sr-users November 2020

sr-users@lists.kamailio.org

52 participants
65 discussions

The Algorithm “13” - latency optimized dispatching

by Julien Chavanton

About The Algorithm “13” - latency optimized dispatching, Is now reviewed once and tested, it will most likely be ready to merge soon. I want to share my thoughts on it one more time as it is not too late to get more feedback before we merge. I think it is the best algorithm in most use cases, here is why : It is providing round-robin and fail-over with automatic de-prioritization of slow/unresponsive gateways. You probably asked yourself the following questions in the past : "How do I set the thresholds to put a gateway out of service ?" *ds_probing_threshold*, *ds_inactive_threshold* and timers ... - If your thresholds are too strict, you may end up running out of gateway. - If your thresholds are too tolerant, you may end up adding excessive delays to call establishment and using degraded gateways. The automatic de-prioritization can help to address this concern more efficiently by providing more flexibility. - it can react faster than lets say 2 consecutive timeouts. - it will not disable gateways but simply de-prioritize / reorder them if needed. The only main drawback I can imagine is when you always need to evenly distribute calls using round-robin. It may be needed sometimes but in this case it means you are willing accept to send calls to a degraded gateway or trough degraded network paths. Even if you may select to preset a mixture of round-robin sets, thanks to *ds_select_routes* however it will stay static, needs to be configured precisely, and will not react to degradation automatically. I hope this will help use to protect QoS and lower latency of calls routed by Kamailio. Feel free to let me know what you think Julien

3 years, 9 months

rport parameter in VIA header in uac remote registration

by Jetse Verschuren

Hi, I'm currently trying to connect Kamailio, running in docker, to my FritzBox. The FritzBox is the modem provided by my ISP, with a builtin voip server which I can use to dial out (basically a PSTN gateway). In my setup I have a custom router/firewall/NAT between the FritzBox and the server. When I connect LinPhone to the FritzBox, from behind the NAT, it works fine. But when I use the Kamailio uac remote registration, I don't get a response. After some investigation with wireshark, I found that the the SIP response from the FritzBox comes on the wrong port, and gets rejected by the NAT. The request comes from port 1025, but the response is sent to port 5060. I compared the requests from LinPhone and Kamailio, and combined with some research, found it probably was the rport option in the VIA header. >From what I read, it basically signals the uas to use the source port to reply to, instead of the port indicated in the request, which would be necessary to effectively go through a NAT. After reading a lot of documentation (and even parts of the source), I wasn't able to find an option to enable this option. Is there some option I completely missed, or is it not possible? Or am I completely wrong in my "diagnosis"? Thanks in advance, Jetse

3 years, 10 months

Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher

by Usama Shaikh

Hello Everyone. I am having trouble to setting up kamailio for load balancing for asterisk servers. I am using module dispatcher for load balancing AND two different servers for kamailio and asterisk. I am doing it first time and have not exact idea about its all configs… but i have researched and done so far, now need help from you guys. Currently, trying to setup with the one kamailio and one asterisk then will go ahead if I get any success… using MariaDB MySQL for the dispatcher data. Simply I want to do is : 1. Register my asterisk devices with kamailioIP (and I am able to do this using below configs) Devices are getting register with the asterisk from with kamailioIP 2. I want to receive calls on the registered devices with KamailioIP (and i am not able to do this) WORK FLOW: KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060 SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load balancer)--------->>> ASTERISK SERVER Please have a look into my configuration files and let me know if anything is missing or wrong in these. Or anything to add or remove from the confs… my kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #define WITH_NAT #!define WITH_DEBUG #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://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 ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 log_name="kamailio" children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:0.0.0.0:5060 #listen=udp: KamailioIP:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = " KamailioIP .95" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!else mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "sctp.so" loadmodule "dialog.so" loadmodule "siptrace.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" loadmodule "dispatcher.so" loadmodule "jsonrpcs.so" loadmodule "htable.so" # ---------------------- mod dispatcher params ------------------------------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_ping_from", "sip:dispathcer@ KamailioIP ") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_method", "INFO") #modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;code=481;class=3") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") #modparam("dispatcher", "priority_col", "dstpriority") #modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # do failover modparam("dispatcher", "flags", 2) #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #!endif # ----------------- setting module-specific parameters --------------- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ---------------------- mod uac params ------------------------------ modparam("uac", "reg_contact_addr", "ASTERISKIP:5060") modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)") modparam("uac","auth_username_avp","$avp(s:auth_username_avp)") modparam("uac","auth_password_avp","$avp(s:auth_password_avp)") # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "ruri_matching", 0) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "db_insert_update", 1) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- 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", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes #modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "stats=>size=6;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- #modparam("debugger", "cfgtrace", 1) #modparam("debugger", "breakpoint", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(DISPATCH); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(is_method("NOTIFY")) { sl_send_reply("200","OK"); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio append_hf("FromIp: $src_ip\r\n", "Call-ID"); route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } if (is_method("REGISTER") || from_uri==myself){ if (!allow_trusted()) { xlog('wokred here for ipauth check'); sl_send_reply("403", "Forbidden"); } } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_users", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if(t_check_status("401")) { uac_auth(); } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } t_relay(); #!endif } # Test if coming from Asterisk route[FROMASTERISK] { # if($si==$sel(cfg_get.asterisk.bindip) # && $sp==$sel(cfg_get.asterisk.bindport)) # return 1; # return -1; # dispatch: is this asterisk ip ? if (ds_is_from_list("1")) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { # $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" # + $sel(cfg_get.asterisk.bindport); # append_hf("X-Asterisk-Context: dexter-phones\r\n"); # route(RELAY); # exit; # select asterisk port append_hf("X-Asterisk-Context: dexter-phones\r\n"); route(DISPATCH); } route[DISPATCH] { # round robin dispatching on gateways group '1' #xlog("<<<<<<<<<<<<<<<<trying for dispatch>>>>>>>>>>>>>>>>>>>"); #if (is_method("INVITE")) { # dst_select( "GROUP", "HASH METHOD") #ds_select_dst("1","4"); #sl_send_reply("100","Trying"); # forward(); # exit(); #} if (!ds_select_domain("1", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } “kamctl dispatcher dump” looks like this: { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:ASTERISKIP:5060", "FLAGS": "AP", "PRIORITY": 1, "ATTRS": { "BODY": "rweight=50;weight=50;cc=1", "DUID": "", "MAXLOAD": 0, "WEIGHT": 50, "RWEIGHT": 50, "SOCKET": "", "SOCKNAME": "", "OBPROXY": "" } } }] } }] }, "id": 13403 } And also have set trunk in ASTERISK sip.conf register => kamailioSIP:XXXX@KamailioIP:5060 [kamailio] type=friend host= KamailioIP port=5060 transport=udp allow=all allow=gsm allow=alaw,ulaw insecure=invite,port Permit = 0.0.0.0/0.0.0.0 sipdebug=yes username=kamailioSIP canreinvite=no secret=XXXX qualify=yes dtmfmode=auto when i Register my asterisk devices with kamailioIP (and I am able to do this) Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here 8101 and 8100 is registered with Kamailio IP ) [Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060 [Nov 9 10:36:02] -- Unregistered SIP '8101' [Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060 [Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1 [Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 9 10:36:07] -- Unregistered SIP '8100' [Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060 [Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100 [Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer '8100' is now Reachable. (51ms / 2000ms) when i dial 8100 to 8101 i get in my asteris -r NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101' (95.217.223.95:5060) to extension '8100' rejected because extension not found in context 'none-dial'. and not able to call extension to extension, PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no success still kindly guide me if anything I am missing to do in the process or anything I am doing wrong. I will be eagerly waiting for any help or pointer… Thanks in advance

3 years, 10 months

Re: [SR-Users] Checking user registration

by ahmed moghazy

When i used lockup location it is done but if the user registered one time on server the location will be saved on kamailio so i want a method to check registered users live every second for example and i want to delete locations of un registered users ? Thank u. > On 10/11/2020, at 12:41 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

3 years, 10 months

Checking callee registration

by ahmed moghazy

I want to check if callee is registered or not before routing call because this made error and make caller unregistered untill i restarted the linphone

3 years, 10 months

Checking user registration

by ahmed moghazy

I want kamailio to reject call and send message user is not registered when a caller is trying to call unregistered user (callee) , Thank you.

3 years, 10 months

Expire timeout for notify message answer

by Victor Litvinov

Hello dear community. We use a presence module with the enabled option "timeout_rm_subs". And we face issues that subscriptions remove too early. We expect that it will be removed after the fr_timer, but it is removed after 1 second. We I can see in log (i'm add additional logs): when we send NOTIFY: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309038 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:536]: retr_buf_handler(): new interval 4000 ms / 64 ticks (max 4000 ms) when timer expire: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309054 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:367]: final_response_handler(): transaction [0x7f4708f458a0 <-> 11592:145228527] scheduled for deletion tm [timer.c:424]: final_response_handler(): stop retr. and send CANCEL (0x7f4708f458a0) and then subscription remove. As we can see the fr_expire variable was set to ticks+16 ( 1 second). And another strange thing that T1 and T2 timers are the same. So the question is if this behaviour is correct, how can we configure the presence module fr_timer as an expiration value for the answer to notify? And can we use it without retransmissions?

3 years, 10 months

How to modify INVITE Request Line string

by Jose Figueroa

Hello everybody, Today I received a request from a carrier which needs us to send the calls using a prefix in the INVITE header. I am reading the documentation for kamailio 4.3.x but I havent found anything I can use in order to modify the Request Line (INVITE) string. I need to change from this: INVITE sip:123456789@IP:5060 to this: INVITE sip:PREFIX123456789@IP:5060 If you can guide me to the right direction would be more than great. Thanks -- Jose Figueroa Senior Software Engineer

3 years, 10 months

(no subject)

by ahmed moghazy

Hello, When i make a call from specific user to another it didnt make call and it gave missed call to another one after a minute so the call doesnot complete i dont know why , Thank you.

3 years, 10 months

Kamailio ims stable version

by Pavithra Mohanraja

Hi, How to do git clone of kamailio source code with minor version. I could only do git checkout of 5.3 version and it is automatically taking latest stable version. I don't need in this way. I need to do git clone of exact version of kamailio source code . Example - if I need 5.3.6 version.. exact version source code has to be git checked out from kamailio website. How to do that .. kindly let me know.. Thanks, Pavithra

3 years, 10 months

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sr-users November 2020