sr-users June 2017

sr-users@lists.kamailio.org

79 participants
124 discussions

two way routing
by Jean Cérien 19 Jun '17

19 Jun '17

Hello I am looking at the following configuration: 1/ voip providers, with different external sip servers, will send calls to kamailio, that will dispatch them to asterisk servers, with load balancing & failover. The IP of the asterisk servers are known and fixed. 2/ the asterisk servers will also place calls, going through kamailio, that will dispatch them to the voip provider sip servers, based on R-URI / Prefix. I've achieved #1 with dispatcher module, I've seen carrierroute module that can achieve #2, but I dont really grasp how to put the two together. I have the feeling that #1 can also be done with carrierroute, and that I only need to check if the 'from' domain is one my asterisk, and chose a different routing table, but I dont see how to do that is that a valid approch ? where should I look to do that ? thanks J.

3 7

Multiple FreeSWITCH servers behind kamailio-websocket
by Karsten Horsmann 19 Jun '17

19 Jun '17

Hello List, is there any howto about webrtc loadbalance in combination with kamailio and FreeSWITCH? I want to share one WSS address/endpoint to multiple FreeSWITCH backends. Or is there any other best practice? My callflow is mostly that my internal SIP Servers called my registered webrtc clients. Would be nice to get some input. -- Kind Regards *Karsten Horsmann*

5 7

How to change MSRP from sip address?
by Pranathi Venkatayogi 19 Jun '17

19 Jun '17

I need to replace the "From" address of MSRP message with something different. I noticed that module does not allow one to set values from config file. What does it take to enable this functionality? Does anyone have a prototype I can leverage? Need to replace from domain with some other domain - Given: MSRP body is - From: cust3 <sip:cust3@devtranslation.sms-test.cyracom.com>#015#012To: <sip:agent1@xmpp.xgw.cyracomdev.com>#015#012DateTime: ..... Wanted: MSRP body is - From: cust3 <sip:cust3@new domain >#015#012To: <sip:agent1@xmpp.xgw.cyracomdev.com>#015#012DateTime: ..... Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

Call and Voice issues
by Momen Amin 18 Jun '17

18 Jun '17

Description Hello! I'm beginner, and i have installed Kamailio Proxy SIP Server on Ubuntu 14.04.5 LTS and using it to register many SIP Servers (Asterisk) I have some issues: 1- When i call another extension in not ringing. 2- Sometimes ringing but when call is get answered there is no voice. 3- I need to receive inbound calls from my Trunks, but i did not know how to make my proxy able to manage it. 4- I need to call some futures codes like (*45 & 555) but it is unable to call. Thanks to all My "kamailio.cfg": #!define WITH_TLS #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_PSTN # Kamailio (OpenSER) SIP Server v5.0 - 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/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 /* 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=yes /* add local domain aliases */ /* 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:127.0.0.1:6000 listen=udp:XXX.XXX.XXX.XXX:6000 listen=tls:XXX.XXX.XXX.XXX:5061 /* port to listen to (default 5060 for udp, tcp, scrtp, or 5061 for tls)*/ # 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 = "XXX.XXX.XXX.XXX" desc "PSTN GW Address" pstn.gw_port = "5060" 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/local//lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_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", "/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;") #!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 requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); 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); } # 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; } } if($ua =~ "friendly-scanner|sipcli") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } #!endif 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("1511", "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 statelessy 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; } #!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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($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 }

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Strange issue in registering users in Kamailio
by Mojtaba 17 Jun '17

17 Jun '17

Hello guys, Finally i found what was my problem. It was about ARP table size, becuase is set to 1024 record in ubuntu. when i try to register a lot of users with different source ip to kamailio (or other sip server like asterisk and etc), after some minutes, the arp table is fulled and these warrning may be issue: In kamailio: Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31864]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2518774590 r: 0x7f2518774610 Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31859]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f2518d4ebf0 r: 0x7f2518d4ec70 Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.13.227]:59948 -> Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.0.221]:5060) Jun 14 16:06:50 172-1-1-158 /usr/local/kamailio-5.0/sbin/kamailio[31868]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f25191387c0 r: 0x7f2519138840 J In Asterisk: Jun 16 00:59:09 asterisk11 kernel: [ 5042.988477] Neighbour table overflow [Jun 16 00:59:14] WARNING[2811] chan_sip.c: sip_xmit of 0x7f94fd08a4e0 (len 488) to 192.168.14.116:5060 returned -1: No buffer space available you should add some parameters in sysctl.conf in ubuntu: net.ipv4.neigh.default.gc_thresh1 = 512 net.ipv4.neigh.default.gc_thresh2 = 1024 net.ipv4.neigh.default.gc_thresh3 = 2048 Good luck With regards.Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S

1 0

Kamailio v4.4.6 Released
by Daniel-Constantin Mierla 16 Jun '17

16 Jun '17

Hello, Kamailio SIP Server v4.4.6 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.6. For more details about version 4.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/06/kamailio-v4-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.0, at this moment with its latest release v5.0.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Planning Kamailio v4.4.6
by Daniel-Constantin Mierla 16 Jun '17

16 Jun '17

Hello, I backported a consistent set of patches to branch 4.4 and plan to release v4.4.6 in the next days. Should anyone be interested in that branch, it would be good to review and see if any relevant fix from newer branches is missing in order to assert the feasibility of backporting. If time allows, 4.4.6 can be out as soon as tomorrow afternoon, if not, sometime during following days. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 3

Re: [SR-Users] Topos: contact header is not updated avec RE-INVITE
by Sergey Basov 16 Jun '17

16 Jun '17

You are welcome. -- WBR Sergey Basov 15 июня 2017 г. 10:13 PM пользователь "Abdoul Osséni" < abdoul.osseni(a)gmail.com> написал: > Hello, yes, you were right. > > There was something wrong in my routing config. > See below, I remove the bad config. > > Thank you very much > > route[WITHINDLG] { > > …… > > …. > > 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 statelessy > > route(NATMANAGE); > > } else if ( is_method("NOTIFY") ) { > > # Add Record-Route for in-dialog NOTIFY as per RFC > 6665. > > record_route(); > > } > > else if ( is_method("INVITE") ) { > > # even if in most of the cases is useless, do RR > for > > # re-INVITEs alos, as some buggy clients do change route > set > > # during the dialog. > > xlog("Route with in Dialog: loose route $rU -> $fU\n"); > > record_route(); > > } > > route(RELAY); > > exit; > > } > > …… > > …. > > } > > > Regards > > > 2017-06-15 16:27 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > >> Do you only change topoh to topos in your config? >> >> You can find my dump attached. There is duplicate packets this is issue >> of my capture architecture.. >> >> -- >> WBR >> Sergey Basov >> >> 15 июня 2017 г. 5:20 PM пользователь "Sergey Basov" < >> sergey.v.basov(a)gmail.com> написал: >> >> Its very strange... >> I have record routes from 1 side and from another side, they are stripped >> and restored normaly. >> >> On reinvite you have incorrect contact restored. >> >> I will look at my config. >> >> -- >> WBR >> Sergey Basov >> >> 15 июня 2017 г. 5:09 PM пользователь "Abdoul Osséni" < >> abdoul.osseni(a)gmail.com> написал: >> >> The record route is missing with topos module. Maybe it is that the >>> problem? >>> Regards. >>> >>> 2017-06-15 16:04 GMT+02:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>> >>>> >>>> I tried with an another uac (without +sip.instance="<urn:uuid:14d62 >>>> d4f-8426-486b-88ee-077b9ff1fae2>" in contact). Same issue. >>>> >>>> I checked my routing logic in Kamailio. Same configs describe ici : >>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri >>>> sk-11.3.0-astdb?s[]=asterisk. >>>> >>>> >>>> [image: Images intégrées 1] >>>> >>>> Regards >>>> >>>> Abdoul. >>>> >>>> 2017-06-15 14:34 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> >>>>> I checked in my environment all ok. >>>>> Can you try without +sip.instance="<urn:uuid:14d62 >>>>> d4f-8426-486b-88ee-077b9ff1fae2>" in contact? >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> 2017-06-15 12:52 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>>>> >>>>>> >>>>>> Please find in attachment. >>>>>> >>>>>> - topos module is disabled >>>>>> - topoh module is enabled >>>>>> >>>>>> I have no issue -- the contact header is updated. >>>>>> >>>>>> I have only this issue when topos module is enabled but I will check >>>>>> again my routing logic in kamailio. >>>>>> >>>>>> Regards >>>>>> >>>>>> 2017-06-15 11:39 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>>> >>>>>>> I think you may have some error in your routing logic in kamailio... >>>>>>> >>>>>>> Can you try to disable topos module and try with topoh to see does >>>>>>> it handles contact modification properly? >>>>>>> >>>>>>> -- >>>>>>> Best regards, >>>>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>>>> >>>>>>> 2017-06-15 12:07 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com>: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Please find in attachment the pcap file. >>>>>>>> >>>>>>>> Regards >>>>>>>> >>>>>>>> Abdoul >>>>>>>> >>>>>>>> 2017-06-15 9:43 GMT+02:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>>>>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> Can you send dump of the both side of call. >>>>>>>>> I using this module into production I have a lot of Re-Invites and >>>>>>>>> does not have this problem. >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Best regards, >>>>>>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>>>>>> >>>>>>>>> 2017-06-15 10:18 GMT+03:00 Abdoul Osséni <abdoul.osseni(a)gmail.com> >>>>>>>>> : >>>>>>>>> >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have an issue when loading topos module on Kamailio 5.0.2 >>>>>>>>>> version. >>>>>>>>>> >>>>>>>>>> root@proxy:/home/tcpdump# kamailio -V >>>>>>>>>> version: kamailio 5.0.2 (x86_64/linux) >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> The call flow is: uac --> kamailio --> Asterisk >>>>>>>>>> >>>>>>>>>> 1) Invite from uac >>>>>>>>>> 2) Kamailio forward the invite to asterisk >>>>>>>>>> 3) asterisk send 200 OK to kamailio >>>>>>>>>> 4) kamailio send 200 OK to uac -> in the conctact header, i can >>>>>>>>>> see the IP address of kamailio >>>>>>>>>> 5) uac send ACK to kamailio and kamailio forward the ACK to >>>>>>>>>> Asterisk >>>>>>>>>> 6) uac send re-invite: "media change" to kamailio. >>>>>>>>>> 7) kamailio forward the re-invite to asterisk and asterisk send >>>>>>>>>> 200 OK to kamailio >>>>>>>>>> 8) kamailio send 200 ok to uac --> but the contact header >>>>>>>>>> contains IP of Asterisk and not kamailio: is it normal? >>>>>>>>>> >>>>>>>>>> Thank for you help. >>>>>>>>>> >>>>>>>>>> Regards >>>>>>>>>> -- >>>>>>>>>> Abdoul OSSENI >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Abdoul OSSENI >>>>>> Ingénieur Réseaux et systèmes chez THALES >>>>>> Co-Fondateur de ON SERVICES >>>>>> Tél : +33 601 135 167 >>>>>> >>>>> >>>>> >>>> >>>> >>>> -- >>>> Abdoul OSSENI >>>> Ingénieur Réseaux et systèmes chez THALES >>>> Co-Fondateur de ON SERVICES >>>> Tél : +33 601 135 167 >>>> >>> >>> >>> >>> -- >>> Abdoul OSSENI >>> Ingénieur Réseaux et systèmes chez THALES >>> Co-Fondateur de ON SERVICES >>> Tél : +33 601 135 167 >>> >> >> > > > -- > Abdoul OSSENI > Ingénieur Réseaux et systèmes chez THALES > Co-Fondateur de ON SERVICES > Tél : +33 601 135 167 >

1 0

re.subst does not work with $var?
by Pranathi Venkatayogi 15 Jun '17

15 Jun '17

I am trying to replace "domain1" with "domain2" in the msrp(body). Following stmt works where in I specify domain strings as literals. var(modBody) = $(msrp(body){re.subst,/()(domain1)()/domain2/g}); I am having trouble making the same work with $var variables. See below - this code does not do the replacement - $var(domain1) = "()(domain1)()"; $var(domain2) = "domain2"; var(modBody2) = $(msrp(body){re.subst,/$var(domain1)/$var(domain2)/g}); Any clue/ideas or is it by design? Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio+Asterisk - Kamailio doesn't forward INVITE to 2nd Kamailio
by Marko Tirs 15 Jun '17

15 Jun '17

Hi all, I have here:Kamailio 1: 192.168.0.11Kamailio 2: 192.168.0.21Asterisk 1: 192.168.0.12 Asterisk1 is registered on Kamailio1 as User 100. Asterisk1 shouldn't be registered on Kamailio2. User 111 registered on Kamailio1User 211 registered on Kamailio2User 121 registered on Asterisk1 Test cases:---------------------111 calls 211 - OK211 calls 111 - OK121 calls 111 - OK121 calls 211 - Kamailio1 doesn't forward INVITE to Kamailio2, Asterisk gets 'Busy' !? Is the approach just to replace destination IP-address in $ru, depending from 1st digit, right? If not, what is the right approach to reach the remote users from Asterisk, which are registered on remote Kamailio and not on the local Kamailio? Thank youRegardsMarko My changes in kamailio.cfg in both kamailios (based on kamailio-basic.cfg) :--------------------------------------------------------------------------------------------------------------kamailio1.bindip = "192.168.0.11" kamailio2.bindip = "192.168.0.21" ...route { if (is_method("INVITE")) { if($rU=~"^1[0-9][0-9]$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio1.bindip) + ":5060"; } else if($rU=~"^2[0-9][0-9]$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.kamailio2.bindip) + ":5060"; } } ... sip.conf in Asterisk1:---------------------------------- register => 100:abc@192.168.0.124:5060/100 [100] type=friend host=192.168.0.11 secret=abc context=kamailio [121] type=friend secret=abc host=dynamic context=kamailio extensions.conf-------------------------------[kamailio] exten => _[1-4]XX,1,Dial(SIP/100/${EXTEN},30) exten => _[1-4]XX,n,Hangup()

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sr-users June 2017