rate) group: $xavp(row[0]=>dispatcher_group) cps: $xavp(row[0]=>cps) tag: $xavp(row[0]=>tag) \n"); if ($xavp(row[0]=>cash) <= $xavp(row[0]=>nobal_amount)) { sl_send_reply("403", "Insufficient balance"); exit; } append_hf("X-Account: $dlg_var(account),$dlg_var(retail_rate),$dlg_var(wholesale_rate)\r\n"); } else { sl_send_reply("500", "Server Error!"); exit; } #!ifdef WITH_CPSLIMIT # perform pipe match for INVITE if (is_method("INVITE")) { $var(invlimit) = $(dlg_var(cps){s.int}); if ($var(invlimit) > 0) { if (!pl_check("$si-$dlg_var(dispatcher_group)", "TAILDROP", "$var(invlimit)")) { pl_drop("5"); exit; } } } #!endif # round robin dispatching on gateways group '1' if(!ds_select_dst("$dlg_var(dispatcher_group)", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } …… ……. …… # Billing and Rating using acc_cdrs and a customer_balance table event_route[dialog:end] { $var(call_duration) = $DLG_lifetime; # Fetch call duration $avp(originator) = $dlg_var(originator); $avp(accountcode) = $dlg_var(account); $avp(caller) = $dlg_var(caller); $avp(callee) = $dlg_var(callee); $avp(pdd) = $dlg_var(pdd); $avp(service_plan) = $dlg_var(service_plan); $avp(hangup_source) = 'CALLEE'; $avp(hangup_code) = '200'; $avp(hangup_reason) = 'Normal Clearing'; $avp(elapsed) = ( $TV(s) - $dlg_var(init_timestamp) ); if ($dlg_var(auth) != $null) { setbflag(FLB_SRC_CARRIER); } if (isbflagset(FLB_SRC_CARRIER) && $dlg_var(originator) == "CUSTOMER") { $avp(hangup_source) = 'CALLER'; } if (isbflagset(FLB_SRC_PBX) && $dlg_var(originator) == "SYSTEM") { $avp(hangup_source) = 'CALLER'; } # Calculate billed amount using postgres query $var(bill_query) = $_s(select cast(round($dlg_var(retail_rate)/60::numeric * $var(call_duration), 4) as varchar) as bill;); if (sql_xquery("dbh", "$var(bill_query)", "row") == 1) { $avp(billed) = $xavp(row[0]=>bill); } if (isbflagset(FLB_SRC_CARRIER)) { $var(query) = "UPDATE customer_balance set cash = cash - (" + $dlg_var(retail_rate) + " / 60 * " + $var(call_duration) + ") WHERE customer_id = " + $dlg_var(account) + ";"; sql_xquery("dbh", "$var(query)", "ra"); } # xalert("~~~~~ Call ended (duration: $var(call_duration)) total_billed: $avp(billed) ~ $avp(hangup_source) ~ $T(reply_code) ~ $T(reply_reason) ~ $T_reply_code ~ \n"); } My questions are: • Is this a sustainable approach for a Kamailio billing/rating setup? Are there performance or scalability concerns? • Would it be better to handle rating and billing directly on our FreeSWITCH B2BUA? What are the advantages and disadvantages of each approach? • Are there best practices or alternative solutions I should consider? Thank you in advance for sharing your experience and insights!

2 months, 1 week

4
3
1 0

compile without downtime

by satyaprakash ch

HI I currently have two proxy servers running in my environment, and I want to upgrade Kamailio from version 5.4.7 to 5.6. Can I recompile the Kamailio service before restarting it without downtime ? Will there be any impact on my production environment if I recompile Kamailio to the newer version? If there is no impact, I expect approximately 2 minutes of downtime during the service restart. Here's the scenario: *No Downtime:* Clone the Kamailio repository with git clone https://github.com/kamailio/kamailio.git Navigate to the Kamailio directory with cd kamailio Update the modules.lst file with any necessary additions Compile with make Install with make install *Downtime:*Restart the Kamailio service. Please advise on this. Thanks.

2 months, 1 week

3
2
0 0

make Debian deb package + non kstandard modules

by Victor Velo

Hello, I would like to build some additional modules to the standard by following this guide (old) https://www.kamailio.org/wikidocs/install/4.0.x/debian/ and I don't get these additional .deb packages. Could someone please help. Thanks, Victor

2 months, 2 weeks

1
0
0 0

Re: Real life examples of cost saving from using Kamailio and other FOSS SIP software

by JR Richardson

Hey Folks, What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection. FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in voice switches. A key consideration is expected/targeted growth of the solution and operational expediency, how big with the subscriber base get, voice service is a high touch point for customer support. Once you (as in personally) deploy a FOSS solution in production, you are tied to that solution for its foreseeable existence, good luck finding competent and like-minded staff that see your vision and can support the solution as you do. Deployment guys will always have a deeper understanding/knowledge of the solution, architecture, configuration, pitfalls and work around than the operations support folks. If you don't have staff resources, having consultants you can rely on is crucial for success. I can't say enough about Kamailio for building call routing solutions, mature, stable, reliable, continued improvements, phenomenal developer and community base for support. Good luck whichever direction you go in! JR Richardson On Apr 14, 2024, at 12:25 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Medium to large organisations, in particular, tend to extract labour from consistent (if unextraordinary) output of everyday do-gooders, and not the more stochastic and volatile heroics of open-source superstars. Sorry, I meant to say "extract value" here, not "extract labour". But the larger point is that any successful formula heavily reliant on open-source is going to be a lot more dependent on culture, and on the strengths of individual people and their specific skills, than a formula reliant on a third-party vendor, all other things being equal. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 End of sr-users Digest, Vol 227, Issue 36 *****************************************

2 months, 2 weeks

1
0
0 0

Issue with MS Teams OPTIONS ping - no response

by Barry Flanagan

Hi Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71 CSeq: 10 OPTIONS Call-ID: 6ea8091201db565b-605531(a)159.65.196.10 Max-Forwards: 70 Content-Length: 0 User-Agent: ABC SBC Contact: <sip:msteams.domain.com:5061;transport=tls> From the logs (below) I can see that w are accepting the MS cert OK but get no response and ends with a 408. The domain is configured in Teams, and a user with a license was added. MY Cert is issued by GeoTrust TLS RSA CA G1. Any idea if the issue might be on my side? 024-04-15T14:53:19.302610+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_domain.c:798]: sr_ssl_ctx_info_callback(): SSL handshake done 2024-04-15T14:53:19.302692+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful 2024-04-15T14:53:19.302762+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:545]: tls_connect(): tls_connect: new connection to 52.114.14.70:5061 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 2024-04-15T14:53:19.302824+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:550]: tls_connect(): tls_connect: sending socket: 159.65.196.10:0 2024-04-15T14:53:19.302868+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 2024-04-15T14:53:19.302917+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 03 2024-04-15T14:53:19.470476+00:00 msteams /usr/sbin/kamailio[605606]: DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fd3c77639c8 2024-04-15T14:53:28.770710+00:00 msteams /usr/sbin/kamailio[605591]: DEBUG: dispatcher [dispatch.c:3617]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sip:sip.pstnhub.microsoft.com;transport=tls, group 1) -Barry

2 months, 2 weeks

3
3
0 0

Memory leaks - KEMI Lua

by Muhammad Danish Moosa

Hi, I am using kamailio with KEMI Lua. The script has some medium level complexity as it stores some information from the invite in redis cache during the call and uses that information during the call transfers (REFER and Invite triggered by REFER). Lua script itself creates/sets some variables, and sends/retrieves information during the flow for example KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " .. KSR.pv.getw("$ci") .. " EX 3600", "r"); KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r"); reft=KSR.pv.getw("$redis(r=>value)"); I made a test call and disconnected, what I am observing is that even if the call is finished the used memory is slightly increased. before: < real_used: 844240 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 after: > real_used: 846080 > real_used: 845864 > real_used: 846584 > real_used: 846304 > real_used: 846200 > real_used: 846120 > real_used: 845248 > real_used: 846168 My assumption is after finishing the call, memory used by variables and redis ops is released and should be back to original values but its not happening. In production this issue is causing some problems as I have to restart the kamailio process after every 3-4 weeks so that it starts breathing with more available memory. Initially I thought its Kamailio version issue as in my last posts in this forum people advised me to upgrade to the latest kamailio version (original one was 5.4.0) but the same issue is happening on 5.7.0. Can someone explain how to debug/troubleshoot this situation . I have a parallel lab environment, I do not see anything interesting in syslogs. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 months, 2 weeks

3
4
0 0

About the operating environment of Kamailio

by 山本哲也

Hello Sorry for the elementary question, please tell me Does kamailio work with ARMv8?

2 months, 2 weeks

5
5
0 0

Maintenance work overnight on kamailio.org server

by Daniel-Constantin Mierla

Hello, there is some planned work on the data center infrastructure hosting kamailio.org server that is going to affect briefly the web server (web site, wiki) and the mailing lists. The downtime is expected to be 10-15 minutes around 1:00am Western Europe time (CEST / GMT+2), but if it takes longer or repeats, be patient for a while. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

2 months, 2 weeks

1
0
0 0

Real life examples of cost saving from using Kamailio and other FOSS SIP software

by Mahmood Alkhalil

Hi Everyone, I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was. I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt. Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!

2 months, 2 weeks

4
13
0 0

show final SIP INVITE before going out

by marek

hi, i have kamailio with TLS i dont have homer and voipmonitor with TLS decryption doesnt work how can i check final SIP INVITE (after all rewrites/modifications) before is send to the peer? tried onsend_route { if ($rm == "INVITE") { xlog("L_INFO", "Final SIP Message before Sending: $mbu\n"); } } but record-route/via are different than reality (checked with sngrep with TLS disabled) kamailio log Record-Route: <sip:;r2=on;lr;ftag=b0d1d4ce-... Record-Route: <sip:10.10.10.10;r2=on;lr;ftag=b0d1d4.. Via: SIP/2.0/UDP 30.30.30.30:5080;TH=div;rport;branch=z9hG4bK sngrep Record-Route: <sip:9.9.9.9;r2=on;lr;ftag=246bdf1c... Record-Route: <sip:9.9.9.9;line=sr-Jf7BrWSBdWSzrD Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKdb73.0735168f3e5f.... Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKsr-2y7Ud3XCMD4N6Vi8M-iCM-XGdWSBMDGzuhJZu-iGMh9ztEu7q..... 10.10.10.10 - private kamailio ip (fictional) 30.30.30.30 - ip of SIP PBX (caller) 9.9.9.9 - public kamailio ip Marek

2 months, 2 weeks

2
1
0 0

Unable to remove/modify/replace "Contact SIP header'

by ramaseshi.kolli＠gmail.com

Hi All, My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please? KSR.textops.remove_hf("Contact"); KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>"); KSR.rtpengine.rtpengine_manage0(); if KSR.tm.t_relay()<0 then KSR.sl.sl_reply_error(); end

2 months, 2 weeks

3
5
0 0

503(service unavailable)

by masoud96＠mail.ru

Great everyone! It often happens to me that the service completely fails and refuses to service calls. For example, yesterday there was such a situation. I looked at the logs and at the beginning of the incident I noticed this strange entry: kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1297]: find_listening_sock_info(): binding to source address 109.68.232.40:15061 failed: Address already in use [98] kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1441]: tcp_do_connect(): 10.241.60.46:15061: could not find corresponding listening socket kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3078]: tcpconn_1st_send(): connect 10.241.60.46:15061 failed (RST) Connection refused kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3087]: tcpconn_1st_send(): 10.241.60.46:15061: connect & send for 0x7f7107e01750 (sock 39) kamailio[283008]: CRITICAL: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) And after several such entries this begins: /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 31.173.85.36:10588 failed (timeout) /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 95.24.163.132:48075 failed (timeout) /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668011 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668304 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668309 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668316 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668192 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668228 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668236 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668427 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668428 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668429 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668430 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668431 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668432 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668433 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668434 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668435 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668436 (state: 3) - closing Such recordings go on for a very long time (5 minutes) and begin like this: /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 286 requests queued (total handled 2421746) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 286 requests queued (total handled 2419460) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 4, socket 241: queue full, 287 requests queued (total handled 2451838) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 5, socket 243: queue full, 287 requests queued (total handled 2445182) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 18, socket 269: queue full, 287 requests queued (total handled 2422113) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 287 requests queued (total handled 2421747) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 27, socket 287: queue full, 287 requests queued (total handled 2419986) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 287 requests queued (total handled 2419461) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 30, socket 293: queue full, 287 requests queued (total handled 2419979) Also, my eyes are distracted by such records and it’s difficult to understand the logs: 1) kamailio[282798]: CRITICAL: {2 12704 INVITE 4cb232e6-9b5a-4a5b-aa0e-1ed73539039f} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed 2) /usr/local/sbin/kamailio[282970]: ERROR: sl [sl.c:285]: send_reply(): failed to reply stateful (tm) 3) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection reset by peer (104) ([217.66.156.172]:11604 -> kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f71215bb610 r: 0x7f71215bb738 (-1) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection timed out (110) ([188.170.204.16]:52148 -> kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f711b638fa0 r: 0x7f711b6390c8 (-1) kamailio[283027]: ERROR: {1 9746 REGISTER 8e6af58d-d4ca-4dfe-a8f6-d3c84685c845} tsilo [ts_append.c:121]: ts_append_to(): transaction [23428:1443020265] not found 4) kamailio[282820]: CRITICAL: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[282820]: WARNING: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [t_fwd.c:1581]: t_send_branch(): sending request on branch 0 failed kamailio[282820]: ERROR: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) Once the queue is full, the system stops servicing new calls and freezes. kamailio -v version: kamailio 5.7.2 (x86_64/linux) 95629c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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: 95629c OS: Debian 11, 64 cores, 125GB of DDR4 RAM; DB: MariaDB 10.5.15; udp_children: 30; tcp_children: 34; TLS: YES; async_workers: 64. Incoming calls are sent via push notifications( Federico Cabiddu method: https://www.voztovoice.org/sites/default/files/KamilioWorld2015%20-Federico…). NetBridging(for SIP and RTPEngine). ims_charging for billing (integration with our billing system using the Diameter protocol). As soon as the service starts responding with 503, it does not recover on its own, you have to restart. My server is powerful enough for the number of users of my service (what I mean is that there is no problem with resources). For example, 64 cores, 125GB of DDR4 RAM, 2TB of disk, of which 1TB SSD for the database, 10Gigabit channel, this is only a server for Kamailio for Rtpengines separate servers. Please help me figure out what causes a service failure!?

2 months, 2 weeks

1
0
0 0

Kamailio with rtpengine for RTP encryption/decryption

by David Cunningham

Hello, We're using kamailio, with rtpengine for RTP encryption/decryption. It works fine but I would like to understand the integration better. How does rtpengine get the TLS certificates, and what crypto library does it use (openssl?). Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 months, 2 weeks

3
9
0 0

dlg_vars of existing dialog not set after loose_route()

by Benoît Panizzon

Hi Gang I have a strange issue after upgrading to 5.7 I use dlg_vars to keep some information until the end of the call to write them in a CDR. One of those variables records the timestamp of the first invite to determine how long it took from the invite to the 'connection established' if at all. The CDR is written when the BYE is being processed by the dialog module. Strangely, since updating to 5.7 (used to work with 5.5 and 5.6 I think, but I could be mistaking as other changes were made to the config) my dialog variables are null when BYE is being processed. reading the docs, I learned that that dlg_var are only polulated after loose_route() has been called. So I added some logging statement to track if the BYE in question is processed by loose_route(). Yes, it is, loose_route() returns success on the BYE in question. But when I log the content of my dlg_var after loose_route() it is still null. Any idea what the cause could be? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 months, 2 weeks

2
4
0 0

Urgent: High CPU Load and MySQL Connection Issue on Kamailio Proxy Server

by satyaprakash ch

Hi, We're encountering a CPU load of approximately 90% on the Kamailio proxy server. Additionally, there's an error logged indicating a lost connection to MySQL. Could this indicate a problem with the configuration file or is it related to MySQL itself? Your guidance would be appreciated. MySQL version: Ver 14.14 Distrib 5.5.47 Kamailio version: 5.5.5 Error in logs: /usr/local/sbin/kamailio[24933]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013)

2 months, 3 weeks

2
1
0 0

Newbie looking for advice

by Wayne Wenthin

Hi All, I'm new to Kamailio but have been running dozens of Freepbx systems for the last 10 years. We are an Education Service District and service K-12 schools. Many of these we provide dial tone to, the rest we have connections to to provide for toll avoidance and backup outbound calling when the LEC fails. Our main focus is to save schools money that can be used for education instead of infrastructure. Right now I have a single freepbx machine that is using inbound routes to send calls to the proper school district. I would like to replace this with a Kamailio server instead but can't seem to get my mind around the best way to do this. I have 119 inbound routes in my current system with 22 trunks to the other districts. This may increase in the future as we try to migrate districts off of the expensive systems to a more open source way of calling. Currently calls between districts take the route out of the system just to come right back in to be routed to a different district. I would like to avoid that if possible. I've been trying to educate myself on the best practice in this type of structure but can't seem to figure out the best way to make this work. Most of the videos and websites assume a pretty intimate knowledge of Kamailio. That and the database structure doesn't make sense to me. I would like to keep each freepbx as its own registrar as we have lots of crossover with extensions and a few of the districts have their own staff that manage that aspect of the machines. . Any suggestions would be appreciated. Please ask questions if this just looks like I'm rambling. -- Wayne Wenthin Wide Area Network Administrator | CTA@MESD 503.257.1562 Phone/Fax www.cascadetech.org

2 months, 3 weeks

2
1
0 0

Troubleshooting acc_radius module with 5.7

by santiago.troncoso＠quobis.com

Hi community, I am preparing a setup of Kamailio with radius accounting on a docker-compose environment. When configuring the acc_radius module I had to navigate through some old doc by Daniel and inspecting the code itself. The thing is that it seems working, but not. And, yes, I already trying without docker :) Context: two containers, host newtork, to keep it simple - one with kamailio + modules (acc, acc_radius with rad_cli) + radius client configuration - another with freeradius + radius client configuration. I ensure that all data correspond and radius config is fine and spin them up. Run some radius commands to test and let's try some accounting... Executing a failed call and triggering the account flag method, I receive an opaque -1 error from the module: ``` kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: Initial requests kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: FLAG FLT_ACC set kamailio-1 | 1(20) NOTICE: {1 22 INVITE yutncivdtflvzga@PT-96} acc [acc.c:286]: acc_log_request(): ACC: call missed: timestamp=1709746392;method=INVITE;from_tag=wvaqi;to_tag=1a86669fc745788378faa461e7993953-78550000;call_id=yutncivdtflvzga@PT-96;code=404;reason=Not Found;src_user=1000;src_domain=kamailio-local;src_ip=172.20.0.1;dst_ouser=1001;dst_user=1001;dst_domain=kamailio-local kamailio-1 | 1(20) ERROR: {1 22 INVITE yutncivdtflvzga@PT-96} acc_radius [acc_radius_mod.c:497]: acc_radius_send_request(): Radius accounting - ERROR - result: -1 ``` Log accounting is working ok, but radius accounting not. Time for troubleshooting. I tried to enable client debugging, but the module does not respond. So I modify the source directly... tested with `radclient` , `radtest`, `radsniff`... all seems to work when trying manually... I also used an internal radcli fuction `rc_check()` and it works. Meanwhile, I was unable to make it work with the module and the radcli library. So another log output using internal functions to debug the configuration: ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: nas-identifier, Valor: radius-wac kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: authserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: acctserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: servers, Valor: /usr/local/etc/radcli/servers - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: dictionary, Valor: /usr/local/etc/radcli/dictionary - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:300]: check_cfg(): Clave: default_realm, Valor: -- not set -- kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_timeout, Valor: 10 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_retries, Valor: 3 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: bindaddr, Valor: * kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: clientdebug, Valor: 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:369]: init_acc_rad(): RC_ACCT_INIT: rc_test_config looks good kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:375]: init_acc_rad(): RC_ACCT_INIT: Config applyed ok ``` The configuration seems ok, accepted by the library and seems to be applied properly, but when trying to parse its values... all seems like rubish... ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:401]: init_acc_rad(): RC_ACCT_INIT: Name: P�y:^ kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:402]: init_acc_rad(): RC_ACCT_INIT: Max 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:403]: init_acc_rad(): RC_ACCT_INIT: Port: 125489528 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:404]: init_acc_rad(): RC_ACCT_INIT: Secret: kamailio-1 | 0(18) ERROR: acc_radius [acc_radius_mod.c:410]: init_acc_rad(): RC_CLIENTDEBUG: No clientdebug specified ``` I'm using `libradcli4 1.2.11-1+b2` for running on the container and `libradcli-dev 1.2.11-1build1` for compiling. Open to hear ideas, somebody using it actively on latest 5.7 release? Some updated documentation flying around? I was thinking to open a GitHub issue to handle the documentation issue. Thanks in advance!

2 months, 3 weeks

1
1
0 0

Updating $fn malforms packet.

by James Lipski

Hello, I have remote users registering/ routing calls through my server. When calls do pass through the server, the server updates the from name, to uri and from uri via updating the pseudovariables $fn, $tu, and $fu respectively. While this is has been working, I do occasionally see calls failing to process as the remote gateway fails the call with a "400 Bad From header". Inspecting the A-leg INVITE and the B-leg INVITE after updating the pseudovariables, I see the following: A-leg INVITE: INVITE sip:15551112222@example.com SIP/2.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;rport;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 70 From: sip:12223334444@example.com;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@example.com Contact: <sip:12223334444@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="12223334444", realm="example.com", nonce="ZhUw02YVL6cdi51KMymTxGpzCZCZhd8G", [uri="sip:15551112222@example.com](mailto:uri=)", response="3a25b4ea0b1154e31caee1184a92f12b" Content-Type: application/sdp Content-Length: 627 v=0 o=- 3921660487 3921660487 IN IP4 79..xx.xx.xx s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 79..xx.xx.xx b=TIAS:64000 a=rtcp:4007 IN IP4 79..xx.xx.xx a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=ssrc:63772004 cname:3ff849bd611036a7 ---------------------------------- B-leg INVITE: INVITE sip:15551112222@togateway.com SIP/2.0 Record-Route: <sip:10.64.54.207;lr;ftag=dff3c585a6154f23898e1121918f1752;did=e21.65c1> Via: SIP/2.0/UDP 10.64.54.207:5060;branch=z9hG4bK919d.5ca83e030c0879093169546cd80591a6.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;received=79..xx.xx.xx;rport=61379;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 15 From: sip:12125557777@example.com"Updated Name" ;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@togateway.com Contact: <sip:12125557777@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 609 v=0 o=- 3921660487 3921660487 IN IP4 10.64.54.207 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 18400 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 10.64.54.207 b=TIAS:64000 a=ssrc:63772004 cname:3ff849bd611036a7 a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=sendrecv a=rtcp:18401 Note that I've updated the following on the B-leg: $fn = "Updated Name" // originally nothing $fu = sip:12125557777@example.com // originally sip:12223334444@example.com $tu = sip:15551112222@togateway.com // originally sip:15551112222@example.com While the from uri, and to uri do get updated, the from name is injected in the wrong place of the from header. This only seems to happen when the A-leg is missing the "<" and ">" characters enclosing the from headers. Should I not be updating the pseudo variables in this manner to update the from name, to uri and from uri; is there another way to update these attributes of the header without malforming it? Thank you.

2 months, 3 weeks

2
1
0 0

making HTTP requests in stateless redirects

by Sergio Charrua

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the sessions, iterates through one of the JSON branches and adds the destination URLs to the Contact header and replies a SIP 300 Multiple Choices. This works: the UAC does receive the SIP response with the Contacts , but it also receives a SIP 500 error message right after, and I can't figure out why. And i bet this is simple to solve.... Any clue? I'm aware I'm not processing ACK messages, but the sl_send_reply 403 should do the trick right? ... at least while testing.... Also, as this is a stateless script, is there another way of using async http or making http requests to a REST API without having to use TM Module? Thanks in advance. The kamailio script is has follows: #!KAMAILIO > /* add API http timeout */ > #!define HTTP_API_TIMEOUT 5000 > #!define HTTP_API_ROUTING_ENDPOINT "http://some_python_rest_api/get_route" > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=4 > log_stderror=no > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=2 > /* set paths to location of modules */ > loadmodule "tm.so" > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "sl.so" > loadmodule "db_mysql.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "ipops.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "uac.so" > loadmodule "counters.so" > loadmodule "http_async_client.so" > loadmodule "jansson.so" > loadmodule "usrloc.so" > /* listen addresses */ > listen=udp:10.20.0.2:5060 > listen=udp:10.20.0.2:5062 > advertised_address="10.20.0.2"; > # ----- http_async_client params ----- > modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) > modparam("http_async_client", "connection_timeout", 2000) > request_route { > #route(HANDLE_DMQ); > route(HANDLE_OPTIONS); > if (is_method("INVITE")) > { > xlog("L_INFO","MAIN - calling TO_CARRIER"); > route(TO_CARRIER); > exit; > } > else{ > sl_send_reply("401","UNAUTHORIZED"); > } > } > route[TO_CARRIER]{ > xlog("L_INFO","TO_CARRIER - calling RELAY_API"); > route(RELAY_API); #Route relay > xlog("L_INFO","TO_CARRIER - return"); > return; > } > # Relay request using the API (response) > route[RELAY_API_RESPONSE] { > xlog("L_INFO","RELAY_API_RESPONSE - got response from REST API"); > if ($http_ok==1 && $http_rs==200) > { > xlog("L_INFO","RELAY_API_RESPONSE - HTTP RESPONSE: > $http_rb\n"); > if (jansson_get("json", $http_rb, "$var(json)")) { > xlog("L_INFO","RELAY_API_RESPONSE - JSON = > $var(json)"); > $var(count) = 0; > jansson_array_size("routes", $var(json), > "$var(size)"); > xlog("L_INFO","RELAY_API_RESPONSE - > jansson_array_size"); > >> >> while ( $var(count) < $var(size) ){ > > >> jansson_get("routes[$var(count)].headers.to.uri", $var(rtjson), "$var(v)"); > > xlog("L_INFO","JSON - routes[$var(count)] >> - $var(v)"); > > #$(avp(mycontacts)[$var(count)]) = >> $avp(mycontacts) + $var(v) + "\r\n"; > > $avp(mycontacts) = $avp(mycontacts) + >> $var(v) + ";"; > > > *append_to_reply("Contact: <" + $var(v) >> +">"+ "\r\n"); /* ---- IS THERE A BETTER ?? ..... */* > > > $var(count) = $var(count) + 1; > > } > > > xlog("L_INFO","RELAY_API_RESPONSE - RELAY"); > xlog("L_INFO","MAIN - calling REPLY_302"); > route(REPLY_302); > return; > } > } > send_reply(500, "API Not Available - http response = $http_rs > $http_ok"); > exit; > } > route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $si:$sp from_number $fU > to_number $ru"); > $http_req(all) = $null; > $http_req(suspend) = 1; > $http_req(timeout) = HTTP_API_TIMEOUT; > $http_req(method) = "POST"; > $http_req(hdr) = "Content-Type: application/json"; > jansson_set("string","from_ip",$si, "$var(http_routing_query)"); > jansson_set("string","from_port",$sp, "$var(http_routing_query)"); > jansson_set("string","from_number",$fU, > "$var(http_routing_query)"); > jansson_set("string","to_number",$ru , "$var(http_routing_query)"); > > xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: > $var(http_routing_query)\n"); > $http_req(body) = $var(http_routing_query); > t_newtran(); > http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); > xlog("L_INFO","RELAY_API - Waiting for Response"); > } > route[REPLY_302] { > # Sends a 300 Multiple Choices back to the proxy that requested the > routing lookup > xlog("L_INFO","REPLY_302 - send reply"); > sl_send_reply("300", "Multiple Choices"); > xlog("L_INFO","REPLY_302 - exit"); > exit; > } > route[HANDLE_OPTIONS]{ > if(is_method("OPTIONS")) > { > sl_send_reply(200, "OK"); > exit; > } > if ($fU=="ping") > { > sl_send_reply("200","OK"); > exit; > } > } SNGrep output: 1 - INVITE from UAC to Kamailio INVITE sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > User-Agent: SIPp > Date: Mon, 08 Apr 2024 15:09:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 270 > v=0 > o=root 2021555890 2021555890 IN IP4 10.20.0.1 > s=SIPp > c=IN IP4 10.20.0.1 > t=0 0 > m=audio 18422 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv 2 - Reply from Kamailio to UAC SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 3 - SIP 300 Multiple Choices from Kamailio to UAC SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Contact: <sip:sip:918228990@10.20.0.2@sip.domain.io> > Contact: <sip:+351sip:918228990@10.20.0.2@10.20.0.3> > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 4 - SIP 500 from Kamailio to UAC SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM) > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=4eb2322b7d2b68e6fc3168f503344c21-32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 5 - ACK from UAC to Kamailio ACK sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 ACK > User-Agent: SIPp > Content-Length: 0 and continues with ACK and repeating SIP 500 above.... *Sérgio Charrua* .

2 months, 3 weeks

4
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Kamailio Admin Book email

by sadik.oualla.mohamed＠gmail.com

Hello, Last week, I submitted a request on the Asipto website to purchase the Kamailio Admin Book and received an automatic confirmation email. However, I have not received any further instructions on how to complete the purchase. I replied to the automatic email to follow up, but still, there has been no response. I need to get the book as soon as possible. Please note, I am writing from a different email address than the one I used to make the request on Asipto's site. If there are any issues with their contact email, or if further verification is needed, please let me know how to proceed securely without sharing personal details publicly on this list. Regards, Mohamed.

2 months, 3 weeks

2
2
1 0

customizing the ds_ping_interval dispatcher parameter for destination

by Giuseppe Pandolfi

Hi, I have one question about dispatcher module in kamailio V.5.3.1. Why the ds_ping_interval parameter is only global value? Is it possible introduce the customization value for single destination? Customization from destination is already possible for ds_ping_from parameter using special attributes in destination list file or db table, is it possible introduce the same management for ds_ping_interval? Thanks, Giuseppe ________________________________ NOTA BENE: Le informazioni contenute nel presente messaggio, compresi gli eventuali allegati, sono strettamente confidenziali ed indirizzate unicamente al destinatario indicato. Qualora non foste il destinatario Vi invitiamo a cancellare il messaggio, ed ogni eventuale documento allegato, non divulgare il contenuto a terze persone ed a volerci avvisare a mezzo posta elettronica. Grazie. A TLC S.r.l. tratta i vostri dati personali nel rispetto della normativa vigente in materia di Privacy (D.Lgs.196/2003 modificato dal D.Lgs.101/2018 e GDPR Regolamento UE n.679/2016). Per consultare l’Informativa completa la invitiamo a visitare il nostro sito web “www.aethra.com<http://www.aethra.com/>” PLEASE NOTE: This message and any attachments are confidential and may contain privileged information. If you are not the intended recipient, you are kindly requested to cancel it, not to disclose the contents to any other person and to notify the sender immediately by return e-mail. Thanks. A TLC S.r.l. collects your personal data in compliance with applicable data protection laws (D.Lgs. 196/2003 amended by Legislative Decree 101/018 and GDPR EU Regulation n. 679/2016). Please visit our website ”www.aethra.com<http://www.aethra.com>” to consult the complete Policy.

2 months, 3 weeks

1
0
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RFC: uptime metric for xhttp_prom module

by Ovidiu Sas

Hello all, The xhttp_module can export all stats under "stats.get_statistics" RPC command. I was thinking of adding an optional "uptime" stat that will return the kamailio server uptime (like the "core.uptime" RPC command). This will make it easier to add an uptime panel in grafana. The new stat would be controlled by a module parameter (by default disabled). Something like: modparam("xhttp_prom", "xhttp_prom_uptime_stat" 1) And this will generate something like: kam_uptime 671 1712249054631 This can be already implemented in the script using the "prom_gauge_set()" and retrieving the uptime via "jsonrpc_exec()", but it would be nicer and faster to implement it in the module itself. Comments? -ovidiu

2 months, 3 weeks

3
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Kamailio Visions 2030 for Appalachia

by Alex Balashov

For immediate release: HAZARD, Kentucky (1 April 2024)--Evariste Systems LLC, Georgia-based veteran brokers of international charity partnerships, are pleased to announce the conclusion of lengthy trilateral negotiations among Taylor Swift, the Public Investment Fund (the sovereign wealth fund of the Kingdom of Saudi Arabia), and the Kamailio open-source community to bring Kamailio to underserved areas of the US Appalachia region. In a concert in Lexington, Kentucky last Sunday evening, and in her signature style of mystical "reveals" through equivocal song lyrics, Swift auspiciously inaugurated to her Swifties, as die-hard fans of the pop star are known, the beginning of Kamailio Visions 2030. (Editorial note: Swift's agent has threatened copyright litigation in the event that our press agency were to reproduce her meaning-rich lyrics directly, driven by the contention that fair use considerations do not apply to lyrics so "deep" as Swift's. At press time, the matter is being reviewed by our counsel.) Alex Balashov, principal of Evariste, praised Kamailio Visions 2030 as a continuation of Swift's long-standing commitment to forward thinking and innovation: "Taylor has been the driving force behind the use of sustainable, plant-based materials in her 'merch. Today, aided by the internationalisation of Saudi capital, she is helping to bring that same kind of ethical prosperity to Eastern Kentucky, West Virginia and the wider Appalachian coal country." Continued Balashov: "Between Alan Maimon and J.D. Vance, and the revelations of the Purdue Pharma litigation, it's fair to say that the plight of Appalachia, with its high structural unemployment, declining economic base, and the ravages of the opioid epidemic and deaths of despair, has been exposed to a broader audience than ever before. And with that knowledge comes responsibility, because we live with that knowledge now, and we can no longer turn a blind eye. The SIP and VoIP industries have always been a force of conscientious capitalism, and Kamailio a beacon of rectitude in a jaded, amoral, cynical world. A world in which Appalachian children are denied the SIP proxy is not a just world, and we cannot any longer live in such a world." Fred Posner, a deputy Evariste negotiator who was said to have thrived in the pressure of high-stakes negotiations with PIF governor Yasir Al-Rumayyan over the exact implementational modalities of the deal, reviewed the high notes of the landmark economic development accord: "It was a good start to defund the police. But with the unique social challenges of Appalachia, what every struggling family in the ruins of coal country really needs is access to clean, ethically sourced, fair-trade SIP proxies. That is why I am uplifted to say that we will be dropping, through a coordinated, large-scale airlift operation, a Raspberry Pi with Kamailio pre-installed, to every household with an income at or below 250% of the Federal Poverty Level (FPL) in the Appalachian region. The Pi chassis will be dye-free, BPA-free, and will be held together with a plant-based, biodegradable, non-toxic glue that does not produce intoxicating effects when sniffed. By 2030, every child alive in Appalachia today will have had several years to benefit from the power of a miniaturised SIP proxy appliance, and that is why we are calling it Kamailio Visions 2030." When pressed by us, Balashov noted that the prosocial objectives and economic development potential of Kamailio Visions 2030 has drawn scepticism from some quarters. "These are those people who think that Rasperry Pi devices preloaded with Kamailio are not what Appalachia most needs today. Those people would be wrong. As any metaphysicist can tell you, causes do not necessarily resemble their effects." "Taylor and her people know that Kamailio Visions 2030 draws on a long history of highly effective technological and app-based interventions in economically underprivileged areas, such as Nicholas Negroponte's One Laptop per Child (OLPC) scheme, and the famous carpet-bombing of Eritrea with second-generation iPads by Selena Gomez. All of this had tremendous results." "Salutary results", added a representative of the PIF. Balashov pointed us to the foundational problem statement of Kamailio Visions 2030, which cited the concept of "Technik", as developed in "Hybrid Reality", a technocratic manifesto (in e-book form) by much-loved TED visionaries Parag and Ayesha Khanna. There, the authors declared: "Good Technik requires a combination of the attributes that deliver high human development, economic growth, political inclusiveness, and technology preparedness.” "In short", said Balashov, "Taylor, with some help from the efficient deployment of Saudi capital, will bring good technik to Appalachia. The tide of technik lifts all ships." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 months, 3 weeks

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low performance with no apparent reason

by Sergio Charrua

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 stateful with RTJson and simulating requests to routing engine and accounting - script #2 stateful but with just a simple routing to UAS, no rules, no DB, - script #3 stateless with a forward to UAS With script #3 I can go up to 2000CPS without issues with CPU at 37%! Above that value, I get retransmissions everywhere. On both scripts #1 and #2, the limit is 330CPS max after which I get a lot of retransmissions, while CPU/Core usage on Kamailio server stays below 10%. So I do not expect this to be a CPU issue. I could not understand why such (low) results, so I followed this article found at https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c <https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c> and created exact same scenarios, with kamailio script and SIPP templates available on the article, hoping for better results. But I get the same results: between 300 and 330CPS which is far, very far from the 7000CPS found in the article! I understand that I'm using VMs and probably the tests made for the article, which is pretty old already, were made on physical servers. Still, I would not expect 95% of lower performance! Any clue what could be the issue? I suspect NICs, but.... Any tips anyone could share? Thanks in advance! *Sérgio Charrua*

2 months, 3 weeks

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Statistics for all generated

by Alfredo Román Domínguez

I am using Kamailio versions 5.5.7 and 5.8.0 as a stateful proxy (with tm module) and I want to have metrics for all the SIP messages received and sent. I am using the statistics module for counting messages. I have in configuration file the following routes to capture events and count them: onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response, tm:local-request, sl:filtered-ack and network:msg) I have found the following problems that I need some hints how to solve them: 1) When a response is received to a request, I can count the response received in the reply_route but I don't find a way that works to count the reply being forwarded to the other side. onsend_route works fine for requests being forwarded. Is there any event_route or any trick that can be used to count the reply being forwarded similar to the onsend_route for requests? 2) When Kamailio receives a CANCEL it generates and sends automatically the 200 OK for the CANCEL and a 487 for the INVITE which started that dialog which is being cancelled, but I don't find a way that works to count those two responses Kamailio generates. 3) After having cancelled a dialog on one direction due to receipt of a CANCEL, kamailio generates local requests on the other direction, a CANCEL and an ACK also when it receives a 487 for the INVITE being cancelled. I don't find the way to count those. onsend_route does not work for locally generated CANCEL and ACK requests and the event_route[tm:local-request] only works for requests created outside of the tm module but not those created by the tm module itself (the CANCEL and the ACKs) since the tm module is not using its own function t_uac_prepare to create those requests (that function is the one that calls the event_route). Is there any statistic generated by the tm module or elsewhere or any other trick so I can count those requests? 4) When Kamailio generates retransmissions how can I count those? onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response) doesn't work for retransmissions. Are there statistics generated somewhere in the tm module for retransmissions? Is there any other way to count those. -- El contenido de este mensaje puede contener información confidencial sometida a secreto profesional. Si usted ha recibido este correo por error, no está permitida su distribución. Por favor elimine cualquier copia, archivo adjunto e infórmenos por esta vía. Los datos personales contenidos en este correo y que han sido facilitados por usted o provienen de fuentes de acceso público serán utilizados por Future Space, S.A., para gestionar nuestra relación contractual o pre-contractual, resolver su solicitud y mantenerle informado sobre los productos o servicios solicitados o similares. Se conservarán mientras exista un interés mutuo para ello, no se oponga al tratamiento o solicite su supresión. Los datos no serán comunicados a terceros, salvo obligación legal. Puede ejercitar sus derechos de acceso, rectificación, supresión, limitación del tratamiento, portabilidad de los datos y oposición mediante escrito dirigido a Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid o mediante correo electrónico a dpo(a)futurespace.com <mailto:dpo@futurespace.com>. Si consideras que se han infringido tus derechos puedes presentar una reclamación ante la www.agpd.es <http://www.agpd.es>. The contents of this electronic mail may be confidential or privileged. If you have received this message by mistake, please do not send it to anyone. We ask you to delete any existing copies, attached files and to notify us accordingly. The personal data provided or obtained from publicly accessible resources will be processed by Future Space, S.A. in order to manage our contractual or pre-contractual relationship, solve your request and keep you informed about activities or products requested by you or similar. Personal data will be stored while a mutual interest exist or either erasure or objection to process them is requested. Data will not be disclosed to third parties unless there is a legal obligation to do so. You have the rights to access, rectify, erasure, restriction of processing, data portability and oppose any processing of your personal data held by Future Space, S.A. To make effective use of your rights, please contact us at Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid or by email at dpo(a)futurespace.com <mailto:dpo@futurespace.com>. If you consider processing of your personal data infringes data protection regulations you can file a claim with www.agpd.es <http://www.agpd.es>

2 months, 3 weeks

1
0
0 0

TLS module doesn't support TLSV1

by o.atef＠fiberme.com

Hello Brothers, I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux). Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said: # We do not enable anything else than TLSv1.2+ # over the public internet. Clients do not have # to present client certificates by default. How could I avoid this restriction please to enable TLSv1? Thank you,

2 months, 3 weeks

4
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Kamailio v5.8.1 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.8.1 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.1. For more details about version 5.8.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/04/kamailio-v5-8-1-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 Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

2 months, 4 weeks

1
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Releasing v5.8.1

by Daniel-Constantin Mierla

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

2 months, 4 weeks

1
1
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start_recording' or 'record-call on' is not invoking

by pawan srivastava

Hi I am testing rtpengine recording with kamailio in lab. I have 1 endpoint registered to Kamailio. My kamailio and rtpengine are running on same box. I have another kamailio2 instance with 2nd endpoint to make a call to 1st endpoint. I have observed that Response dump for 'offer'/'answer' but further start recording didn't invoke. root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all callid: J0WtgYIKd0Kwj1iL6BiFOg.. | deletionmark: no | created: 1712120753 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: uH_ozKj18Nafshsm_sTElQ.. | deletionmark: no | created: 1712120730 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: VYIw791dJska8PiIFspy8w.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:54804 | redis_keyspace:0 | foreign:no callid: nP89gWwjhsg_TSZrqu16nQ.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:56207 | redis_keyspace:0 | foreign:no root@ip-kamailio:/etc/kamailio# I am new to kamailio and missing configuration. Suggest me the correct configuration to invoke recording. attached config and logs Regards, Pawan

2 months, 4 weeks

1
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How to relay SIP messages to specific registered user in Kamailio?

by o.atef＠fiberme.com

Hello guys, I'm recently new player in Kamailio SIP server and want to use it as SIP proxy server to Asterisk PBX server. first of all, I've successfully registered my Asterisk to Kamailio as user "test" to get its "location" because my Asterisk is behind NAT. Simply now I want to relay SIP messages to this user's location to receive them all in Asterisk. After I searched, I found this function rewritehost() but it needs to be a host or IP address, but in my case I want some way or function to relay to user "test" and then it well get the user location from Kamailio datebase automatically? I hope it's clear XD I tried to find a way to do MSQL query in kamalio.cfg to get the user contact from "location" table and then relay to the result but it did not work. I guess that I did it in wrong way but finally the Kamailio tutorials are so rare on google. Thanks in advance, Omar

2 months, 4 weeks

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Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?

by Inextrix Technologies Pvt Ltd

This question digs into the heart of Kamailio's functionality. It asks about: Routing Logic: How Kamailio decides where to send SIP messages (calls, registrations, etc.) based on various factors like caller ID, destination number, or custom rules. Customization: How you can modify this built-in logic to fit your specific needs. This could involve using scripting languages, manipulating SIP messages, or defining custom routing rules. The discussion would be for developers who want to unlock Kamailio's full potential and tailor it for their unique VoIP deployments. Visit us for More Info : https://inextrix.com/services/kamailio-development/

3 months

2
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Callee details not visible in dialog

by egemen ulus

Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. I have also tried to update to header before dlg_manage function, it didn't help either. Any way to have asterisk IP in dialog details. It shows only this: { h_entry: 483 h_id: 8644 ref: 1 call-id: v489LdfOFD5mnlHupQHOvw.. from_uri: sip:1515@12.12.12.12;transport=UDP to_uri: sip:2222@12.12.12.12 state: 5 start_ts: 0 init_ts: xxxx end_ts: xxxx duration: xxxx timeout: 0 lifetime: 3600 dflags: 512 sflags: 0 iflags: 0 caller: { tag: d0283007 contact: sip:1515@1.1.1.1:64966;transport=UDP cseq: 1 route_set: socket: udp:172.22.10.10:5060 } callee: { tag: <null string> contact: <null string> cseq: <null string> route_set: <null string> socket: <null string> } profiles: { } variables: { } } Regards Get Outlook for Android<https://aka.ms/AAb9ysg>

3 months

1
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[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE

by Denys Pozniak

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 5.1.2 (x86_64/linux) -- BR, Denys Pozniak

3 months

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RFC: move app_lua_sr to kamailio-archive

by Victor Seva

Hi, I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua Cheers -- ----------------------------------------------------------------- | ,''`. Victor Seva | | : :' : linuxmaniac(a)torreviejawireless.org | | `. `' PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A 5068 | | `- Debian Developer | -----------------------------------------------------------------

3 months

1
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Dialog and DMQ on restart

by Björn Klasen

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

3 months, 1 week

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RTP Engine with Kamailio

by Vishal Pai

Hello Everyone. I am looking for documentation to have multiple RTP Engine servers connected with kamailio using db url and balance the load accordingly. Thanks

3 months, 1 week

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Kamailio to aid in seamless data center migration?

by Brad

I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center. I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime. Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests. My current train of thought is to put a clone the VMs in the new data center and use Kamailio to route the SIP traffic to the new servers/IPs. I’ve never used it before, so I may be barking up the wrong tree, but it /appears/ to do what I’m suggesting. If so, I’m thinking I can install Kamailio on each VM, point it to the local asterisk/FreePBX initially, and clone the VM. Then, after the new instance is up and running, point Kamailio on the original VM to the cloned VM's asterisk, after which I can make appropriate DNS changes. Another option would be to stand up a single Kamailio server and redirect SIP traffic destined for individual asterisk servers to it at the router. Endpoints are mostly Yealink, and I’m not sure if they’ll feel the need to restart when the registration/SIP server’s IP changes, but a quick bounce when not in use isn’t the end of the world. I’d very much like to avoid having to send an update to each phone manually, but I can script a SIP NOTIFY if required. Anything stupid, wrong, ignorant, or just smelly about this tactic? Or, for that matter, any other suggestions?

3 months, 1 week

2
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