- sr-users - kamailio.org

TO Header corrupted in RE-INVITE
by alirsm 07 Feb '21

07 Feb '21

Hi, Call to a number forwarded to a FAX by using two Kamailio proxies. (INVITE) --> KamailioA --> GW1 --> KamailioB --> GW2 --> FAX FAX --> (RE-INVITE) --> GW2 --> KamailioB --> (TO Header being corrupted) KamailioB is corrupting TO header when receiving RE-INVITE and causing the call to fail. 1) Only one "uac_replace_from()" is being used in each Kamailio. 2) GW1 is adding +1 (using E164 format) to URI and FROM headers. I was wondering if any of the above might cause the TO header corruption. Any help would be appreciated. AL

2 1

Failover with Dispatcher Module
by alirsm 05 Feb '21

05 Feb '21

Hi I have setup dispatcher failover on Kamailio 5.1 as below. If destination X.X.X.X is down, traffic should be routed to Y.Y.Y.Y. The failover is working but every time after 30 sec Timeout. Is there a way to setup health check properly so the failed destination can be ignored while it is down? Your help is greatly appreciated, AL # dispatcher.list 1 sip:X.X.X.X:5060 0 0 1 sip:Y.Y.Y.Y:5060 0 0 # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_inactive_threshold", 4) modparam("dispatcher", "ds_ping_latency_stats", 1) modparam("dispatcher", "ds_probing_threshold", 1) # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "4")) { send_reply("503", "No destination"); exit; } t_on_failure("DISPATCH_FAILOVER"); route(RELAY); exit; } # Failure Route failure_route[DISPATCH_FAILOVER]{ if (t_is_canceled()) exit; if(!ds_next_dst()) { send_reply("503", "Out of gateways"); exit; } t_on_failure("DISPATCHER_FAILOVER"); t_relay(); }

2 1

Make configuration as proxy, push, renew server
by James Chan 02 Feb '21

02 Feb '21

Hi We use a Cloud-based SIP-server that does not support the Push-Notifications Service. We try to use Kamailio as the front-end to work as a proxy and push-server. We need help configuring Kamailio as a proxy and push-server. In addition, we want to implement auto renew configuration as UAC using Kamailio. Because all clients need to renew registration our Cloud-SIP-server every 59 minutes. ( registration expires after 60 minutes) Summary: UAC -> Kamailio (proxy & push & auto renew) -> Cloud-SIP-Server -> UAC I have attached our config file and screens And if phone has multiple contacts, there is call bug, call will be drop since there are two incoming call #!KAMAILIO #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ACCDB #james #!define WITH_UAC #!define WITH_TSILO # # Kamailio (OpenSER) SIP Server v5.3 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!ifdef WITH_TSILO #!define FLB_TSILO 8 #!endif ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif fork=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:185.224.139.45:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_UAC loadmodule "uac.so" #!endif #!ifdef WITH_TSILO loadmodule "tsilo.so" loadmodule "htable.so" loadmodule "cfgutils.so" loadmodule "http_client.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_UAC modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 1) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif #!ifdef WITH_UAC modparam("uac", "reg_contact_addr", "pn.telesonique.ch:5060;transport=udp") modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_db_table", "uacreg") #!endif #!ifdef WITH_TSILO modparam("htable", "htable", "vtp=>size=10;autoexpire=180") modparam("http_client", "httpcon", "pushapi=>https://www.myswissline.com") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (is_method("REGISTER")) { save("location", "0x06"); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } #james set_forward_no_connect(); if (!t_relay()) { sl_reply_error(); } #!ifdef WITH_TSILO else if(isflagset(FLB_TSILO)) { resetflag(FLB_TSILO); if (ts_store()) { $sht(tsilo=>stored::$rU) = 1; xdbg("stored transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route(PUSH_NOTIF); } #!endif exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies #set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } save("location", "0x06"); switch($rc) { case -1: case -2: sl_reply_error(); break; #!ifdef WITH_TSILO case 1: $var(new_contact) = 1; #!endif } #!ifdef WITH_TSILO if ($var(new_contact)) route(WAKE_SUSPENDED); #!endif exit; } #!ifdef WITH_TSILO route[WAKE_SUSPENDED] { $var(suspended) = 0; lock("$tU"); $var(suspended) = $sht(tsilo=>suspended::$tU); $sht(tsilo=>suspended::$tU) = $null; $var(stored) = $sht(tsilo=>stored::$tU); unlock("$tU"); if ($var(suspended) == 0) { if ($var(stored)) { # no suspended INVITE (i.e. there was at least one # registered contact for target) and 'stored' set # (i.e. ongoing transaction) -> this newly just # registered contact needs alerting. ts_append("location", "$tu"); } return; } $var(id_index) = $(var(suspended){s.select,0,:}{s.int}); $var(id_label) = $(var(suspended){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU .\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); } route[RESUME] { if(! lookup("location")) { xlog("failed to load just added destinations\n"); exit; } setflag(FLB_TSILO); route(RELAY); } #!endif # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; #!ifdef WITH_TSILO send_reply("100", "Trying"); route(SUSPEND); route(PUSH_NOTIF); #!else route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } #!endif } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } #!ifdef WITH_TSILO setflag(FLB_TSILO); #!endif route(RELAY); exit; } #!ifdef WITH_TSILO route[SUSPEND] { if(! t_suspend()) { xlog("SUSPEND: failed to suspend transaction.\n"); # TODO: actually, the destination is not really checked: don't know if it's # own subscriber or invalid dialing in the first place. send_reply("503", "Temporarily unavailable"); exit; } $sht(tsilo=>suspended::$rU) = $_s($T(id_index):$T(id_label)); xdbg("suspended transaction [$T(id_index):$T(id_label)] $fU => $rU\n"); } route[PUSH_NOTIF] { xdbg("TODO: send push for $rU\n"); $var(res) = http_connect("pushapi", "/send_push.php?username=$tU&dev=1", "application/json", "", "$avp(gurka)"); xlog("L_INFO", "Push notification: $avp(gurka) Result code $var(res)\n"); if ($var(res)==200) { if ($var(gurka)=~"^.*rejected.*$") { sl_send_reply("503", "Not available right now"); } else { sl_send_reply("101", "Push Sent"); } } if ($var(res)==28) { xlog("L_INFO", "Push notification timed out. Do we care?"); } exit; } #!endif # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_TSILO return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } [image: call_issue.png][image: register.png]

1 0

failover on SIP servers
by AL RSM 02 Feb '21

02 Feb '21

Hi, I was wondering what is the recommended approach to implement a failover on several destination SIP servers: 1) using DNS SRV records 2) using dispatcher list 3) combination of both. Thanks, AL

3 2

Acc module unable to insert cdr in mysql server
by M Arqum CH 01 Feb '21

01 Feb '21

Hello Members, Kamailio acc module is not inserting cdr into MySQL DB, sometimes it works and sometimes didn't, the number of parameters/values are similar in failed as well success scenario. I also check with tcpdump for MySQL requests, unfortunately, Kamailio didn't issue insert query toward DB server, so it is clear to me that the MySQL server is nothing to do with failed queries. Please guide me on what I should do to find and fix the issue. Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:691]: tmcb_func(): acc callback called for t(0x7fe29b2a7870) event type 512, reply code 200 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:413]: should_acc_reply(): probing acc state - code: 200 flags: 0x6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_logic.c:446]: should_acc_reply(): acc is on Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 6 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 7 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 10 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 13 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 16 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 17 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 19 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 22 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 26 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 28 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 29 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 32 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 6, index i 33 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_hooks.c:256]: run_trans_callbacks_internal(): DBG: trans=0x7fe29b2a7870, callback type 512, id 0 entered Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:267]: db_write_cdr(): failed to insert into database Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 0 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 1 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 2 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 3 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 4 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 5 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 8 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 9 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 11 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 12 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 14 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 18 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 20 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 21 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 24 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 3, index i 25 Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: ERROR: acc [acc_cdr.c:682]: cdr_on_end_confirmed(): failed to write cdr! Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103637]: DEBUG: tm [t_reply.c:1758]: cleanup_uac_timers(): RETR/FR timers reset Jan 28 13:38:29 /root/mykamailio/sbin/kamailio[103682]: DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7fe29727ba88 (p:0x7fe294d27288/n:0x7fe294d27288) -- Regards M Arqum

2 2

UACREG to many Asterisk
by Javier Valencia 01 Feb '21

01 Feb '21

Hi there! Go ahead I apologize for my English. I have a Kamailio 5.4 with the DISPATCHER module and UAC enabled (for remote REGISTER), the DISPATCHER has several SETIDs, and they have one or more Asterisk balanced behind it. In Kamalio script, when an endpoint does REGISTER, I enable the REGISTER of the UAC module. And when it expires or expires=0 I disable the REGISTER in the UAC module. In the UACREG table in the database, L_UUID (unique key), L_USERNAME, R_USERNAME, and AUTH_USERNAME have the same value. For example CST100EXT100. Knowing that the DISPATCHER and UAC modules have no relationship, but that when a SETID has several balanced Asterisks, how can I make the UAC module send the REGISTER to all the Asterisks of a SETID? Clarifications; 1.- I cannot update the Asterisk, which are currently in version 11. They do not support PATH. 2.- I can modify the content of the UACREG table so that l_uuid has a suffix '_X'. 3.- Call jsonrpc_exec several times to enable the REGISTER in the UAC module. (diff emulated) < jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_1"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_2"]}') ; > jsonrpc_exec ('{"id": 1, "jsonrpc": "2.0", "method": "uac.reg_enable", "params": ["l_uuid", "s: $ au_3"]}') ; But I want to know if there is any more elegant way to do this. Cheers, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

externally triggered message sending to specific clients
by Syed Shahryar 01 Feb '21

01 Feb '21

Is it possible to send arbitrary text messages triggered by an external application (evapi/http) to registered SIP clients via websocket? The following doesn't seem to do anything. KSR.corex.send_data("sip:user@example.com", "Message at $Ts"); -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

about kamailio-sbc-teams tutorial
by Juha Heinanen 01 Feb '21

01 Feb '21

I read kamailio-sbc-teams tutorial https://skalatan.de/blog/kamailio-sbc-teams and it does not mention, which CA root certificates need to be included tls config ca_list so that Kamalio is able to verify sipX.pstnhub.microsoft.com server certificates. -- Juha

2 1

logging error events to database
by Sergiu Pojoga 30 Jan '21

30 Jan '21

Hi there, I was wondering if there's a way to log various core error events in some DB. For example, below error occurred Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp_helpr_funcs.c:499]: extract_mediaip(): no `IP[4|6]' address in `c=' field Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE 28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core> [core/parser/sdp/sdp.c:430]: parse_sdp_session(): can't extract common media IP from the message Thanks in advance. Regards, --Sergiu

4 5

Establish connection between MsTeams and Kamailio.
by Willy Valles Rios 30 Jan '21

30 Jan '21

Hello community, Please your support, I tell you that I am integrating Kamailio with MsTeams, after some time reviewing this, I finally achieved the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

2 1

registering gsm gateway by siremis
by shashank shekhar 29 Jan '21

29 Jan '21

Hi, I want to register a gsm gateway on my kamailio server with siremis ...kindly guide me on this Thankyou Regards Shashank shekhar 9162735161

2 1

Kamailio as inbound and outbound proxy
by Michael Friesen 29 Jan '21

29 Jan '21

Hello, I am trying to configure Kamailio as both an inbound and outbound proxy. Inbound requests flow Sip Trunk -> Kamailio -> Asterisk. Outbound requests flow Asterisk -> Kamailio -> Sip Trunk. Inbound traffic is sent to port 5060 on kamailio which listens on the PRIVATE_IP:5060 and advertises PUBLIC_IP:5060. How can we ensure that when messages are sent to the Sip Trunk they have Record-Route headers with the PUBLIC_IP and when messages are sent to asterisk they have Record-Route headers with the PRIVATE_IP?

2 1

kamailio 5.4.3 ubuntu 20.04 tls - http_async_client
by Filippo Graziola 28 Jan '21

28 Jan '21

Hi all, I have an issue related (my guess) to tls and http_async_client module that result in a segmentation fault and a not correct handle of tls connections. First with only tls module loaded, not forked: 0(1021) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1021) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1021) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1021) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1021) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1021) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 4718592 and 2359296 bytes 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 4718592 0(1021) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2359296 0(1021) INFO: <core> [main.c:2833]: main(): processes (at least): 9 - shm size: 67108864 - pkg size: 67108864 0(1021) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1021) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1021) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1021) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1021) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1021) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1021) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1021) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1021) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1021) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1021) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1021) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1021) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1021) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1021) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 6(1027) ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error 6(1027) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:141FC044:SSL routines:tls_setup_handshake:internal error 6(1027) ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: XXXXXXXXXXXXXXX 6(1027) ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: XXXXXXXXXX 6(1027) ERROR: <core> [core/tcp_read.c:1498]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2cbc1b3948 r: 0x7f2cbc1b3a70 (-1) so no segmentation fault but error in handling. Second one also with http_async_client loaded: 0(1059) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1061) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1061) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1061) INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! 0(1061) INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman 0(1061) INFO: http_async_client [http_async_client_mod.c:222]: mod_init(): Initializing Http Async module 0(1061) WARNING: tls [tls_init.c:784]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(1061) INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(1061) INFO: <core> [main.c:2833]: main(): processes (at least): 10 - shm size: 67108864 - pkg size: 67108864 0(1061) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1061) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/fullchain.pem' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/privkey.pem' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 0(1061) NOTICE: tls [tls_domain.c:1105]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... 0(1061) INFO: tls [tls_domain.c:711]: set_verification(): TLSs<default>: No client certificate required and no checks performed 0(1061) INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 0(1061) INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' 0(1061) INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' 0(1061) INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' 0(1061) INFO: tls [tls_domain.c:334]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 0(1061) INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' 0(1061) INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' 0(1061) INFO: tls [tls_domain.c:352]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 0(1061) INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 0(1061) INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc<default>: verify_client=0 0(1061) INFO: tls [tls_domain.c:714]: set_verification(): TLSc<default>: Server MAY present invalid certificate 0(1061) INFO: http_async_client [async_http.c:101]: async_http_init_sockets(): inter-process event notification sockets initialized 0(1061) INFO: http_async_client [async_http.c:84]: async_http_init_worker(): started worker process: 1 0(1059) CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x1 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring Segmentation fault this time, there is a segmentation fault. The above is a result of this minimal configuration: #!KAMAILIO ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=2 tcp_children=2 auto_aliases=no alias="XXXXXXXXXXXXX" listen=udp:eth0 server_signature=no tcp_connection_lifetime=3605 tcp_max_connections=40960 tcp_accept_no_cl=yes enable_tls=yes listen=tls:XXXXXXXXXX:5061 advertise XXXXXXXXXXXX:5061 tls_max_connections=40000 enable_sctp=no ####### Modules Section ######## loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "tls.so" loadmodule "http_async_client.so" #----------------- setting module-specific parameters --------------- #----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #----- http client ---- modparam("http_async_client", "workers", 1) ####### Routing Logic ######## request_route { exit; } I used the above configuration to take out as much as possible my mistakes in the configuration, but with my full kamailio configuration, tls connections give the above errors but everything else works just fine (also http_async_client module functions which are used on INVITES) and calls are going properly (unfortunately tls is required). I found a couple of issues that are similar https://github.com/kamailio/kamailio/issues/2560 and https://github.com/kamailio/kamailio/issues/2466# but as far as I understood the issue 2466 is closed because fixes are already included. I tried in any case to compile from source a few older releases with the same result, changed also the certificate and private key (letsencrypt), moreover I have another kamailio (v5.3.4) running on ubuntu 18.04 (same configuration) without any issues. I saw that there is a different version of openssl version 1.0.. in ubuntu 18.04, version 1.1 in ubuntu 20.04, but the segmentation fault seems to happen after an error on free some memory. Have you some ideas? tell me if you need more info from me. Thanks Filippo

4 7

Problems establishing SIP signaling between MsTeams and Kamailio
by Willy Valles Rios 28 Jan '21

28 Jan '21

Hello community, I am having trouble establishing SIP signaling between MsTeams and Kamailio. I currently have this configuration in my tls.cfg file [server: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem [client: default] method = TLSv1.2 + verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certificates/private-key.pem certificate = /etc/kamailio/certificates/certificate.pem My domain was certified with ssl through an authoritative certifier (GoDaddy), however I see these errors in the / var / log / messages of the Kamailio server. Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:389]: mod_init(): With ECDH-Support! Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_mod.c:392]: mod_init(): With Diffie Hellman Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, compression: on Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, zlib compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM -DECP_NISTZ256_ASM Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 13107200 and 6553600 bytes Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 13107200 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6553600 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - pkg size: 4194304 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='' ... Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: certificate='/etc/kamailio/certificados/certificate.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: private_key='/etc/kamailio/certificados/private-key.pem' Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid certificate Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f45242be028 r: 0x7f45242be150 (-1) Could you help me identify the problem please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com

9 18

Info: Upcoming online events - February 2021
by Daniel-Constantin Mierla 27 Jan '21

27 Jan '21

Hello, short note to announce a news article listing the upcoming online events where you can watch presentation related to Kamailio, respectively FOSDEM and ClueCon TGI. More details at: * https://www.kamailio.org/w/2021/01/upcoming-online-events-february-2021/ During this time where the travelling is under heavy restrictions, such events can be a chance to keep in touch with peers in the RTC space or learn new usefuk things. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Loop with unable to find dialog for xxx
by David VILLAUME 27 Jan '21

27 Jan '21

Hello, I got some time Ack packets looping on my servers during 4 hours if I don’t restart the system ( 4 hours is my dialog timeout). Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020' It generates approximatively 2 millions packets per loop. On the last occurrence I noticed a ‘SIP/2.0 200 canceling’ generated per Kamailio before the loop started (which seems uncomon on this host) Bellow is a screenshot of the beginning of the loop and may dialog and topos config. Has anyone encountered such behavour ? Regards, David [cid:image001.png@01D6F4E2.F68B5140] modparam("dialog", "dlg_flag", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "default_timeout", 14500) modparam("dialog", "event_callback", "ksr_dialog_event") modparam("dialog", "db_url", DBURL_DLG) modparam("dialog", "db_update_period", 15) modparam("dialog", "db_mode", 1) modparam("topos", "storage", "redis") modparam("topos", "dialog_expire", 14500) modparam("topos", "contact_host", "PUBLICIP") modparam("topos_redis", "serverid", "topos")

1 0

websocket rd buffer size limit?
by Andrew Chen 27 Jan '21

27 Jan '21

Hey group, I was wondering if there is a way to control the websocket buffer size to support large SIP messages. I received the following error when sending content-length of 65000 2021-01-26T17:32:19.251408+00:00 sjomainkama50 kamailio[31871]: 109(32008) WARNING: websocket [*ws_frame.c*:482]: decode_and_validate_ws_frame(): message is too long for our buffer size (65535 / 66151) I am assuming this is controlled by the tcp_rd_buf params which I have set to 131k: *tcp_*rd_buf_size=131072 and this is my tcp_wr_buf params: *tcp_*wq_blk_size=131072 *tcp_*conn_wq_max=131072 Please advise. Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 4

event_route[core:receive-parse-error] missing from docs
by Henning Westerholt 27 Jan '21

27 Jan '21

Hello, I noticed that the core "event_route[core:receive-parse-error]" was missing from the cookbook, so I have added it there. https://www.kamailio.org/wiki/cookbooks/devel/core#event_route Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

FQDN endpoint as rd
by AL RSM 27 Jan '21

27 Jan '21

Hi, I was wondering if Kamailio can support FQDN endpoint which resolves to multiple IPs, checks if the first IP is not reachable, try the next one in the list. Thanks, AL

2 1

Re: [SR-Users] sr-users Digest, Vol 188, Issue 9
by Willy Valles Rios 26 Jan '21

26 Jan '21

Hello community, After a while, I finally got the connection from my Kamailio server to MsTeams and I can validate that the connection to MsTeams is in AP. [root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI | FLAGS" URI: sip: sip.pstnhub.microsoft.com; transport = tls FLAGS: AP However, from the admin panel of MsTeams (Direct Routing) I see that the connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status in "Active" but the SIP options status is in "Warning". I have made calls from MsTeams thinking that the SIP options status would change to "active" but it is still in "Warning" state. On the other hand, I have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from kamailio are being sent in the following format to MsTeams. OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0 Via: SIP / 2.0 / TLS 161.35.44.66:5061 ;branch=z9hG4bKea07.52224687000000000000000000000000.0 To: <sip: sip.pstnhub.microsoft.com; transport = tls> From: <sip: sbc.netvoiceperu.com>; tag = d3569c818b500aeb8c373426e76c2884-81763c71 CSeq: 10 OPTIONS Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.0 (x86_64 / linux)) As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not contain the "Contact" field, which in theory said "Contact" field should have been added by Kamailio according to the configuration added in kamailio.cfg event_route [tm: local-request] { sip_trace (); if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; transport = tls> \ r \ n"); } xlog ("L_INFO", "Sent out tm request: $ mb \ n"); } As additional information, I inform you that I also managed to observe the SIP OPTIONS that MsTeams sends to Kamailio. OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0 FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag = f1bdeb5f-662f-4544-a436-e9aa9ad78da4 TO: <sip: sbc.netvoiceperu.com> CSEQ: 1 OPTIONS CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a MAX-FORWARDS: 70 VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10 ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY However I don't see the 200 OK SIP responses from Kamailio to MsTeams. I think this may be the reason why I see the SIP OPTIONS status in "Warning" from the MsTeams panel. Maybe the contact field is not being added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for that reason I don't see 200OK responses from MsTeams. Could you help me solve this please. Cheers Saludos Cordiales -- *Willy Valles Rios* *Unified Communications Specialist* phone: +51955747343 em@il: willyvalles17(a)gmail.com El dom, 10 ene 2021 a las 6:03, <sr-users-request(a)lists.kamailio.org> escribió: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Carlos Mestanza T.) > 2. Re: Problems establishing SIP signaling between MsTeams and > Kamailio (Rob van den Bulk) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 8 Jan 2021 19:16:30 -0500 > From: "Carlos Mestanza T." <mestacart(a)gmail.com> > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, "Daniel Constantin Mierla > (asipto)" > <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems establishing SIP signaling between > MsTeams and Kamailio > Message-ID: > < > CAHcXM7PN0zuJC9jONTERZdfur3b+ucwnCC712Ss1OBxCiMctUg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Dear Kamilio Community, > > After doing several reads to our Kamailio configuration, the dispatcher > list is in AP: > > URI: sip: sip.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip2.pstnhub.microsoft.com; transport = tls > FLAGS: AP > URI: sip: sip3.pstnhub.microsoft.com; transport = tls > FLAGS: AP > > But in the MS Teams dashboard the SIP OPTIONS STATUS column is WARNING. > > In the LOGs sent to OPTIONS, it gives us to understand that with sent to MS > TEAMS: > > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKd25f.2835f676000000000000000000000000.0#015#012To: > <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-213e3c71#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d0-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d0-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip2.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKe25f.b14dc514000000000000000000000000.0#015#012To: > <sip:sip2.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-44c3af70#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d1-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d1-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:40 Kamailio-Server /usr/sbin/kamailio[1444]: INFO: <script>: > Sent out tm request: OPTIONS sip:sip3.pstnhub.microsoft.com;transport=tls > SIP/2.0#015#012Via: SIP/2.0/TLS > 161.35.44.66:5061 > ;branch=z9hG4bKb25f.8442f914000000000000000000000000.0#015#012To: > <sip:sip3.pstnhub.microsoft.com;transport=tls>#015#012From: <sip: > sbc.netvoiceperu.com > >;tag=69ae0da9200ed8d142f2e4a69f531080-fa555adb#015#012CSeq: > 10 OPTIONS#015#012Call-ID: > 07561978687e60d2-1444(a)10.131.245.99#015#012Max-Forwards > <http://07561978687e60d2-1444@10.131.245.99#015%23012Max-Forwards>: > 70#015#012Content-Length: 0#015#012User-Agent: kamailio (5.4.0 > (x86_64/linux)) > Jan 8 19:01:42 Kamailio-Server journal: Suppressed 103 messages from > /system.slice/kamailio.service > > In the kamailio.cfg configuration it is declared: > > listen=tls:161.35.44.66:5061 > listen=tcp:10.131.245.99:5061 > > modparam ("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") > modparam ("dispatcher", "ds_probing_mode", 1) > modparam ("dispatcher", "ds_ping_interval", 60) > > And this fragment was also added: > > event_route [tm: local-request] { > > if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") { > append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061; > transport = tls> \ r \ n"); > } > xlog ("L_INFO", "Sent out tm request: $ mb \ n"); > } > > There is something additional that has to be declared so that in the MS > Teams panel the SIP OPTIONS STATUS column is shown as ACTIVE and not as > WARNNING, in the MS TEAMS documentation it is a possible problem related to > OPTIONS events. > > Atentamente > > *Adalberto Carlos Mestanza T.* > > > > El jue, 7 ene 2021 a las 21:54, Ovidiu Sas (<osas(a)voipembedded.com>) > escribi?: > > > That certificate should already be present under the OS's trusted > > certificates directory (debian and ubuntu certs are stored under > > /etc/ssl/certs), maybe under a different name, and is required for > > remote endpoint's certificate validation. > > One can load a particular certificate or a list of certificates. > > Multiple certificates can be concatenated into one single file as > > stated in the documentation: > > https://kamailio.org/docs/modules/devel/modules/tls.html#tls.p.ca_list > > Hope this helps a little bit in understanding of the ca_list param. > > > > Regards, > > Ovidiu Sas > > > > On Thu, Jan 7, 2021 at 8:10 AM <rob.van.den.bulk(a)gmail.com> wrote: > > > > > > I Used this tls.cfg > > > > > > > > > > > > Use bc2025.pem as extra, Microsoft needs this? > > > > > > > > > > > > And works fine on different Kamailio-msteams sbcs > > > > > > > > > > > > > > > > > > [server:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > server_name = sbc.combivoipdom.nl > > > > > > > > > > > > [client:default] > > > > > > method = TLSv1.2+ > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/privkey.pem > > > > > > certificate = > > /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/fullchain.pem > > > > > > ca_list = /etc/letsencrypt/live/sbc.combivoipdom.nl-0001/bc2025.pem > > > > > > > > > > > > > > > > > > > > > > > > Cheers Rob > > > > > > > > > > > > Van: sr-users <sr-users-bounces(a)lists.kamailio.org> Namens > > Daniel-Constantin Mierla > > > Verzonden: donderdag 7 januari 2021 08:53 > > > Aan: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org > >; > > Willy Valles Rios <willyvalles17(a)gmail.com> > > > CC: Carlos Mestanza T. <mestacart(a)gmail.com> > > > Onderwerp: Re: [SR-Users] Problems establishing SIP signaling between > > MsTeams and Kamailio > > > > > > > > > > > > Does this happen when Kamailio connects to MS Teams? The logs indicate > > the received TLS certificate is not trusted: > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > > > > > > > You can set debug=3 in kamailio.cfg and see if the DEBUG messages > > provide more hints. For me it worked fine with Letsencrypt certs in > > Kamailio and accepting what ever MS sent back. I used Debian 10 and > libssl > > 1.1. > > > > > > > > > > > > Cheers, > > > Daniel > > > > > > > > > > > > On 06.01.21 21:47, Willy Valles Rios wrote: > > > > > > Hello community, > > > > > > > > > > > > I am having trouble establishing SIP signaling between MsTeams and > > Kamailio. I currently have this configuration in my tls.cfg file > > > > > > > > > > > > [server: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > [client: default] > > > > > > method = TLSv1.2 + > > > > > > verify_certificate = yes > > > > > > require_certificate = yes > > > > > > private_key = /etc/kamailio/certificates/private-key.pem > > > > > > certificate = /etc/kamailio/certificates/certificate.pem > > > > > > > > > > > > My domain was certified with ssl through an authoritative certifier > > (GoDaddy), however I see these errors in the / var / log / messages of > the > > Kamailio server. > > > > > > > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:389]: mod_init(): With ECDH-Support! > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_mod.c:392]: mod_init(): With Diffie Hellman > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:722]: tls_h_mod_init_f(): compiled with openssl version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > > compression: on > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_init.c:730]: tls_h_mod_init_f(): installed openssl library version > > "OpenSSL 1.0.2k-fips 26 Jan 2017" (0x100020bf), kerberos support: on, > zlib > > compression: on#012 compiler: gcc -I. -I.. -I../include -fPIC > -DOPENSSL_PIC > > -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H > -DKRB5_MIT > > -m64 -DL_ENDIAN -Wall -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > > -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 > > -grecord-gcc-switches -m64 -mtune=generic -Wa,--noexecstack -DPURIFY > > -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 > > -DOPENSSL_BN_ASM_GF2m -DRC4_ASM -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM > > -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM > > -DECP_NISTZ256_ASM > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: WARNING: tls > > [tls_init.c:787]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks > on > > low memory) workaround enabled (on low memory tls operations will fail > > preemptively) with free memory thresholds 13107200 and 6553600 bytes > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold1 has been > > changed to 13107200 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/cfg/cfg_ctx.c:598]: cfg_set_now(): tls.low_mem_threshold2 has been > > changed to 6553600 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [main.c:2834]: main(): processes (at least): 25 - shm size: 67108864 - > pkg > > size: 4194304 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is > initially > > 212992 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: > > [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally > > 425984 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSs: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSs: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSs: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSs: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSs: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSs: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSs: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSs: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSs: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSs: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: NOTICE: tls > > [tls_domain.c:1107]: ksr_tls_fix_domain(): registered server_name > callback > > handler for socket [:0], server_name='' ... > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSs: Client MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:305]: ksr_tls_fill_missing(): TLSc: tls_method=22 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:317]: ksr_tls_fill_missing(): TLSc: > > certificate='/etc/kamailio/certificados/certificate.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:324]: ksr_tls_fill_missing(): TLSc: ca_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:331]: ksr_tls_fill_missing(): TLSc: crl='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:335]: ksr_tls_fill_missing(): TLSc: require_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:342]: ksr_tls_fill_missing(): TLSc: cipher_list='(null)' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:349]: ksr_tls_fill_missing(): TLSc: > > private_key='/etc/kamailio/certificados/private-key.pem' > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:353]: ksr_tls_fill_missing(): TLSc: verify_certificate=1 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:356]: ksr_tls_fill_missing(): TLSc: verify_depth=9 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:359]: ksr_tls_fill_missing(): TLSc: verify_client=0 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32409]: INFO: tls > > [tls_domain.c:697]: set_verification(): TLSc: Server MUST present valid > > certificate > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32422]: INFO: > jsonrpcs > > [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/32422 > > > > > > Jan 6 15:13:45 Kamailio-Server /usr/sbin/kamailio[32424]: INFO: ctl > > [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io > > watch method (config) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.75.24 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32425]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.132.46 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32426]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242d9278 r: 0x7f45242d93a0 (-1) > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1283]: tls_h_read_f(): protocol level error > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_util.h:42]: tls_err_ret(): TLS write:error:14090086:SSL > > routines:ssl3_get_server_certificate:certificate verify failed > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1287]: tls_h_read_f(): source IP: 52.114.14.70 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: tls > > [tls_server.c:1290]: tls_h_read_f(): destination IP: 161.35.44.66 > > > > > > Jan 6 15:13:55 Kamailio-Server /usr/sbin/kamailio[32427]: ERROR: > > [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error > reading > > - c: 0x7f45242be028 r: 0x7f45242be150 (-1) > > > > > > > > > > > > Could you help me identify the problem please. > > > > > > > > > > > > Cheers > > > > > > > > > > > > Saludos Cordiales > > > > > > -- > > > > > > Willy Valles Rios > > > > > > Unified Communications Specialist > > > > > > > > > > > > phone: +51955747343 > > > > > > em@il: willyvalles17(a)gmail.com > > > > > > > > > > > > _______________________________________________ > > > > > > Kamailio (SER) - Users Mailing List > > > > > > sr-users(a)lists.kamailio.org > > > > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > -- > > > > > > Daniel-Constantin Mierla -- www.asipto.com > > > > > > www.twitter.com/miconda -- www.linkedin.com/in/miconda > > > > > > Funding: https://www.paypal.me/dcmierla > > > > > > _______________________________________________ > > > Kamailio (SER) - Users Mailing List > > > sr-users(a)lists.kamailio.org > > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > VoIP Embedded, Inc. > > http://www.voipembedded.com > > >

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by Dmitry Sytchev 22 Jan '21

22 Jan '21

Hello! In 5.4 release announce there is a line about contact rewriting. "new Contact address encoding/decoding functions – replace/restore Contact address with SIP server address, by encoding/decoding old address in a parameter" Where can I find some documentation or examples about NEW functions? At present we handle this task exactly the same way via parameters manually. It is kinda verbose and hard to understand. Maybe new functions will simplify this. Please advise! Thanks! -- Best regards, Dmitry Sytchev, IT Engineer

2 3

Help required in sipwise rtpengine-ng-client and rtpengine-ctl
by mahesh prasad behera 22 Jan '21

22 Jan '21

Hi Team, We are using sipwise rtpengine on platform centos 7. To get call related statistics from rtpengine. We tried to use utils "rtpengine-ng-client" and "rtpengine-ctl". But for us both of them are not working. *Rtpengine process status:* [root@ctl utils]# ps -ef |grep rtpengine root 3924 668 0 00:11 pts/0 00:00:00 grep --color=auto rtpengine root 20502 20499 0 Jan19 ? 00:10:02 ../sbin/rtpengine -f --num-threads 4 -i pub/10.211.160.132 -i priv/10.211.160.132 -n 127.0.0.1:8500 -c 127.0.0.1:8500 -m 32001 -M 32500 -T 184 -o 90 -d 4 -s 900 -p /var/run/rtpengine1.pid --scheduling rr --priority 37 [root@ctl utils]# * rtpengine-ng-client:* When i ran rtpengine-ng-client, I was getting Bencode.pm missing so I have manually installed perl bencode library "perl-Convert-Bencode-1.03-9.el7.noarch.rpm" Now when i ran rtpengine-ng-client i am getting below error [root@ctl utils]# ./rtpengine-ng-client list *Undefined subroutine &Bencode::bencode called at /usr/local/lib64/perl5/NGCP/Rtpengine.pm line 33. * *rtpengine-ctl:* When i ran ./rtpengine-ctl -ip 127.0.0.1:8500 list , We are not getting any valid response from rtpengine. [root@ctl utils]# ./rtpengine-ctl -ip 127.0.0.1:8500 list *Inside do while after call socket->recv(response, 1024*1024*10)* Please find the below attached source code for rtpengine-ng-client and rtpengine-ctl. Any help and suggestion will be highly appreciated. Thanks, Mahesh

2 1

Kemi execute a route from configuration file
by David VILLAUME 22 Jan '21

22 Jan '21

Hello, I was pretty sure there was a way to call a route from config file from Kemi, but I can’t find documentation about it. Does that really exists or did I dreamed it ? Regards, David

2 2

Kamailio, debian and MySQL caching_sha2_password
by Nicki Otte 22 Jan '21

22 Jan '21

Hi all Have any of you experienced problems using MySQL(Not MariaDB) on a debian machine, with caching_sha2_password? I am having some trouble: Jan 21 08:13:41 MobileValue-Kamailio systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Jan 21 08:13:42 MobileValue-Kamailio kamailio: INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Listening on Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: 185.31.176.39:5060 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: 185.31.176.39:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: Aliases: Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: tcp: voipgw01.evercall.dk:80 Jan 21 08:13:42 MobileValue-Kamailio kamailio[24260]: udp: voipgw01.evercall.dk:5060 Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Plugin caching_sha2_password could not be loaded: /usr/lib/x86_64-linux-gnu/mariadb19/plugin/caching_sha2_password.so: cannot open shared object file: No such file or directory Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:903]: ds_connect_db(): cannot initialize db connection Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatch.c:937]: ds_init_db(): unable to connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: dispatcher [dispatcher.c:404]: mod_init(): could not initiate a connect to the database Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Jan 21 08:13:42 MobileValue-Kamailio /usr/sbin/kamailio[24262]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Control process exited, code=exited, status=255/EXCEPTION Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: kamailio.service: Failed with result 'exit-code'. Jan 21 08:13:42 MobileValue-Kamailio systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. I can get around the problem, by making the kamailio & kamailioro user use mysql_native_password instead. But am i wrong to belieave that it should Work out of the box? Here are the installed packages from APT: kamailio/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-json-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-mysql-modules/unknown,now 5.4.3+bpo10 amd64 [installed] kamailio-outbound-modules/unknown,now 5.4.3+bpo10 amd64 [installed] default-libmysqlclient-dev/stable,now 1.0.5 amd64 [installed] default-libmysqld-dev/stable,now 1.0.5 amd64 [installed] libmariadb-dev/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] libmariadb-dev-compat/stable,now 1:10.3.27-0+deb10u1 amd64 [installed,automatic] mysql-apt-config/unknown,now 0.8.16-1 all [installed] mysql-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-common/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-client-plugins/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-community-server-core/unknown,now 8.0.23-1debian10 amd64 [installed,automatic] mysql-server/unknown,now 8.0.23-1debian10 amd64 [installed] Thanks in advance Nicki Bo Otte

4 8

RTP Querys
by Ahmed Marsou 22 Jan '21

22 Jan '21

Hi guys; I have some rtpproxy questions: - When a proxy is in a failed state, what happens to the active sessions / streams? - How many times the module should retry to send and receive after timeout was generated? - *How often are the control messages sent to the rtp proxy’s in the pool?* *- I would like to know the exact nc message that’s used for the control messages* *Thank you.*

2 2

tracking usrloc clean
by Michal Popovic 22 Jan '21

22 Jan '21

Hi All, I would like to ask you if there is any way how to track usrloc location cleaning operation with some event route, so I can broadcast the change with for example kafka message. If no, how do you keep track of online/offlice status of the user? I am trying to avoid using change data capture in mysql proxy for location table. Michal

2 2

Kamailio and Serverless
by H Yavari 22 Jan '21

22 Jan '21

Hi to all, Does anyone have experience using Kamailio in front of some Lambda / Cloud functions? For example, passing call info with JSON to functions.Please share your ideas. Regards,HYavari

4 4

Kamailio v5.4 install
by Jewna 21 Jan '21

21 Jan '21

I'm having a problem installing kamailio 5.4 using the following guide https://kamailio.org/docs/tutorials/5.4.x/kamailio-install-guide-git/ . I have a debian 10.7 machine where I install default-mysql-server package for the database. After that I follow the install guide and everything goes smoothly until I have to install the init.d scripts. 1. I do "cd /usr/local/src/kamailio-5.4/kamailio' 2. I do "make install-initd-debian" Then make fails with the following error " root@kamServ:/usr/local/src/kamailio-5.4/kamailio# make install-initd-debian make -C src/ install-initd-debian make[1]: Entering directory '/usr/local/src/kamailio-5.4/kamailio/src' sed -e "s#DAEMON=/usr/sbin/kamailio#DAEMON=/usr/local/sbin/kamailio#g" \ -e "s#NAME=kamailio#NAME=kamailio#g" \ -e "s#DESC=Kamailio#DESC=kamailio#g" \ -e "s#HOMEDIR=/run/kamailio#HOMEDIR=/run/kamailio#g" \ -e "s#DEFAULTS=/etc/default/kamailio#DEFAULTS=/etc/default/kamailio#g" \ -e "s#CFGFILE=/etc/kamailio/kamailio.cfg#CFGFILE=/usr/local/etc/kamailio/kamailio.cfg#g" \ < pkg/kamailio/deb/debian/kamailio.init \ > /etc/init.d/kamailio /bin/sh: 1: cannot open pkg/kamailio/deb/debian/kamailio.init: No such file make[1]: *** [Makefile:982: install-initd-debian] Error 2 make[1]: Leaving directory '/usr/local/src/kamailio-5.4/kamailio/src' make: *** [Makefile:34: install-initd-debian] Error 2 " Am I doing something wrong? Possibly in the wrong directory? If you need any more information please don't hesitate to ask. Sincerely, Matthew Sherwood

2 1

[kamailio 5.4.3] xavp_rm_by_index removes (N+1) first indexes instead of index N
by Chaigneau, Nicolas 20 Jan '21

20 Jan '21

Hello I’m trying to use function xavp_rm_by_index to remove a specific index from a given xavp. I’m observing an unexpected behaviour, I think it might be a bug : Instead of just removing the index, the function removes the index and all others before it. E.g : if called with « idx = 1 », it removes indexes 0 and 1. Reading the code of xavp_rm_internal (which is called by xavp_rm_by_index), I don’t think it behaves at it should : /* Remove xavps * idx: <0 remove all xavps with the same name * >=0 remove only the specified index xavp * Returns number of xavps that were deleted */ For example when I do this : LM_ERR("COUNT BEFORE: %d\n", xavp_count(rname, NULL)); int rm_count = xavp_rm_by_index(rname, 1, NULL); // TEST LM_ERR("REMOVED: %d\n", rm_count); LM_ERR("COUNT AFTER: %d\n", xavp_count(rname, NULL)); « rm_count » returned is 1 (which is expected). However, « count after » is decremented by 2. And when I look at the contents of my xavp, I notice that the first two elements have been removed. Likewise if xavp_rm_by_index is invoked with idx = 2 => the first 3 elements are removed. So… do you think this is a bug ? or something I didn’t understand ? Thanks for your help ! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Dialog Tracking with Inconsistent RR
by Charles Phillips 20 Jan '21

20 Jan '21

I am working to track concurrent channels with the dialog module, but I have run in to an issue. Due to the nature of my required configuration, I have a need to strip the record route headers from packets that are sent to the upstream proxy. This leaves no way for the dialog module to track the state of the call. I was hoping that DID_FALLBACK or DID_NONE modes would track by call id or some other constant, but it does not appear to. So, before I start down the path of using htable to store and reinsert the did when a packet returns, is there something I am missing with dialog? Is there some more efficient way to do this? - Charles

2 2

Error: invalid json doc in command "kamctl ping" kamailio 5.4.3
by Julia Bo 20 Jan '21

20 Jan '21

Hello, Kamailio 5.4.3 has been installed from source. "kamctl ping sip:10.10.10.10:5060" is sent from the command line. Errors: Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_mod.c:1352]: jsonrpc_exec_ex(): invalid json doc [[{"jsonrpc": "2.0", "method": "tm.t_uac", "], "reply_name": "kamailio_receiver_85791", "id": 85791}]].", "From:sip:daemon@KAM-TEST Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_fifo.c:461]: jsonrpc_fifo_server(): failed to execute the json document from fifo stream This command is working in 4.4. Has anyone tried using kamctl ping in 5.4? Is it a bug in 5.4.3 ? Thank you, Julia.

3 10

Header Manipulation with returned REST data
by Mark Lawrence 19 Jan '21

19 Jan '21

Hi All, Sorry for the intrusion and asking directly, but there’s so much information out there that I would spend years looking for the answer myself and spending equal amount of time in a lab playing. I do like the last idea but time is a little short. I’m thinking of using Kamailio to overcome a few downfalls in the main SBC/PBX commercial product space that I’ve typically deployed and wanted to ask if it is possible to use as a stateful proxy and carry out the following header manipulation type scenario with a bit more dynamassium: Use the FROM and TO header to create two value pairs that would be sent via a REST request to an external application, which will return two corresponding values to (a) supply a different value to update the original FROM header and (b) a full PAI sting to add on to the outgoing message. If you have experience of this or if it’s a nonstarter I’d love to know. There’s so many modules and great stuff that you have in your toy chest it is amazing and kudos to all you developers being so inventive out there: it leaves me to think I’ve been deploying SIP wrong all these years. Thanks and keep safe and well, Mark

2 1

Count the rows in a xavp from script
by Chaigneau, Nicolas 19 Jan '21

19 Jan '21

Hello, I'm looking for a way to retrieve the number of rows stored in a xavp. (Something better than manually iterating over the xavp) Basically, I'd like to call xavp_count from the script. Is that possible ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Kamailio High Availability questions
by Andriano 17 Jan '21

17 Jan '21

Hello have some questions about Kamailio HA + loadbalancers (dispatcher modules) 1. is posible to make HA for RTP media? maybe RTPEngine +redis? is not possible to do this with RTProxy? where is some examples? 2. What do Kamailio db cluster module? it can replicate every DB query record like normal DB cluster? or better is to use native MySQL cluster mode? 3. How is working dispatcher modules in HA? how call is handled and htable in memory between loadbalancers endpoint nodes etc... IS it posible to do only with DMQ ? Just need understand how to do basic Kamailio setup with HA and load balancers in general, for example if will be 2 phisical servers, in every server is virtualisation and every server has 2 VPS. 1 VPS for frontend Kamailio loadbanancer, 1 VPS backend Kamailio app with accounts, routing logic, LCR, RTP media app etc... between load balancers i know maybe is possible to do floating ip, dns round robinetc... and if one VPs server down, call will be dropet in every situation? or call will not be droped and how to prevent it. Just need to know in general schema and architecture how to do this real HA in Kamailio and avoind sinlge point of failure and where to read and learn it ho to make HA in real. In the internet no many docs about it, many conference video how good it is, but no good examples with config, i did not founded, so will be very glad for any help in these newbie questions ;) with single kamailio is many answers, but is realy difficult to understand how to do voip HA with Kamailio without single point of failure. Thanks for any answers...

1 0

http_client module not found
by Joli Martinez 15 Jan '21

15 Jan '21

Hello, I am running kamailio 5.3.8 on Debian 9. Kamailio is installed via apt package manager. I am trying to get apiban.org to work and it seems to be having a problem with the http_client module. I have searched for the kamailio modules and I am unable to find anything that resembles http. I have also installed the extra module and no luck. What would be the correct module to install? kamailio: ERROR: <core> [core/sr_module.c:500]: load_module(): could not find module <http_client> in </usr/lib/x86_64-linux-gnu/kamailio/modules/>

2 2

Stubs for KEMI Javascript
by Nicki Otte 15 Jan '21

15 Jan '21

Hi everyone. I wanted all of you to know, that I have made JavaScript stubs, for autocompletion in IDE’s, for those of you that use KEMI JavaScript. Specifically tested on Jetbrains PHPStorm/Webstorm. The stubs are available for free at my GitHub page. The link is https://github.com/captainDuckay/Kamailio-javascript-stubs My main language isn't typescript or JavaScript, and this is one of my first public releases. With that said though, I am open for critics, or even guidance. I hope it's usable for you. // Nicki Bo Otte

2 3

To header being malformatted.
by AL S 14 Jan '21

14 Jan '21

Hi, We noticed an issue with the way the Kamailio is handling certain messages being sent to them, specifically, the To header being relayed from Kamailio to the next hop is malformatted. Any assistance is greatly appreciated INVITE received by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+12053140346@72.31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 INVITE forwarded by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+120531403Oh.lc".3/+0]1..31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 Another example in BYE: BYE received by Kamailio: To: "WIRELESS CALLER" <sip:+18131111111@71.44.13.84>;tag=gK0a788ca6 BYE forwarded by Kamailio: To: "WIRELESS CALLER" <sip:+18131112222A.++.+,-94 Thanks. AL

2 2

Source Port on TLS OPTIONS from Dispatcher
by Charles Phillips 14 Jan '21

14 Jan '21

Hello all. As they say in radio, “long time listener, first time caller” Anyway, I am having trouble getting past the following road block and any help would be greatly appreciated. Kamailio version is 5.4.3 When attempting to use dispatcher to send OPTIONS packets to several TLS destinations, the packets are leaving the Kamailio server on random ports. This is a problem because the servers I am sending the OPTIONS to (MS Teams) are enforcing rport so the responses are returned to a port that Kamailio is not listening on. I have tried to force the socket in the event route (relevant parts of snippet below) but it does not appear to help. I should also mention that I am not behind NAT and the TLS socket is specified in the dispatcher attrs. event_route[tm:local-request] { sip_trace(); $fs = “tls:**ip-address**:5061”; } I have used Kamailio as a TLS server for many projects, but this is my first time as a client. I am sure I am missing something. - Charles

4 10

BYE and TCP
by Kjeld Flarup 13 Jan '21

13 Jan '21

Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

5 17

handling rejected REFER messages
by Jaime Gajardo 13 Jan '21

13 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux

2 1

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environmet. behind the nat we have 2 identycal nodes using REFER for transfer call to another destiny. in the other NAt side we have our public carrier. When some node send to kamailio the REFER message and kamailio reroute this message to the carrier. The call is rejected due company policies. We want know how can we do for convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

1 0

Re: [SR-Users] Using websockets with kamailio 5.4
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Daniel - Google is your friend. I use it often. https://github.com/havfo/WEBRTC-to-SIP

1 0

TSILO - transaction is not suspended
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Folks - I am attempting to set up a mobile push configuration, by following the published example(s). If we have more than one endpoint registered, this configuration works as expected (via the ts_store() path below). If we do not have another registered endpoint the transaction seems to be suspended but then cannot be resumed (t_suspend) after REGISTER arrives. INFO: <script>: [R-LOG]: INVITE from IP:[24.XXX.XXX.XXX:62665] [sip:jeremy2@stage][<null>] NFO: <script>: [R-SUSPEND]: Suspended transaction [41359:415220139] INFO: <script>: [R-SUSPEND]: fU:[jeremy] rU:[jeremy2] tU:[jeremy2] INFO: <script>: [R-SUSPEND]: fU:[jeremy] => tU:[jeremy2] INFO: <script>: [R-SUSPEND]: vtp:[41359:415220139] INFO: <script>: [R-SENDPUSH]: Push sent [(55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78][com][hq01gt5utegi18ipkh1b] [Node-Push]: About to push: 55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78 INFO: <script>: R-REGISTER]: jeremy2 registered INFO: <script>: [R-REGISTER]: Contact: [<sip:ilyartjr@qjwhdafxlbtk.invalid ;transport=ws>;expires=10;app-id=stage;pn-voip-tok=55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78] INFO: <script>: [R-REGISTER]: App id: stage INFO: <script>: [R-REGISTER]: Expires: 10 INFO: <script>: [R-JOIN]: jeremy2 INFO: <script>: [R-JOIN]: Resuming trasaction [41359:415220139] jeremy2 (41359:415220139) WARNING: tm [t_suspend.c:197]: t_continue_helper(): transaction is not suspended [41359:415220139] WARNING: tmx [tmx_mod.c:847]: w_t_continue(): resuming the processing of transaction [41359:415220139] failed Relvant kamailio.cfg [Note: I did notice we had to switch things up a bit if calls come from a PSTN number via SBC] modparam("htable", "htable", "vtp=>size=10;autoexpire=300") ... if (!lookup("location")) { send_reply("100", "Suspending"); route(SUSPEND); // *testing this path here * } else { route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-LOCATION]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); } ... route[SUSPEND] { if (!t_suspend()) { xlog("L_ERROR", "[R-SUSPEND]: Failed to suspend transaction [$T(id_index):$T(id_label)]\n"); send_reply("501", "Unknown Destinaton"); exit; } xlog("L_INFO", "[R-SUSPEND]: Suspended transaction [$T(id_index):$T(id_label)]\n"); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] rU:[$rU] tU:[$tU]\n"); if (isbflagset(FLB_PSTN)) { $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => rU:[$rU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$rU)]\n"); } else { $sht(vtp=>join::$tU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => tU:[$tU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$tU)]\n"); } } route[REGISTER] { if (!is_method("REGISTER")) { return; } if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!save("location")) { sl_reply_error(); } route(JOIN); } # Append branches or resume the transaction route[JOIN] { xlog("L_INFO","[R-JOIN]: $tU\n"); $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) { xlog("L_INFO","[R-JOIN]: Appending transaction ($tu)\n"); ts_append("location", "$tu"); exit; } xlog("L_INFO","[R-JOIN]: No INVITE to resume\n"); exit; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xlog("L_INFO", "[R-JOIN]: Resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); exit; } # Lookup and relay after resuming transaction route[RESUME] { xlog("L_INFO", "[R-RESUME]: ru: [$ru] tu: [$tu]\n"); lookup("location"); route(SETUP_ROUTES); route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-RESUME]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); exit; } What did I miss?

4 6

Profile name as variable in set_dlg_profile
by Muhammad Sohaib 11 Jan '21

11 Jan '21

Is it possible to insert the current dialog into the profile by using the following way? modparam("dialog", "profiles_no_value", "mytest;xyz") $var(profilename)="xyz"; set_dlg_profile(" $var(profilename) "); Currently it is giving errors and not allowing to use variable instead of static profile name. Or is there any alternative to pass profile name as variable? --- Thanks,

4 6

Unload UAC remote registry
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm on Kamailio 4.4.7 and I'm trying to do uac.reg_refresh over a row non existing on DB to try to unload, but data still persist on Kamailio. How to unload only one registry from UACs remote registrations when delete a record from DB without do a uac.reg_reload? UAC remote registrations is above 2k rows, I don't know if it's much or not. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

3 2

Different responses for domain.dump
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there! I'm getting different responses from Kamailio 4.4.7 when domain.dump exec; From *kamcmd*: I'm getting two domains; .es and .net. Request to jsonrpc-s module: Response from jsonrpc-s module: And now, I'm getting only one domain. What's I'm doing wrong?

2 2

uac.reg_refresh command
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm playing with RPC on my Kamailio 4.4.7, and I getting rare behaivor with *uac.reg_refresh* command. When I run it thru kamcmd like docs describes, this is without *l_uuid* parameter, I'm geting "*error: 500 - Invalid Parameters*" output. And when I run with *l_uuid* parameter, I get no output. But in both cases, Kamalio don't refresh data from DB. But I run this command thru HTTP (JSONRPC-S), both call modes are equivalent. With or without l_uuid parameter geting no output and refresh data from DB. It's a normal behaivor? Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 3

list myself values
by Javier Valencia 11 Jan '21

11 Jan '21

Hi, a question from a Kamailio newbie. How can I list or show variable *myself* values? Regards, JV

2 2

UAC module registrations interval
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there, I'm using Kamailio 4.4.7 with UAC remote registrations module. In my database have all rows with expires column value's 3600, and my PBX (asterisk) min, max and default expires to 3600. But I can see how Kamailio it's sending REGISTER each 150segs. UAC module parameters values; modparam("uac", "reg_db_url", "mysql://XXX") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "XXX") modparam("uac", "reg_timer_interval", 5) modparam("uac", "reg_retry_interval", 60) My uacreg table it's above 2k rows. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

2 1

how does rtjson work?
by Syed Shahryar 11 Jan '21

11 Jan '21

My use case is such that I want to be able to send a SIP MESSAGE destined for a single username to multiple users. In order to be able to map that target username to multiple usernames, I tried using rtjson and evapi/http (to generate the routing json), but none of them worked. Is it the right approach? Is there any other way to programmatically generate multiple targets for SIP request? I am familiar with dynamic routing but I need more flexibility. Initially, I tried sending the request to a different username but that didn't work and got the following message in the http async result route. kamailio | 19(25) ERROR: nathelper [nathelper.c:1461]: sdp_1918(): Unable to parse sdp body kamailio | 19(25) NOTICE: <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi msg branch kamailio | 28(34) NOTICE: {2 6661 MESSAGE q6hikd6tumsr1dh35kvj} <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi reply branch The message was received on the originally requested to user and also $tu still points to the original user instead of the new user even after executing the following code. KSR.rtjson.init_routes(response); KSR.rtjson.push_routes(); KSR.nathelper.fix_nated_contact(); In http response, I receive the following and don't see any rtjson errors when init_routes is executed on it. {"version":"1.0","routing":"parallel","routes":[{"headers":{"from":{"display":"test_1","uri":"sip: *test_1*@test.com:9908"},"to":{"display":"test_4","uri":"sip:*test_4*@ test.com:9908 "},"extra":""},"branch_flags":8,"fr_timer":5000,"fr_inv_timer":30000}]} Please note that the request was sent to *test_2*(a)test.com. I have tried other combinations of above but nothing works. I have already searched on Google and read relevant articles and blog posts. Thanks -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Lost MySQL connections - Error 500
by Duarte Rocha 11 Jan '21

11 Jan '21

Hi all, I've made a post last month regarding losing MySQL connections - https://lists.kamailio.org/pipermail/sr-users/2020-December/111389.html At the time I thought connections were dying as a consequence of low activity and traffic on the proxy. Meanwhile, I've migrated a great number of equipments to the proxy with Registers being refreshed every 10minutes and the problem still persists. In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. My SQL client on the Kamailio machine is mysql-community-client 5.6.50-2.el7. It writes and reads in a remote InnoDB database. This are the logs i get from Kamailio when the problem appears : Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lock wait timeout exceeded; try restarting transaction (1205) Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} <core> [db_query.c:348]: db_do_update(): error while submitting query Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1663]: update_contact_db(): failed to update database Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} registrar [save.c:784]: update_contacts(): failed to update contact Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Originally I had usrloc db_mode on mode 3 - DB-Only Scheme. In order to try to mitigate the issue I changed it to mode 1 - Write-Through scheme but even then I get the same log errors and an "500" error is still sent to the client. I chose this mode since, as far as I can understand it applies changes directly to DB but also uses cache. Please correct me if i'm wrong on that. Has this issue happened with anyone before? Is there a way to mitigate this issue? My only constraint is that I need the database to be always updated since I have an HA setup, and as such, I can't use cache only methods. Best Regards,

2 5

loose_route() - Local Proxy
by Duarte Rocha 08 Jan '21

08 Jan '21

Greetings, I'm using loose_route() from the RR module and i'm having troubles making it use the following exception from the code : "There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." My example is a REGISTER without To-TAG which has a Route header with kamailio address. If i use proxy IP on Route header , loose_route() returns false as it should. However, if i use an hostname belonging to the proxy in the route, loose_route() returns true. I have hostnames and local ips defined in the "DOMAIN" table but it doesn't seem to be working. Which other places can local hostnames and ips be configured in order to be seen as local to loose_route() ? Best Regards

4 4

Contact header not removed properly Getting some junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my scenario call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org/> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060/>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.xx.xx.xxx 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.xx.xx.xxx 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

about Contact Header Junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my requirement call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

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Dispatcher SNI Questions
by Tim Chubb 06 Jan '21

06 Jan '21

Hi All Is it possible to explicitly define the tls cert to be used per destination by dispatcher? I'm attempting to integrate with a service that requires domain name presented to match that of the cert. Where the use of dispatcher is probably not as intended anyway as it targets the same set of destinations multiple times (primary and secondary endpoints for the service im integrating with), with multiple distinct set id's (basically a set per customer for multi tenancy) which expects a sip ping per account to verify that connection is alive and healthy. #Example Dispatcher list #Account 1 1 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061 1 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061; #Account 2 2 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; 2 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; Currently I'm running a solution based around certs with multiple SAN (subject alternative name) defined, but this is a pain to administrate, and not as scalable as I would like. I want to be able to define multiple client:any tls profiles and explicitly send via that profile. Its easy enough to do that in config using the tls xavps to define a server name and/or id. But for the sip options ping by dispatcher to work you need to specify the server name or id before the tm:local-request event route fire's. I've resorted to hacking away at dispatcher to see what would happen if I set the tls server name or id before the OPTIONS message is sent, and it works great for the first tls client profile matched, but any others you define re-use the initial tcp connection so reuses the first connections tls config, thus presenting the wrong cert (I'm probably wrong but logs show only one tls complete_init() related to dispatcher pings and the second set's pings skips all the tls init logging so suggests some thing is cached/reused, and logging on the service indicates cert name mismatch with set1s cert being offered) So I've hit a bit of a dead end, is there a way to force a new tcp connection per server name/id I fear outbound SNI might be a bit of a can of worms, for it to work nicely with dispatcher as is, would be an extension of the proto:host:port format to include a name or id, maybe something like: tls(name=sub.domain.com):10.0.0.5:5061 But that looks like it would have a lot of wide reaching knock-on effects, so setting it as an attribute like ping_from works fine as I'm currently doing, if I could force a new connection from dispatch.c.... The other thing that's occurred to me is that it might be better to stop trying to adapt a small percentage of dispatcher's functionality and trying to hammer a round peg into a square hole and instead try and create a new module similar to dispatcher in so much that it sends pings (that's all I really need in this case) and use dispatcher as intended with a single set of non-repeated destinations for routing. This would also allow much simpler integration with MS teams (if added ability to set contact address on the background pings) as well as the service im trying to integrate, and a more natural data set to be used as in my example you can see the only difference is the customers domain between each set. How I would imagine it to work would be to create a tls connection per customer domain, then iterate through the connections per destination in pseudo json the data would look like this: { Destinations:[ [1,["a.cloudservice.com","b.cloudservice.com"]], [2,["sip.pstnhub.microsoft.com", "sip2.pstnhub.microsoft.com", "sip3.pstnhub.microsoft.com"] ], Sources:[ {uri:"sip.account1.com",destinationMapping:[1]}, {uri:"sip.account2.com",destinationMapping:[1,2]} ] } And would just run in the background sending pings with SNI support, and if based on dispatcher should make it easy enough adapt things like the destination up and down event routes, and would allow me to use Kamailio much like I would use nginx for web stuff as a reverse proxy and handle TLS offload on my edge gateway proxy. Any thoughts?, im frustratingly close to having something that will do if I could get a new connection to be created, but I fear my use case might be sufficiently different from dispatchers to make a new module a sensible approach.. Regards Tim.

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Pre-loaded routes verification
by Duarte Rocha 04 Jan '21

04 Jan '21

Greetings, I'm doing a migration from a opensips system to kamailio one and i'm trying to replicate its functionalities. The system is a Registrar with a pstn gateway. I noticed that on the opensips version, loose_route() is called on initial requests and it seems to be used to protect against preloaded routes. However, on the default Kamailio configuration file loose_route() is only called on requests in-dialog and the verification mentioned above is not used. Why doesn't Kamailio includes this verification? Is there are any security concerns not using it? Best Regards,

2 1

using kamailio 5.4 with the websocket or xmlhttprequest
by Daniel Hermann N'don 04 Jan '21

04 Jan '21

Hello everyone, I'm a beginner in kamailio technology, and I would like to know how to communicate with kamailio from the web.

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Unable to connect sip with Kamailio
by Rohit Kumar 04 Jan '21

04 Jan '21

Hello everyone, I'm beginner in Kamailio, I want to connect my asterisk Server with Kamailio. Can you please help me in this context. I shall be very thankful to you. Regards, Rohit Kumar On Thu, Dec 31, 2020, 16:31 <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 2. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Antony Stone) > 3. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 4. Using a specific log level for "script" ? (Chaigneau, Nicolas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 30 Dec 2020 14:15:19 +0000 > From: Daniel Hermann N'don <danielhermann.ndon(a)outlook.com> > To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> > Subject: [SR-Users] using kamailio 5.4 with the websocket or > xmlhttprequest > Message-ID: > < > MWHPR05MB291167705F86195717CA7E3181D70(a)MWHPR05MB2911.namprd05.prod.outlook.com > > > > Content-Type: text/plain; charset="iso-8859-1" > > Hello everyone, I'm a beginner in kamailio technology, and I would like to > know how to communicate with kamailio from the web. >

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Using a specific log level for "script" ?
by Chaigneau, Nicolas 04 Jan '21

04 Jan '21

Hello, Is it possible to set a specific log level for the Kamailio script ? I've seen that the < debugger > module allows to do that with modules or < core > : https://kamailio.org/docs/modules/5.4.x/modules/debugger.html#dbg.p.mod_lev… So I've tried to do : modparam("debugger", "mod_level_mode", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level", "my_module=3") // this works modparam("debugger", "mod_level", "core=3") // also works modparam("debugger", "mod_level", "script=3") // nope... But it doesn't seem to work for < script >... :/ Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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[Presence] publ_cache=2 usage
by Denys Pozniak 30 Dec '20

30 Dec '20

Hello! Could somebody explain how the parameters *publ_cache=2* and *subs_db_mode* depend on each other? Do I understand correctly that only the following modes of *subs_db_mode* are allowed 0,1,2? *subs_db_mode:0 - This disables database completely. Only memory will be used. Subscriptions will not survive restart. Use this value if you need a really fast presence module and subscription persistence is not necessary or is provided by other means.1 - Write-Through scheme. Subscriptions are updated synchronously in database and in memory(used for read operations). Use this scheme if speed is not top priority, but it's important that no subscriptions will be lost during crash or reboot or if you have an external application that reads the state of the subscriptions from database and they need to be updated synchronously.2 - Write-Back scheme. This is a combination of previous two schemes. All changes are made to memory and database synchronization is done in the timer. The timer deletes all expired contacts and flushes all modified or new subscriptions to database. Use this scheme if you encounter high-load peaks and want them to process as fast as possible. Latency of this mode is much lower than latency of mode 1, but slightly higher than latency of mode 0. To control the interval at which data is flushed to database, set the db_update_period parameter.3 - DB-Only scheme. No memory cache is kept, all operations being directly performed with the database. The timer deletes all expired subscriptions from database. The mode is useful if you configure more servers sharing the same DB without any replication at SIP level. The mode may be slower due the high number of DB operation.* -- BR, Denys Pozniak

1 0

Presence - issue with NOTIFY "terminated" state
by Zhan Bazarov 29 Dec '20

29 Dec '20

Hello everyone! I faced some unfortunately issue with presence module. Let me describe setup scheme a bit: I'm publishing dialog states via pua_dialoginfo from one kamailio server(for calls proccessing itself) to another one kamailio(standalone kamailio just for handling presence,blf,mwi etc) So, about issue, let's say we have device A which has outgoing subscription to extension which belongs to device B and device C(both devices has same extension but different usernames, webrtc+sip phone, effectively its the same Phone, I'm using pubruri_avps to change entities from usernames to extension) . And, in some cases,for examle, when thirtparty is making call to this extension(B,C devices) and making CANCEL after couple seconds - I see the PUBLISH from callserver to presence server with state "terminated', I see 200ok response.. but presence server isn't sending NOTIFY to device A about state "terminated", everything is good with "trying" and "early" states... That problem is happening 1 of 50 tries.. I've tried a lot of things to fix that - still are not succeed. p.s. everything is good from logs/sip point of view... And also, when I disable the webrtc device - I'm not able to reproduce that issue anymore. Looks like something wrong with case, when you have multiple different username behind the same presentity. Maybe someone had same issue and may help me. Thanks! -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Kamailio as a presence watcher
by Jukka Heikkilä 29 Dec '20

29 Dec '20

Hello, I'm working with IMS network and VoLTE related services. My problem is that I would like to use Kamailio as a presence watcher but I haven't managed to make it happen yet. The approach would be that once Kamailio gets 3rd party registration from S-CSCF the Kamailio would send SUBSCRIBE back to S-CSCF to subscribe the presence of subscription. Once a subscription place or get calls the status of subscription would be updated to Kamailio and to the database where it can be fetched. I have checked presence and pua modules, but haven't found a way to make it work as expected. Is there way to do that or is this totally incompatible approach to do that? Best Regards, Jukka Heikkilä jutsco(a)gmail.com

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How to implement a self-sustained Presence server
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration? That is I want the Presence module to be fed status updates as Kamailio is routing calls and handling registrations. So when a user registers, his phone is ringing, he is on the phone, etc.., the Presence module gets notified and can update its subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

2 1

Kamailio suddenly stops
by Javier Valencia 28 Dec '20

28 Dec '20

Hi there! When I start my webrtc kamailio suddenly stops. I can see this on syslog: Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c2ab8), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c3138), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring ... omit vervose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c1f38), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring ...omit verbose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557569f250), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_realloc(299) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557565cf60), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized OS info: root@webrtc:/etc/kamailio# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster root@webrtc:/etc/kamailio# free -m total used free shared buff/cache available Mem: 16041 134 15551 108 356 15524 Swap: 974 0 974 Kamailio version: version: kamailio 5.4.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 8.3.0 Kamailio params: SHM_MEMORY=512 PKG_MEMORY=32 SSL info: root@webrtc:/etc/kamailio# apt show openssl Package: openssl Version: 1.1.1d-0+deb10u4 Priority: optional Section: utils Maintainer: Debian OpenSSL Team <pkg-openssl-devel(a)lists.alioth.debian.org> Installed-Size: 1.495 kB Depends: libc6 (>= 2.15), libssl1.1 (>= 1.1.1) Suggests: ca-certificates Homepage: https://www.openssl.org/ Download-Size: 843 kB APT-Manual-Installed: no APT-Sources: http://security.debian.org/debian-security buster/updates/main amd64 Packages Description: juego de herramientas Secure Sockets Layer - utilidad criptográfica Este paquete forma parte de la implementación del proyecto OpenSSL de los protocolos criptográficos SSL y TLS para la comunicación segura a través de Internet. . Contiene el binario de propósito general para la línea de órdenes /usr/bin/openssl, útil para operaciones criptográficas como: * creación de parámetros clave RSA, DH y DSA; * creación de certificados X.509, CSR y CRL; * calcular resúmenes de mensajes; * cifrado y descifrado con cifras; * probar clientes y servidores SSL/TLS; * gestión de correo firmado o cifrado S/MIME. Should I use another version of openssl, or maybe use gnutls? Regards, Javier Valencia

1 0

Self-sustained Presence server.
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration. That is I want the Presence module to be fed status updates from the same Kamailio that is routing the calls. So when a user registers, his phone is ringing, he is busy, on the phone, etc.., the Presence module gets notified and can update subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

1 0

Read $fd from mysql?
by Mihai Cezar 27 Dec '20

27 Dec '20

Hello, I am redirecting the agents/reps to their asterisk server based on the hostname which they connect, and, i wonder if it's possible to read from db instead of the file cause it's getting big ;) # Dispatch requests route[DISPATCH] { switch ($fd){ case "oro.streamlinepbx.nl": if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; case "oro3.streamlinepbx.nl": if (!ds_select_dst("2", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; ............ default: log("unknow destination?"); send_reply(503, "No service defined"); xlog("--- SCRIPT: going to <$ru> via <$du> ...Exiting"); exit; } xlog("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Thanks,

4 3

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 27 Dec '20

27 Dec '20

A year that was very tough for the entire world is getting close to its end, no doubt that was the most challenging in our life so far for many of us, however, with a distributed, friendly and collaborative community, Kamailio project continued to develop consistently, bringing out a new major release. I trust it helped people and companies world wide to to connect with the dear ones and build a better way of living. The season holidays are ahead, time to rest a bit, spend time with family and friends, being online or face to face. As usual, I take this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2020 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 4

app_lua
by rob.van.den.bulk＠gmail.com 26 Dec '20

26 Dec '20

Hallo, I try to install the module app_lua in Kamailio 5.3. ls: cannot access '/usr/lib/liblua*': No such file or directory ls: cannot access '/usr/lib/liblua*': No such file or directory CC (gcc) [M app_lua.so] app_lua_mod.o In file included from app_lua_mod.c:32: app_lua_api.h:25:10: fatal error: lua.h: No such file or directory #include <lua.h> ^~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: app_lua_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/local/src/kamailio-5.3/kamailio/src' make: *** [Makefile:34: all] Error 2 Any help appreciated Greeting Rob van den Bulk

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Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 23 Dec '20

23 Dec '20

FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

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MID-Registrar Example
by moisesynfante 23 Dec '20

23 Dec '20

Hello, I'm a newbie when it comes to Kamailio and I would like to know if there's a working example of a Mid-Registrar implementation. I read on other posts that the UAC module can be used to achieve this but, honestly, I don't understand how. If any of you could provide some path I could follow I would be very thankful. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Re: [SR-Users] How to use port number using $env in listen
by Syed Shahryar 22 Dec '20

22 Dec '20

Any updates on this? On Tue, Dec 15, 2020 at 8:13 PM Syed Shahryar <syedshahryar(a)gmail.com> wrote: > I am new to Kamailio so may be doing something wrong. I looked at > https://www.kamailio.org/wiki/cookbooks/5.4.x/core. > > I tried the following combinations but getting errors in all cases: > > #!subst "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdef "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdefs "/EVAPI_PORT/$env(EVAPI_PORT)/" > > *Errors:* > 0(6) ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching > pvar "env" > 0(6) ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char > [E/69] in [$env(EVAPI_PORT)/] at [5 (5)] > 0(6) ERROR: <core> [core/re.c:170]: parse_repl(): bad specifier in > replace part /$env(EVAPI_PORT)/ > 0(6) ERROR: <core> [core/ppcfg.c:69]: pp_subst_add(): bad subst > expression: /EVAPI_PORT/$env(EVAPI_PORT)/ > > At a later step, I would like to do: > #!define EVAPI_LISTEN "0.0.0.0:EVAPI_PORT" > > And use EVAP_LISTEN in mod_param for the evapi module. > > And something similar to above for TLS settings. > > > On Tue, Dec 15, 2020 at 12:41 PM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> as written on the tracker, look at using #!subst or #!substdef (see core >> cookbook), they should be able to get the substitution value from >> environment using $env(NAME). >> >> The master branch introduced #!defenv as a (simpler) alternative. >> >> Cheers, >> Daniel >> On 12.12.20 05:33, Syed Shahryar wrote: >> >> I tried many combinations of the following but I keep getting syntax >> errors: >> >> listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) >> >> >> TLS_IP and PUBLIC_IP are constants defined at the top using #!define. >> And, TLS_PORT is supposed to be the env variable. >> >> >> -- >> -- >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee, you should not >> disseminate, distribute or copy this email. Please notify the sender >> immediately by email if you have received this email by mistake and delete >> this email from your system. Email transmission cannot be guaranteed to be >> secure or error-free, as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender, >> therefore, does not accept liability for any errors or omissions in the >> contents of this message which arise as a result of email transmission. If >> verification is required, please request a hard-copy version. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> > > -- > -- > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee, you should not > disseminate, distribute or copy this email. Please notify the sender > immediately by email if you have received this email by mistake and delete > this email from your system. Email transmission cannot be guaranteed to be > secure or error-free, as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender, > therefore, does not accept liability for any errors or omissions in the > contents of this message which arise as a result of email transmission. If > verification is required, please request a hard-copy version. > -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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ndb_redis with keydb master/master
by David VILLAUME 22 Dec '20

22 Dec '20

Hello, I consider using keydb-server in an active/action setup without sentinel. Do you know if there is a way to declare both hosts in ndb_redis with the same server name (target use : topos) ? Regards, David

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Re: [SR-Users] DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 22 Dec '20

22 Dec '20

Apologies for not having included the sr-users in the previous email. Anyhow, the solution suggested by Social Boh indeed worked, as a side effect of which we would not run into the issue described. *NOTE:* There was a peculiar configuration in keepalived with keeping both the nodes as BACKUP and setting the 'nopreempt' flag only in the actual MASTER(one with higher 'priority') side to make the keepalived thing to work properly. However, I was wondering if someone could confirm if this issue has been fixed inside the source code itself as I am using an older release v5.3.2 Regards, Harneet Singh On Thu, Dec 17, 2020 at 9:52 PM harneet singh <hbilling(a)gmail.com> wrote: > Thanks Social Boh, > > I will investigate this possibility and if it does work in our case, that > should circumvent the issue. > In addition, is there a fix known to the SR-Users which could be available > inside the Dispatcher module to cause syncing just like the syncing is > there in the Dialog module via DMQ. I wanted to check this as I am using > v5.3.2 and unaware if new releases could have fixed this issue? > > Regards, > Harneet Singh > > On Wed, Dec 16, 2020 at 10:46 PM Social Boh <social(a)bohboh.info> wrote: > >> Maybe a option is leave the secondary like primary when primary go up >> newly. >> >> Example: Primary go down, Secondary is the new Primary. Primary came back >> up but Secondary still is the VIP and still process the calls. >> >> Only if Secondary go down, primary is the new VIP. >> >> The keepalived parameter for this behaviour is: >> >> *nopreempt* >> >> --- >> I'm SoCIaL, MayBe >> >> El 16/12/2020 a las 11:57 a. m., harneet singh escribió: >> >> Hi All, >> >> I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual >> IP) functionality, and have run into a rather peculiar issue. >> >> *Setup:* >> >> Caller ------------ KamailioVIP(*Primary *and *Secondary* >> Kamailio)------- Callee >> >> HA provided by Keepalived >> DMQ is used between the Primary and Secondary Kamailio for dialog sync >> >> *Issue Seen:* >> >> *Suppose the Primary Kamailio has been brought down* and the Secondary >> one is actually active and tied to the VIP. >> >> When a call is fired from the Caller, it traverses through the Secondary >> Kamailio and lands onto the callee. The dialogs are updated properly. At >> this point, the Primary Kamailio is brought up, and dialog state is synced >> due to the DMQ module. >> The Keepalived will now attach the VIP to the Primary Instance. If the >> caller hangs up the phone at this point, the BYE sip message traverses >> through the Primary Kamailio instance to the callee and the call is >> cleared, but there are two issues here. >> >> 1. The Primary Kamailio throws an error in *ds_load_remove() *saying >> that it cannot find the load for the specific call id. This is apparently >> due to the fact that the dispatcher data is not synced between the two >> modules but dialog data is. So dialog wise, things are fine. Can this be >> fixed somehow? >> >> 2. The above is still as grave a problem as the dialogs are cleared. >> BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* >> kamailio even well after the call is cleared, the *DLGLOAD* shows 1. >> Since we are using the *Call Load based distribution* for the >> dispatching, this is effectively one call stuck on the dispatcher, which >> leads to resource leak. >> >> Is this a known issue, and if so, do we have a way to circumvent this? >> >> Regards, >> Harneet Singh >> -- >> "Once you eliminate the impossible, whatever remains, no matter how >> improbable, must be the truth" - Sir Arthur Conan Doyle >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > "Once you eliminate the impossible, whatever remains, no matter how > improbable, must be the truth" - Sir Arthur Conan Doyle > -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

1 0

Kamailio v5.3.8 released
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, Kamailio SIP Server v5.3.8 stable has been released. This is a maintenance release of the previous stable branch (5.3), that includes fixes since the release of v5.3.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are recommended to be upgraded to v5.3.8. For more details about version 5.3.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/12/kamailio-v5-3-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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MSTeams Kamailio with PBX Hold Refer issue
by me.projects＠yahoo.com 21 Dec '20

21 Dec '20

Hi, I have been trying to integrate MSTeams with PBX Through Kamailio. Calls are working now. Hold has Issues. MSTeams uses REFER to manage this all with NOTIFY. I am sending t_send_reply("202", "Accepted"); when receive Refer now I need to send NOTIFY I think I can use t_uac_send() NOTIFY I am unable to send t_uac_send properly on when recertifications refer from MSTeams. Any one with the experience on this ? Thanks,

2 1

Trying to use $sel(contact.uri.params)
by Chaigneau, Nicolas 21 Dec '20

21 Dec '20

Hello, I'd like to use a < select > as described here: http://www.kamailio.org/wiki/cookbooks/5.4.x/selects#contactparams_s I'm trying to extract a parameter from the Contact URI. What I've tried to do: xlog("L_INFO", "contact.uri: $sel(contact.uri)\n"); # OK xlog("L_INFO", "contact.uri.params: $sel(contact.uri.params)\n"); # null ?? This does not seem to work... see output here : 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri: sip:+33145678901;tgrp=YC031;trunk-context=YC90@10.31.22.2:5099 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri.params: <null> What am I doing wrong ? Maybe this is the wrong approach for what I'm trying to do ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Releasing Kamailio 5.3.8
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, I am planning to release next week (likely on Monday 21st December) the next minor version out of branch 5.3, version 5.3.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

send MESSAGE to other sessions of same user
by Syed Shahryar 21 Dec '20

21 Dec '20

What's the best way of sending a message to all other sessions (and the receiver) of the same sender? For example: If user1(a)example.com is logged in to Kamailio from multiple devices and from one of the devices user1 sends a SIP message to user2(a)example.com. I want the message forwarded not only to user2(a)example.com but also to all other user1(a)example.com sessions. What's the easiest way to achieve this? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

1 0

Kamailio not updating CSeq after uac_auth() for ACK
by Terry Tailor 18 Dec '20

18 Dec '20

Hi, Here is setup details Kamailio Proxy a <-----> Kamailio Proxy b <-----> Asterisk Kamailio proxy b auth based on IP Kamailio Proxy a 1. Kamailio proxy sends an Invite request to Kamailio proxy b with CSeq: 102 INVITE 2. Kamailio proxy b check again IP auth and sends invite request to asterisk with CSeq: 102 INVITE 4 asterisk response with 401 Unauthorized with CSeq: 102 5. Kamailio proxy b sends ACK request with CSeq: 102 ACK 6. Kamailio proxy b sends INVITE request with CSeq: 103 INVITE 7. Asterisk sends 100 trying and 180 ringing with CSeq: 103 INVITE 8. Kamailio proxy b relay 180 ringing with CSeq: 102 INVITE 9. Asterisk send 200 Ok with CSeq: 103 INVITE and proxy b relay it to proxy a with CSeq: 102 INVITE 10. Now Proxy a send an ACK with CSeq: 102 ACK and proxy b relay ACK to asterisk with CSeq: 102 ACK Now asterisk is retransmitting 200 ok with CSeq: 103 INVITE and proxy a is sending ACK with Cseq: 102 ACK Same happing in case of 487 Request Terminated Here is my config modparam("dialog", "default_timeout", 10800) # 3h modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 3 sec modparam("tm", "fr_timer", 2000) modparam("tm", "retr_timer1", 250) modparam("tm", "retr_timer2", 500) # default invite retransmission timeout after 1xx: 60 sec modparam("tm", "fr_inv_timer", 60000) modparam("tm", "reparse_on_dns_failover", 0) if (is_method("INVITE") && !has_totag()) { dlg_setflag(FLD_RTPPROXY); dlg_manage(); } route(RELAY); route[RELAY] { # At this point all checks are passed t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if(is_method("INVITE") && isflagset(FLT_FROMDID)) { #$fd = "sip.comms.kuflink.com"; t_on_failure("RELAY_AUTH"); } if (!t_relay()) { send_reply("500", "Cannot relay to $du/$ru"); } exit; } failure_route[RELAY_AUTH] { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { #$fd = "sip.comms.kuflink.com"; $avp(auser) = "DID_UA_USERNAME"; $avp(apass) = "DID_UA_PASSWORD"; if (uac_auth()) { t_relay(); } exit; } } route[WITHINDLG] { if (!has_totag()) { return; } if (is_method("BYE")) { rtpengine_delete(); } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessly route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK") ) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } send_reply("404","Not here"); exit; } kamailio -v version: kamailio 5.3.7 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 uname -a Linux example.com 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 GNU/Linux SIP logs 2020/12/17 10:08:18.483907 proxy-a.com:5060 -> 10.0.14.118:5060 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.485063 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1;rport=5060 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 2020/12/17 10:08:18.485576 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.622867 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip1.comms.kuflink.com", nonce="11139d33" Content-Length: 0 2020/12/17 10:08:18.624034 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@proxy-a.com SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK Content-Length: 0 2020/12/17 10:08:18.625557 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 103 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 Authorization: Digest username="xxxxx", realm="xxxx", nonce="11139d33", uri="sip:123456@proxy-a.com", response="qwewqeqwe", algorithm=MD5 2020/12/17 10:08:18.779483 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.937756 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.938593 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Length: 0 2020/12/17 10:08:22.075814 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.077049 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.103365 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.103877 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.79f351d031912aae585207d2f26d5650.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:22.575533 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.576777 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.581649 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.582313 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.33626b5695ad4648ba393bc62bed6a8c.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:23.575828 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:23.577863 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316

4 7

Kamailio RTP proxy require?
by Mihai Cezar 17 Dec '20

17 Dec '20

Hi, Can't figure it out when to use RTPproxy, i am forwarding everything to asterisk in seem to work without RTPproxy. The scenario is like this: Softphone -> INTERNET -> Kamailio - > Asterisk (private ip) Now it seems to work without any rtpproxy but is it ok?

5 10

DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 16 Dec '20

16 Dec '20

Hi All, I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual IP) functionality, and have run into a rather peculiar issue. *Setup:* Caller ------------ KamailioVIP(*Primary *and *Secondary* Kamailio)------- Callee HA provided by Keepalived DMQ is used between the Primary and Secondary Kamailio for dialog sync *Issue Seen:* *Suppose the Primary Kamailio has been brought down* and the Secondary one is actually active and tied to the VIP. When a call is fired from the Caller, it traverses through the Secondary Kamailio and lands onto the callee. The dialogs are updated properly. At this point, the Primary Kamailio is brought up, and dialog state is synced due to the DMQ module. The Keepalived will now attach the VIP to the Primary Instance. If the caller hangs up the phone at this point, the BYE sip message traverses through the Primary Kamailio instance to the callee and the call is cleared, but there are two issues here. 1. The Primary Kamailio throws an error in *ds_load_remove() *saying that it cannot find the load for the specific call id. This is apparently due to the fact that the dispatcher data is not synced between the two modules but dialog data is. So dialog wise, things are fine. Can this be fixed somehow? 2. The above is still as grave a problem as the dialogs are cleared. BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* kamailio even well after the call is cleared, the *DLGLOAD* shows 1. Since we are using the *Call Load based distribution* for the dispatching, this is effectively one call stuck on the dispatcher, which leads to resource leak. Is this a known issue, and if so, do we have a way to circumvent this? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 1

INFO: RTC Devroom Fosdem 2021 - Online Event
by Daniel-Constantin Mierla 16 Dec '20

16 Dec '20

Hello, (cross-posting to kamailio public groups, a few may get this message duplicated). Fosdem 2021 is going to be online, meaning that it opens the possibility for everyone to do a presentation on the RTC Devroom, no longer being required to travel to Brussels. Therefore I am sending here the link to CFP in case someone would be interested to present about Kamailio or something else RTC-related: * https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md New speakers are always welcome! Only have in mind that the event is targeting developers and source code development, therefore your proposals have to be made from this perspective. Note also that submission deadline is 20th of December, very short time frame from now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node
by Syed Shahryar 16 Dec '20

16 Dec '20

Here is my setting: modparam("dmq", "server_address", "sip:0.0.0.0:5090" ) modparam("dmq", "notification_address", "sip:0.0.0.0:5090") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 2) modparam("dmq", "worker_usleep", 0) modparam("dmq", "ping_interval", 90) I have tried localhost and I believe other options but to no avail. I also get the following message: ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:0.0.0.0:5090 How do I set up DMQ for adding and removing any Kamailio instance to/from the cluster and any Kamailio instance could serve any request coming from any client? ie. all Kamailio instances keep the current state in DMQ. I also have dmq_usrloc enabled. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Kamailio dtmf signaling
by Amit Pal 16 Dec '20

16 Dec '20

Dear Team, I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog. but can not able to capture in kamailio.cfg. how can I prepared ivr using kamalio? Please find the attached syslog. Regards Amit Pal

3 2

ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri </sip?jwt=..
by Syed Shahryar 15 Dec '20

15 Dec '20

I am sending JWT tokens via SIP URL through JSSIP for authentication before websocket connection is made. However, I am getting errors like: ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri < */sip?jwt=..* I have the following setting in kamailio config: modparam("xhttp", "url_match", "^/sip") // only match this path Is this a bug or some configuration issue? How do I resolve it? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

X-Forwarded-for IP to asterisk
by Mihai Cezar 15 Dec '20

15 Dec '20

Hello, Is it something like the XFF for Kamailio so I can see in asterisk (sip show peers) the ip of the registered softphone and not Kamailio's IP ? Thanks!

4 5

How to use port number using $env in listen
by Syed Shahryar 15 Dec '20

15 Dec '20

I tried many combinations of the following but I keep getting syntax errors: listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) TLS_IP and PUBLIC_IP are constants defined at the top using #!define. And, TLS_PORT is supposed to be the env variable. -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Losing MySQL connections
by Duarte Rocha 15 Dec '20

15 Dec '20

Greetings all, I'm having some dropped MySQL connections in my Kamailio in times of low or no activity in the proxy. These are the errors i'm getting : db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) <core> [db_query.c:348]: db_do_update(): error while submitting query usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed usrloc [ucontact.c:1663]: update_contact_db(): failed to update database registrar [save.c:784]: update_contacts(): failed to update contact sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. Is this the best approach? I've restarted my Kamailio but i don't see any ping going to my BD. Is there any other config that I need to add so that the pings are sent? Is it possible to do a ping to verify connection and reconnect mannualy on the config file? Best Regards,

2 1

Problems debugging memory issues
by Sheldon L 15 Dec '20

15 Dec '20

Hi, I'm trying to investigate a possible memory leak issue in Kamailio and I'm following the instructions indicated in https://www.kamailio.org/wiki/tutorials/troubleshooting/memory I'm using Kamailo 4.4 with the following kamailio internals: Print out of kamailio internals Version: kamailio 4.4.2 (x86_64/linux) 892ad6 Default config: /apollo/env/TachyonPlatformKamailio/etc/kamailio/kamailio.cfg Default paths to modules: /apollo/env/TachyonPlatformKamailio/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 892ad6 Compiled with: x86_64-unknown-linux-gnu-gcc 4.9.4 Compiled on: 23:36:48 Dec 8 2020 Thank you for flying kamailio! I have attached to one of the kamailio tcp receiver processes via GDB and I am trying to dump memory blocks from the memory manager. However, mem_block is not in the current_context. The following is the output from my gdb session: Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-amazon-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /opt/codedeploy-agent/deployment-root/95bb1634-e653-4cec-b327-99b18b9227a8/d-7AJ1XZP58/deployment-archive/sbin/kamailio...done. Attaching to program: /apollo/env/TachyonPlatformKamailio/sbin/kamailio, process 8946 ptrace: Operation not permitted. (gdb) set $i=0 (gdb) set $a = mem_block->first_frag No symbol "mem_block" in current context. (gdb) info list Undefined info command: "list". Try "help info". (gdb) info threads No threads. Is there something I need to do before accessing the mem_block symbol? Also do you have any other suggestions for debugging memory leaks that may not be documented? Note: i can't upgrade to the latest version of Kamailio as we have custom modules and would require considerable effort to upgrade. Thanks, Sheldon.

2 1

sngrep TLS connection
by Agiftel 15 Dec '20

15 Dec '20

Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS connections? I'am doing what help says; used private key is the correct one; but no traffic is seen on sngrep. If i use not ecrypted call i can see everything, so sngrep is working good. Any hint? regards -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

5 8

Kamilio+asterisk = deregister after 2-3 minute
by Mihai 15 Dec '20

15 Dec '20

Hi, I am forwarding all the traffic to my asterisk server ( including registers) everything seems to work when i first register but if stay idle 2-3 minute it gets unregistered and then i can't register back unless i restart kamailio see attached print screen

6 6

voltdb support
by Ali Taher 14 Dec '20

14 Dec '20

Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher

2 8

Kamailio v5.4.3 Released
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, Kamailio SIP Server v5.4.3 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.3. For more details about version 5.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/12/kamailio-v5-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Wrong To tag in t_reply
by Ivan Ribakov 14 Dec '20

14 Dec '20

Hi all, I’m trying to implement following routing behaviour: - ingress INVITE is sequentially forked to 1 or more endpoints - under certain conditions sequential forking may be stopped and pre-defined error returned to the caller Here’s is part of the config responsible for handling errors: failure_route[ROUTE_ELEMENT_FINAL_FAILURE_HELPER] { route(ROUTE_ELEMENT_FINAL_FAILURE); } route[ROUTE_ELEMENT_FINAL_FAILURE] { # We need to check if the error happens before the first egress, because T-state is not defined yet so we can not use t_reply if ($var(before_first_egress) == 0) { t_reply("$var(sipCode)", "$var(sipMessage)"); } else { sl_send_reply("$var(sipCode)", "$var(sipMessage)"); } } Here is a simple scenario where delivering call to first endpoint times out after receiving 180 Ringing (using custom fr_inv_timer value) where: - .33 is caller - .208 is Kamailio - .138 is callee Notice that 504 is retransmitted by Kamailio. Here is a comparison of 180 and 504 returned by Kamailio: As can be seen, 180 was proxied from the callee and has callee’s to-tag (tag=CALLEE1), however 504 that was generated by Kamailio using t_reply() has a different to-tag (presumably generated by Kamailio). How can I tell Kamailio not to generate new to-tag but to use the last one that proxied back to the caller? Thanks in advance for any help! Regards, Ivan

2 2

Releasing Kamailio v5.4.3
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, I am considering to release Kamailio v5.4.3 sometime next week, likely on Monday, December 14, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

how to use environment variables to set listen address
by Syed Shahryar 13 Dec '20

13 Dec '20

This is a generic question about using environment variables in a config file. If I want to use below where TLS_IP itself is a constant defined somewhere else. TLS_IP might be set via an env variable as well. #!define TLS_TO_LISTEN tls:TLS_IP:$env(TLS_PORT) Will that work? I see syntax errors and other weird issues. What's the right syntax? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

DMQ using Consul
by Syed Shahryar 12 Dec '20

12 Dec '20

Can someone share their configuration file explaining how they used consul to achieve high availability? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

empty body on first NOTIFY on REGINFO subscription - bug ?
by Adrian Tabacioiu 11 Dec '20

11 Dec '20

Hello, If the user never registered (or few hours passed from its unregister) the subscription to REGINFO will return an empty body NOTIFY (no xml). I don't see any flag in module to force a body for this case, as in the presence module is force_dummy_presence who can force a body on a first NOTIFY. The RFC3680 says : o Notifications triggered from a SUBSCRIBE contain full state (the list of all contacts bound to the address-of-record). It appears that in this case the response is not conform to RFC. Regards, Adrian

1 0

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