sr-users February 2020

sr-users@lists.kamailio.org

57 participants
76 discussions

Unexpected behavior on tmrec with monthly recurrence
by David Gonçalves 04 Mar '20

04 Mar '20

Hi, During some tests with the module tmrec, we stumble upon an unexpected behavior of the monthly recurrence with days of the week. The last week does not match our expected date. For example, the last Monday of March 2020 is day 30 (fifth Monday of the month) however, neither the 5MO or -1MO configuration on the bday parameter match with a timestamp configured to day 30. The same situation is verified on months with only 4 weeks. To easily explain the behavior we create this configuration: $var(ts)="1584355625"; # Monday, March 16, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|3MO||||","$(var(ts){s.int})")) { xlog("L_INFO","Epoch $var(ts) matches with 3MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 3MO\n"); } $var(ts)="1584960425"; # Monday, March 23, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|4MO||||","$(var(ts){s.int})")) { xlog("L_INFO","Epoch $var(ts) matches with 4MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 4MO\n"); } $var(ts)="1585565225"; # Monday, March 30, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|5MO||||","$(var(ts){s.int })")){ xlog("L_INFO","Epoch $var(ts) matches with 5MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 5MO\n"); } $var(ts)="1584960425"; # Monday, March 23, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|-1MO||||","$(var(ts){s.int })")){ xlog("L_INFO","Epoch $var(ts) matches with -1MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with -1MO\n"); } The results were: Epoch 1584355625 matches with 3MO Epoch 1584960425 matches with 4MO Epoch 1585565225 doesn't match with 5MO Epoch 1584960425 matches with -1MO Are we configuring the tmrec_match wrongly? -- Cumprimentos / Best regards, *David Gonçalves* Research and Development Technician Phone: +351 256 370 980 Email: david.goncalves(a)itcenter.com.pt

2 5

Help me open ports for my kamailio
by Stefan Troplev 04 Mar '20

04 Mar '20

Hi, I've been struggling around with my kamailio configuration. I've managed to install kamailio on Ubuntu Server and configured 2 clients, I've registered them in Zoiper to my server. They can call each other, but are unable to talk. There is no audio heard. Ports opened on the router are 20000-30000 UDP and both TCP and UDP for 5060 and 5062. Are these ports correctly opened? Thank you. kamailio.cfg > #!KAMAILIO > #!define WITH_MYSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > #!define WITH_ACCDB > # > # 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.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 > > ####### 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 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:10.0.0.10:5060 > listen=udp:192.168.1.127:5060 advertise PublicIP: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/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!=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|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 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 == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > route(RELAY); > exit; > } else { > # ACK without matching transaction ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > exit; > } > > # Handle SIP registrations > route[REGISTRAR] { > if (!is_method("REGISTER")) return; > > if(isflagset(FLT_NATS)) { > setbflag(FLB_NATB); > #!ifdef WITH_NATSIPPING > # do SIP NAT pinging > setbflag(FLB_NATSIPPING); > #!endif > } > if (!save("location")) { > sl_reply_error(); > } > exit; > } > > # User location service > route[LOCATION] { > > #!ifdef WITH_SPEEDDIAL > # search for short dialing - 2-digit extension > if($rU=~"^[0-9][0-9]$") { > if(sd_lookup("speed_dial")) { > route(SIPOUT); > } > } > #!endif > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > if(alias_db_lookup("dbaliases")) { > route(SIPOUT); > } > #!endif > > $avp(oexten) = $rU; > if (!lookup("location")) { > $var(rc) = $rc; > route(TOVOICEMAIL); > t_newtran(); > switch ($var(rc)) { > case -1: > case -3: > send_reply("404", "Not Found"); > exit; > case -2: > send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) { > setflag(FLT_ACCMISSED); > } > > route(RELAY); > exit; > } > > # Presence server processing > route[PRESENCE] { > if(!is_method("PUBLISH|SUBSCRIBE")) return; > > if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { > route(TOVOICEMAIL); > # returns here if no voicemail server is configured > sl_send_reply("404", "No voicemail service"); > exit; > } > > #!ifdef WITH_PRESENCE > if (!t_newtran()) { > sl_reply_error(); > exit; > } > > if(is_method("PUBLISH")) { > handle_publish(); > t_release(); > } else if(is_method("SUBSCRIBE")) { > handle_subscribe(); > t_release(); > } > exit; > #!endif > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # IP authorization and user authentication > route[AUTH] { > #!ifdef WITH_AUTH > > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) { > # source IP allowed > return; > } > #!endif > > if (is_method("REGISTER") || from_uri==myself) { > # authenticate requests > if (!auth_check("$fd", "subscriber", "1")) { > auth_challenge("$fd", "0"); > exit; > } > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > } > # if caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (from_uri!=myself && uri!=myself) { > sl_send_reply("403","Not relaying"); > exit; > } > > #!else > > # authentication not enabled - do not relay at all to foreign networks > if(uri!=myself) { > sl_send_reply("403","Not relaying"); > exit; > } > > #!endif > return; > } > > # Caller NAT detection > route[NATDETECT] { > #!ifdef WITH_NAT > 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==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 > 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 > } > # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=192.168.1.127 ## chrooted directory # $CHROOT_DIR="/path/to/chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database port DBPORT=3306 ## database name (for ORACLE this is TNS name) DBNAME=kamailio # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER="kamailio" ## password for database read/write user DBRWPW="kamailiorw" ## database read only user DBROUSER="kamailioro" ## password for database read only user DBROPW="kamailioro" ## database access host (from where is kamctl used) # DBACCESSHOST=192.168.0.1 ## database super user (for ORACLE this is 'scheme-creator' user) # DBROOTUSER="root" ## password for database super user ## - important: this is insecure, targeting the use only for automatic testing ## - known to work for: mysql # DBROOTPW="dbrootpw" ## database character set (used by MySQL when creating database) #CHARSET="latin1" ## user name column # USERCOL="username" # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2030-05-28 21:32:15" # DEFAULT_Q="1.0" # Program to calculate a message-digest fingerprint # MD5="md5sum" # awk tool # AWK="awk" # gdb tool # GDB="gdb" # If you use a system with a grep and egrep that is not 100% gnu grep compatible, # e.g. solaris, install the gnu grep (ggrep) and specify this below. # # grep tool # GREP="grep" # egrep tool # EGREP="egrep" # sed tool # SED="sed" # tail tool # LAST_LINE="tail -n 1" # expr tool # EXPR="expr" # Describe what additional tables to install. Valid values for the variables # below are yes/no/ask. With ask (default) it will interactively ask the user # for an answer, while yes/no allow for automated, unassisted installs. # # If to install tables for the modules in the EXTRA_MODULES variable. # INSTALL_EXTRA_TABLES=ask # If to install presence related tables. # INSTALL_PRESENCE_TABLES=ask # If to install uid modules related tables. # INSTALL_DBUID_TABLES=ask # Define what module tables should be installed. # If you use the postgres database and want to change the installed tables, then you # must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly in the # kamdbctl.base script. # Kamailio standard modules # STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo # alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher # dialplan" # Kamailio extra modules # EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist htable purple sca" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE="DB" ## control engine: RPCFIFO ## - default RPCFIFO CTLENGINE="RPCFIFO" ## path to FIFO file for engine RPCFIFO RPCFIFOPATH="/var/run/kamailio/kamailio_rpc.fifo" ## check ACL names; default on (1); off (0) # VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## check if user exists (used by some commands such as acl); ## - default on (1); off (0) # VERIFY_USER=1 ## verbose - debug purposes - default '0' VERBOSE=1 ## do (1) or don't (0) store plaintext passwords ## in the subscriber table - default '1' # STORE_PLAINTEXT_PW=0 ## Kamailio START Options ## PID file path - default is: /var/run/kamailio/kamailio.pid PID_FILE=/var/run/kamailio/kamailio.pid ## Extra start options - default is: not set # example: start Kamailio with 64MB share memory: STARTOPTIONS="-m 64" # STARTOPTIONS= >

2 3

Dynamically change the transport protocol from UDP to TCP to avoid fragmentation
by supreeth herle 03 Mar '20

03 Mar '20

Hello Everyone, I have an IMS setup using Kamailo and is working as expected when the 4G Core Network and IMS is running in a single VM. However, when running in two different VMs registration is successful but calling fails. As per the scripts in kamailio ims example folder for P-CSCF, uac module is used to generate OPTIONS message to achieve a kind of NATPINGING to UEs. The ping achieved in this way times out with 408 when IMS and Core Network is run in different VMs. I suspect that the increasing packets size could be the issue for UE not responding to OPTIONS from P-CSCF (I have verified that OPTIONS message from P-CSCF is reaching the eNB). In this aspect, I would like to switch the transport protocol to TCP when MTU is greater than 1300 (please guide me if my thinking of switching to TCP to tackle NAT issue is wrong). I have explored the following options: tcp_reuse_port=yes udp_mtu=1300 udp_mtu_try_proto=TCP But, when I use these options, any request message which hit 1300 MTU mark are not sent out at all (in my case it is the NOTIFY request from SCSCF) and Kamailio has a warning saying that binding to address failed as the address is already in use. Following is the small part of the log: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19712]: DEBUG: ims_registrar_scscf [registrar_notify.c:2275]: notification_event_process(): About to free notification Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_reply: Destination URI: <null> Request URI: <null> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (10.4.128.21:6060) Route-URI: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: ERROR: <script>: NOTIFY (tel:0198765432100 (172.24.15.30:6060) to tel:0198765432100, 4125089758_4268630712(a)192.168.101.2) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within DLG Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within loose route Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_indialog: Destination URI: sip:172.24.15.10:5060 Request URI: sip: 192.168.101.2:5060 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (172.24.15.30:6060) Route-URI: sip:mo@172.24.15.30;lr=on;ftag=4125089761 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: WARNING: <core> [core/tcp_main.c:1298]: tcp_do_connect(): binding to source address 10.4.128.21:5060 failed: Address already in use [98] I could be assuming things here about huge packet sizes causing the issue, if anyone has experienced similar issue in NATed scenario please let me know some hints or pointer to solve this. Thanks in advance. Best Regards, Supreeth

1 1

Siremis web interface issue -- siptrace module active
by Bugaian A. Vitalie 02 Mar '20

02 Mar '20

Hi list, there is an issue in latest git version of Siremis with showing sip method content. If I select it from mysql I see it all good: select * from sip_trace where method = 'INVITE'; If I click it from web interface and select needed method I get only first register from sip_trace table. You can reproduce by /siremis/sipadmin/sip_trace_list <http://sbc.pride.md/siremis/sipadmin/sip_trace_list> then click on the method to see details. I tested on opera and chrome. Same result, something wrong with javascript, its not passing correctly parameter for selecting. Please have a look let me know if I can help. Thanks. Regards Vitalie A. Bugaian

2 9

How to detect NAT during authenticated registration of clients which adjust the headers
by Awal Junanto 02 Mar '20

02 Mar '20

Hi, We are building a service where we need to detect NAT when the clients register to our server. We are struggling in analyzing NAT status of some clients which modify their IP addresses/ports in the headers according to the value of "received" parameter sent during "401 Unauthorized" response. Here's the flow: Client->Server REGISTER sip:... Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@192.168.0.1:42251;transport=TLS;ob> ... Server->Client SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias;received=1.2.3.4 WWW-Authenticate: ... ... Client->Server REGISTER sip:... Via: SIP/2.0/TLS 1.2.3.4:6201 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@ 1.2.3.4:6201;transport=TLS;ob> Authorization: ... ... By the time the client is authenticated, there is no way to detect whether the request was coming from a natted device or not by just analysing the Via or Contact headers. Thanks in advance.

3 6

Re: [SR-Users] dlg.end_dlg on RTP timeout
by Miteshkumar Thakkar 28 Feb '20

28 Feb '20

This is really helpful development. I am facing one issue, when I use this. RTPEngine is sending two "dlg.terminate_dlg" events for single call to kamailio with same call-id but reversed to-tag and from-tag in second event. I believe this is due to two separate RTP streams. Problem here is that - in first event kamailio terminates the call, hence when second event comes, kamailio says - call-id not found and send response back to RTPEngine. But still RTPEngine keeps sending that event message for three more times. Kamailio has to respond to it with call-id not found. Which is overhead. Not recommended for production environment. How can I avoid this? Any help is appreciated. Thank you Mitesh That's great, thank you! Cheers, Daniel On 06.09.18 11:15, Richard Fuchs wrote: >* This is now supported as per * >* https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701… <https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701…> * > >* Cheers * > >* On 2018-09-05 05:39, Daniel-Constantin Mierla wrote: * >> >>* There is the dlg.terminate_dlg rpc command that requires callid, * >>* from-tag and to-tag as parameters: * >> >>* * * >>* https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate… <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate…> * >> >>* So it expects something like: * >> >>* <?xml version="1.0" encoding="UTF-8"?> * >>* <methodCall> * >>* <methodName>dlg.terminate_dlg</methodName> * >>* <params> * >>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>* <param><value><string>_FROM_TAG_VALUE__</string></value></param> * >>* <param><value><string>_TO_TAG_VALUE__</string></value></param> * >>* </params> * >>* </methodCall> * >> >>* I planned to make the from-tag and to-tag optional for quite some * >>* time, but didn't get the time for it yet. * >> >>* Cheers, * >>* Daniel * >> >>* On 05.09.18 08:52, Richard Fuchs wrote: * >>>* Yup that's exactly right. * >>> >>>* It would be fairly simple to implement an additional XMLRPC format * >>>* if there's a particular one that's more friendly towards Kamailio. * >>> >>>* Cheers * >>> >>>* On 2018-09-05 02:42, Daniel-Constantin Mierla wrote: * >>>> >>>>* Looking quickly at the readme of rtpengine application and digging * >>>>* a bit with google, it is something like rtpengine has to be started * >>>>* with * >>>> >>>>* -b http(s)://myrpcserver.ip/path -x 1 * >>>> >>>>* and then the xmlrpc request is going to be sent to that url, having * >>>>* a format like: * >>>> >>>>* <?xml version="1.0" encoding="UTF-8"?> * >>>>* <methodCall> * >>>>* <methodName>teardown</methodName> * >>>>* <params> * >>>>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>>>* </params> * >>>>* </methodCall> * >>>> >>>>* Is it right? If yes, then I can try to make a sample config that * >>>>* could handle it using xmlops, xhttp and jsonrpcs modules. * >>>> >>>>* Cheers, * >>>>* Daniel * >>>> >>>> >>>>* On 05.09.18 08:24, Richard Fuchs wrote: * >>>>>* It does an XMLRPC callback. Currently there's two formats for it, * >>>>>* one is a sems sbc teardown request (using the from-tag), the other * >>>>>* is a generic "teardown" command using the call ID. * >>>>> >>>>>* Cheers * >>>>> >>>>>* On 2018-09-04 07:52, Daniel-Constantin Mierla wrote: * >>>>>> >>>>>>* Hello, * >>>>>> >>>>>>* what do you get from rtpengine on rtp timeout? An RPC call back * >>>>>>* or an http request? * >>>>>> >>>>>>* Cheers, * >>>>>>* Daniel * >>>>>> >>>>>> >>>>>>* On 04.09.18 12:48, Igor Olhovskiy wrote: * >>>>>>>* Hi all! * >>>>>>> >>>>>>>* Is there any way to end dialog in Kamailio on RTP Timeout event * >>>>>>>* on RTPEngine? * >>>>>>> >>>>>>>* Or only look at logs/redis database with external tool? * >>>>>>> >>>>>>>* Regards, Igor * >>>>>>> >>>>>>> >>>>>>>* _______________________________________________ * >>>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>> >>>>>>* -- * >>>>>>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>>>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>>>>> >>>>>> >>>>>>* _______________________________________________ * >>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>> >>>>> >>>>>* _______________________________________________ * >>>>>* Kamailio (SER) - Users Mailing List * >>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>> >>>>* -- * >>>>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>> >>> >>>* _______________________________________________ * >>>* Kamailio (SER) - Users Mailing List * >>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >> >>* -- * >>* Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> * >>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * > > >* _______________________________________________ * >* Kamailio (SER) - Users Mailing List * >* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <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 Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

2 11

Teams integration
by Bugaian A. Vitalie 28 Feb '20

28 Feb '20

Hello list, its been about one month for me playing with kamailio and I need some help to sort out a real life situation. I followed this guide https://skalatan.de/en/blog/kamailio-sbc-teams; great article, also got some inspiration from here https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/..articles look very similar. I manged to get my calls out to kamailio from TEAMS it works perfectly call gets connected and codec negotiation is fine, but I am facing issue geting the call inbound to TEAMS. My invite looks like this: ======================== U 217.26.163.205:5060 -> 52.114.75.24:5061 #22 INVITE sip:+37360844269@sip.pstnhub.microsoft.com:5061 SIP/2.0. Record-Route: <sip:sbc.pride.md:5061;transport=tls;lr>. Record-Route: <sip:217.26.163.205:5061;nat=yes;lr>. Via: SIP/2.0/UDP 217.26.163.205;branch=z9hG4bK7838.7417f50ed201fcada9609f5b7c4e520f.0. Via: SIP/2.0/UDP 192.168.169.102:5060 ;received=46.214.187.67;branch=z9hG4bK80e3f7e5cc50ea11806b6eeeb899592c;rport=5060. From: "+37379844267" <sip:+37379844267@sbc.pride.md>;tag=1604785394. To: "+37360844269" <sip:+37360844269@sip.pstnhub.microsoft.com:5061 ;user=phone>. Call-ID: 80E3F7E5-CC50-EA11-8069-6EEEB899592C(a)192.168.169.102. CSeq: 227 INVITE. sip:+37379844267@192.168.169.102:5060;gr=008A94E3-CA50-EA11-805B-6EEEB899592C;alias=46.214.187.67~5060~1Content-Type: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK. Max-Forwards: 69. User-Agent: SIPPER for PhonerLite. Session-Expires: 1800. Supported: 100rel, replaces, from-change, gruu, timer. P-Preferred-Identity: <sip:+37379844267@sbc.pride.md>. Content-Length: 362. Contact: <sip:+37379844267@sbc.pride.md:5061;user=phone;transport=tls>. . v=0. o=- 2307737351 1 IN IP4 217.26.163.205. s=SIPPER for PhonerLite. c=IN IP4 217.26.163.205. t=0 0. m=audio 36864 RTP/SAVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:owGy5+mMZNyO5+9lFmUUOK3WqpEsxJH0+jtWz =============================================== Anybody has a good invite exaple for Teams? Or do you see any issue with my invite? I do use : record_route_preset("sbc.pride.md:5061;transport=tls","217.26.163.205:5060 ;nat=yes"); add_rr_param(";r2=on"); before sending this out. Please let me know if you can help. Thanks. Vitalie.

3 7

Trouble with 302 Redirect
by Jim Leahy 27 Feb '20

27 Feb '20

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks

5 4

Rtpproxy not work in 2 kamailio + rtpproxy setup
by Voip support 27 Feb '20

27 Feb '20

Dear Community, I have a problem with rtpproxy not changing the SDP IP in connection information in following scenario: SIP1 -> kamailio1 + rtpproxy1 -> kamailio2 + rtpproxy2 -> SIP2 The first kamailio put SDP IP correctly (from rtpproxy) then the second kamailio does not change SDP connection information. I think its related to SDP body which has: a=nortpproxy:yes line added by kamailio1. So i tried to remove the a=nortpproxy:yes line from SDP before calling rtpproxy_manage. But it looks like no difference in working - i think the SDP sent to rtpproxy is still having the line even when i removed it. The issue is not happening when i remove kamailio1 + rtpproxy1. In that case my IP is put correctly in SDP in INVITE and in 200 OK. In case of two kamailio with rtpproxy (kamailio1 and kamailio2) in signalling path i am getting: rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy for replies (SIP2 IP address in 200OK SDP is present instead of kamailio2 IP address). For request INVITE i am not getting the error but the IP in SDP doesn't change. kamailio2 config is now like below (but the attempt to remove a=nortpproxy:yes, did not work for me) # 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; -- commented this out because sometimes rtpproxy was not used and i wanted to use it always # added below code to remove a=nortpproxy:yes from SDP before rtpproxy_manage but its still not working if( has_body("application/sdp") && (1 == sdp_get_line_startswith("$avp(nortpproxy)", "a=nortpproxy" )) ){ sdp_remove_line_by_prefix("a=nortpproxy:yes"); xlog("removed nortpproxy:yes from SDP"); } if(nat_uac_test("8")) { xlog("rtpproxy CO"); rtpproxy_manage("co","MY_PUBLIC_IP_HERE"); } else { xlog("rtpproxy COR"); rtpproxy_manage("cor"," MY_PUBLIC_IP_HERE "); } 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; } Best regards, Tom PS. thanks for any hints to resolve the issue.

2 1

record_route_advertised_address causes null send socket error
by Daniel Greenwald 26 Feb '20

26 Feb '20

I am using record_route_advertised_address to insert a record route header with an SRV record. The header is added correctly and everything appears to be working fine however I get a CRITICAL log error when calling record_route_advertised_address('INTERNAL_SRV'); 2020-02-25T20:14:57.078235+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket 2020-02-25T20:14:57.078352+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket Any ideas? I'm considering using insert_hf() to manually add it and bypass whatever check is failing. Thoughts? Thanks

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sr-users February 2020