- sr-users - kamailio.org

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

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Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Kamailio in cloud environments
by Alejandro Recarey 11 Nov '19

11 Nov '19

Hi, I am trying to get kamailio working cloud provider, autoscaling them behind a TCP load balancer. Is there a way for kamailio to discover its Public IP at startup? Or do people generally program their own startup scripts that modify the kamailio config file before starting it? Is there any gotchas you have hit while configuring this setup? Thank you! Alex

8 13

uac_replace_from(uri) alters display name
by Sergiu Pojoga 10 Nov '19

10 Nov '19

Hi ppl, Need to replace origination carrier's domain with my own, so I do: branch_route[MANAGE_PSTN_IN_BRANCH] { uac_replace_from("sip:$fU@mydomain.net"); } The dialog-forming INVITE is relayed properly with ONLY the domain part being replaced. However, subsequent messages such as ACK and BYE have the From DISPLAY NAME (_uac_fdp) stripped off. Not that it affects real functionality in any known way (CLID is still good for callee), but curious if this is intentional or not, as *uac_replace_from(uri)* clearly states that: "*Replace in FROM header the URI part without altering the display name*" # kamcmd dlg.list variables: { { _uac_fdpnew: } { _uac_fdp: "JOHN DOE" } { _uac_funew: sip:18661112222@mydomain.net } { _uac_fu: sip:18661112222@carrierdomain.net } Thanks!

2 4

OPTIONS request not authenticating
by Gustavo A. M. 10 Nov '19

10 Nov '19

Greetings, I'm trying to configure kamailio to get OPTIONS response, but it does not working properly. After a successful REGISTER, I send OPTIONS first time, get Proxy- Authenticate header to calculate Proxy-Authorization, and resend OPTIONS with Proxy-Authenticate header, but server follows answering with a 407 response, apparently ignoring Proxy-Authenticate. Note that same password is used in REGISTER request, and first time REGISTER response with a 401, not a 407 response. I think messages sent from client are right, and problem is on the server configuration. Follow messages and kamailio.cfg. Thanks in advance, Gustavo Adolfo Moennich OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 0 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 0 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.467d Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905;received=192.168.122.12 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1wV3HNJXdfUDchNlEUkT6kHCtqBAn" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 1 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 Proxy-Authorization: Digest uri=sip:6000@kamailio.memorea.com,nonce="Xc c1wV3HNJXdfUDchNlEUkT6kHCtqBAn",realm="kamailio.memorea.com",response=" 5bcd184eea53284f3ec548c2412ce36a",username=6000 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 1 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.32de Via: SIP/2.0/UDP 192.168.122.21:5060;received=192.168.122.12;rport=58905 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1xl3HNJqRobJFuZggPSGvNdtp+GnP" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 kamailio.cfg content's: #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.or g> # # 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:temp@database.memorea.com/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif 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_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=$si) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (uri==$ru) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == $ru) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (is_method("OPTIONS") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("OPTIONS|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!=$fu && uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==$ru) 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}$")) return; # only local users allowed to call if(from_uri!=$fu) { 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 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 }

2 1

dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | 1 | sips:10.0.0.100:5061 | 0 | 0 | transport=tls | SIP SERVER | +----+-------+------------------------+-------+----------+---------------+----------------+ Now when REGISTER is received over websocket, kamailio is responding with error code 500 and phrase "500 I'm terribly sorry, server error occurred (7/SL)". And on the console I see the following error messages. 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: f1ecf7bcb659b07fe81e332e100044e5 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [ut.h:315]: uri2dst2(): no corresponding socket found for "10.0.0.100" af 2 (tls: 10.0.0.100:5061) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:467]: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -7 (-7) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:352]: t_relay_to(): -7 error reply generation delayed 12(33858) exec: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=587 a=24 n=sl_reply_error 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/msg_translator.c:162]: check_via_address(): (10.0.0.14, hsvmphm3ps12.invalid, 0) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} websocket [ws_conn.c:452]: wsconn_get(): wsconn_get for id [3] *tls.cfg contents* [client:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /home/test/kamailio/internal.key certificate = /home/test/kamailio/internal.crt ca_list = /home/test/kamailio/ca_list.pem Any reason why this error is seen? Any inputs appreciated. Thanks.

4 20

Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) Regards, Konstantinos > On Fri Nov 8 10:24:06 CET 2019 Daniel Tryba wrote: > >> On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote: >> >> I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: >> > You appear to do the same thing I do, except setting $du to null. If the user is REGISTERed $du will be set to their endpoint, changing $ru will have no effect on routing. So add $du=$null; in the TOVOICEMAIL route before route(RELAY) and see what happens. >

3 4

_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: 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")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } 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")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!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; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif }

2 1

How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738" <sip:38617774738@10.1.1.1>;tag=105854496. To: <sip:562462354464@192.168.1.242>. Call-ID: 1573138018-1602381967(a)10.1.1.182. CSeq: 20215490 INVITE ..... ETC ETC... Thanks to all

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES it stops handling SRTP correctly to the encryption upstream. And i am sure, its my fault. But i have no clue how to detect that correctly. Any hints where i can read more about an correct handling on this? Here my main-config file / kamailio 5.3.x and rtpengine. https://gist.github.com/khorsmann/02bd98a628a190caee68fc3d876a3448 -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by direct database queries in usr_preferences on INVITEs and REGISTERs. You could use any database (depending on how you store the ACL) and cache it in a htable. My solution is mysql specific (inet_aton) and forces users to have atleast 1 CIDR notated subnet: if(!avp_db_query("select value from usr_preferences where username='$au' and attribute='acl' and inet_aton(substring_index(value,'/',1))&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)=inet_aton('$si')&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)")) { t_reply("404","Go away"); exit; }

3 2

MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla mongo client. How can I limit the result? Thanks Lars

2 3

Group module
by Igor Olhovskiy 06 Nov '19

06 Nov '19

Hi! I'm trying to solve quite simple task and want to solve it as simple as possible. Idea is to have restriction to allow register certain username/domain only from certain subnets. So, I decided to use group and permissions modules. But stuck on group one. Code is fairly simple modparam("group", "db_url", DBURL) modparam("group", "use_domain", 1) ... if (get_user_group("From", "$avp(user_groups)")) { xlog("$fu from $si:$sp is matched group $avp(user_groups)\n"); if (allow_source_address("$avp(user_groups)")) { ...allow furter } } else { xlog("$fu from $si:$sp is not matched any groups\n"); } grp table is looks like kamctl db exec 'select * from grp' +----+-------------+-------------------+-----+---------------------+ | id | username | domain | grp | last_modified | +----+-------------+-------------------+-----+---------------------+ | 1 | 61388140395 | sip.localhost.net | 5 | 2000-01-01 00:00:01 | +----+-------------+-------------------+-----+---------------------+ But actual line of debug in logs saying this kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=720 a=26 n=get_user_group kamailio_1 | 10(18) DEBUG: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} group [re_group.c:188]: get_user_group(): getting groups for <sip:61388140395@sip.localhost.net> kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=727 a=25 n=xlog kamailio_1 | 10(18) ERROR: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} <script>: sip:61388140395@sip.localhost.net;transport=UDP from 172.21.0.1:56787 is not matched any groups What am I missing here? Or module is too old? Thanks!

1 0

RFC: default config with jsonrpc over http instead of xmlrpc
by Daniel-Constantin Mierla 06 Nov '19

06 Nov '19

Hello, I think it was discussed during past IRC meetings or at some events across the world, with jsonrpc being these days more popular than xmlrpc, specially due to easier integration with web based apps, I am considering to replace the example of using xmlrpc over http(s) with jsonrpc in the default configuration file. It is not enabled by default for xmlrpc (requiring to define WITH_XMLRPC), so it will stay also as a define option for jsonrpc. The jsonrpcs module is already loaded for kamctl, it will require the xhttp module to be added. As for those looking for xmlrpc in the future, there is another example kamailio cfg which has the option with it: misc/examples/pkg/kamailio-oob.cfg Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 3

$jsonrpl(body) empty or malfunctioned during higher traffic
by Michal Škuta 05 Nov '19

05 Nov '19

Hello, Kamailio community, we are implementing evapi communication to our Kamailio server. We have implemented and tested it on our testing server. Everything was working OK. Now we are testing it on our production servers with a lot more calls and we have some problems here. Every n seconds we are requesting from Kamailio list of all dialogs. And this list of dialogs (from jsonrpc *dlg.list*) is sent to evapi connection. Here is a snippet of our *Kamailio code*: route[CGR_DLG_LIST] { xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Requested dialog list"); # CGRateS connection is still alive, set check to true $sht(cgrconn=>check) = 1; if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Test1: $jsonrpl(body)"); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); } During a higher number of calls Kamailio is generating ERROR messages from evapi module. Here is the syslog: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [ Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_dispatch.c:764]: _evapi_relay(): cannot serialize event Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_mod.c:265]: w_evapi_relay(): failed to relay event: *We are expecting something like this when nobody is calling* Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Test1: { #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 } Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }}] (96) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:778]: _evapi_relay(): sending [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:789]: _evapi_relay(): dispatching [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) We enabled debug messages for module evapi and are trying to understand where is the problem. To make this work to this state we needed to increase the *pv_buffer* to really big size (2097152). It looks like $jsonrpl(body) is not ready and everything concated with it is not working. Any help or suggestion is welcomed. Sincerely Michal

3 6

Module acc - How to get sip_reason/code on log_write_cdr() messages
by David VILLAUME 05 Nov '19

05 Nov '19

Hello, I'm trying to get the reason in my cdrs in log_write_cdr() messages while I see It in acc_log_request() messages. Is there a way to retrieve it in log_write_cdr() messages ? Bellow is my config : modparam("acc", "log_level", 1) modparam("acc", "log_facility", "LOG_LOCAL3") modparam("acc", "cdr_log_enable", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 2) modparam("acc", "log_missed_flag", 3) modparam("acc", "report_cancels", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_facility", "LOG_LOCAL3") modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_extra", "call_id=$ci;src_ip=$si;fU=$fU;")" David VILLAUME

2 2

Re: [SR-Users] driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 04 Nov '19

04 Nov '19

Hello Henning, is there an planed release date for this 5.3.x version? Unfortunately without this change i cant deploy it without changing all my mysql user credentials to be sure there is no "kamailio DBURL separator" within the password field. Cheers Karsten [...] -- Mit freundlichen Grüßen *Karsten Horsmann*

3 2

kamailio - failover setup
by Youssef Boujraf 04 Nov '19

04 Nov '19

Dear All, I have seen in the new kamailio release 5.3.0, kamailio support for haproxy protocol – both the v1 (human-readable) and v2 (binary) versions of the protocol are supported ( https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ ) but it seems to be for Database, webserver failover not for kamailio itself. Haproxy is ONLY supporting TCP but not UDP. My question is : I am looking for failover service of kamailio servers looks like haproxy but udp & tcp. - one domain name : sip.secure.com pointing my public ip address. - two kamailio server pointing same database content (user, whitelist, ...) no asterisk or pbx. Kamailio is used Only for signaling (users, whitelist & rtp) Monitoring service : identify who is the kamailio server online, offline and the ability to force all the SIP/RTP/MANAGEMENT connection from Kamailio Srv1 to Kamailio Srv2 and vice-versa ? Thanks for your suggestions. -- Youssef

4 4

How to use sdp_remove_media() ?
by Denys Pozniak 04 Nov '19

04 Nov '19

Hello! I'm trying to remove m=video stream from incoming SIP/SDP request in th way below: modparam("rtpengine", "read_sdp_pv", "$avp(read_sdp_pv)") modparam("rtpengine", "write_sdp_pv", "$avp(write_sdp_pv)") ... sdp_remove_media("video"); msg_apply_changes(); sdp_get("$avp(read_sdp_pv)"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| rb=/$rb/ \n"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| avp(read_sdp_pv)=/$avp(read_sdp_pv)/ \n"); rtpengine_manage("$avp(rtpengine_offer)"); set_body("$avp(write_sdp_pv)", "application/sdp"); But no luck, xlog shows m=video in $rb and $avp(rtp_sdp_read) as well. -- BR, Denys Pozniak

3 8

Phonenum module
by Igor Olhovskiy 01 Nov '19

01 Nov '19

Hi! I'm playing around libphonenum to get validation accross e.164 numbers. So, idea is simple if (phonenum_match($fU, "src_num")) { if ($phn(src_num=>valid) != 1) { send_reply("400", "Invalid e.164 format"); xlog("L_ALERT","ALERT: FU: $fU is not e.164 number $phn(src_num=>error)\n"); exit; } } But on every (actually valid e.164 number like 380442459966 or 61388141397) I got Parsing number failed in error result. Tried both phonenum_match($fU, "src_num") and phonenum_match("$fU", "src_num") Actually phonenum_match("380442459966", "src_num") also not working Debian 10 Buster version: kamailio 5.2.5 (x86_64/linux) libphonenumber7 is already the newest version (7.1.0-5+b4).

2 6

kamailio on kubernetes
by Gaurav Bmotra 01 Nov '19

01 Nov '19

hello everyone i m working on a project kamailio on kubernetes for high availability and auto scalability but as i m able to run it in side the cluster with using kubenetes feature but having issue to access the service form outside the cluster i m using the ingress in kubernetse to point the service its flow like that domain.com:80 ------ingress-controler--------kamailio-service:80:5060------- POD(kamailio)5060 ISSUE IS I M NOT GETING CONNECTION BITWEEN DOMAIN.COM TO POD(kamailio) is there any one ho can help me in this thank you -- *Regards:* Gaurav Kumar

2 1

Routing Blocks - Missing in latest core wiki
by Joel Serrano 31 Oct '19

31 Oct '19

Hi, I was looking at the documentation and noticed that "Routing Blocks" section is missing in latest releases... http://www.kamailio.org/wiki/cookbooks/5.2.x/core#routing_blocks is OK But: http://www.kamailio.org/wiki/cookbooks/5.3.x/core#routing_blocks http://www.kamailio.org/wiki/cookbooks/devel/core#routing_blocks Don't exist, have they been moved elsewhere? Thanks! Joel.

3 8

Command Line SIP client
by Gaurav Bmotra 31 Oct '19

31 Oct '19

hi i m looking for Command Line SIP client , where i can register user form CLI for testing thank you -- *Regards:* Gaurav Kumar

9 11

How to intercept an INVITE after waking up (TSILO module) to change SDP
by Oleg Podguyko 31 Oct '19

31 Oct '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using the module rtpgngine I change SDP. Webrtc <-> SIP calls work fine. RTPengine proxies media. It is necessary to understand that when the call comes, I determine what Protocol the call came wss/tcp/udp, and then through the command "rtpengine_manage" change SDP accordingly. Now I have a new task. Wake up mobile customers through push notification. This client may be as WebRTC and also regular SIP. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { route(SEND_PUSH); xlog("L_INFO", "tsilo store\n"); ts_store(); } } switch ($var(rc)) { case 1: xlog("L_INFO", "LOCATION, Subscriber found, RC: $var(rc)\n"); route(RELAY); break; case -1: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); exit; break; case -3: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); send_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", "LOCATION, Method Not Allowed, RC: $var(rc)\n"); send_reply("405", "Method Not Allowed"); exit; } } If a call occurs when the called client is registered, then we go to the route(RELAY), which creates a branch in which I change the SDP. Everything works. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (!t_relay()) { sl_reply_error(); } exit; } And already in MANAGE BRANCH the necessary SDP is put. If the called party is not registered. In the beginning everything goes well. I find this caller through a separate server. Send push notification. The client successfully registers on kamailio. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if(isflagset(FLT_SRC_WSS)) { setbflag(FLB_DST_WS); } if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu") exit; } Next I better show a piece of the log: окт 31 14:55:08 /usr/sbin/kamailio[17488]: exec: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=429 a=26 n=ts_append окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tsilo [ts_append.c:72]: ts_append(): transaction 8800:338205132 found for 1234567890, going to append branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:255]: lookup_helper(): contact for [1234567890] found by address окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:456]: lookup_helper(): instance is окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:77]: t_append_branches(): transaction 8800:338205132 in status 0 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:99]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (1) outgoing branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/msg_translator.c:162]: check_via_address(): (50.7.93.27, uas7i396ceb5.invalid, 0) окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f] with ruid [uloc-5dbacb09-4450-7] окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:163]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (0) outgoing branches after clear_branches() окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (17488 (21)), reusing fd окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2507]: tcpconn_do_send(): after real write: c= 0x7fb39dde4c68 n=3131 fd=12 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2508]: tcpconn_do_send(): buf= Thus, the transaction wakes up and the initial INVITE is sent to the new registered client without any SDP change! Which eventually leads to 415 error. Now the question itself. Tell me! As me at this moment intercept this INVITE, to it has passed through in advance prepared path for changing SDP. Thank you. -- Oleg Podguyko

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