sr-users August 2013

sr-users@lists.kamailio.org

93 participants
176 discussions

tls.so: mod_register failed on fedora 18 x86_64

by Roberto Fichera

Hi All, I'm just trying to setup the TLS on the last Kamailio v4.0.3 on Fedora 18 64bits. Kamailio doesn't start and terminate with the error below: loading modules under /usr/local/lib/kamailio/modules_k/:/usr/lib64/kamailio/modules/ 0(1676) ERROR: tls [tls_init.c:471]: tls_pre_init(): Unable to set the memory allocation functions 0(1676) ERROR: <core> [sr_module.c:600]: load_module(): ERROR: load_module: /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(1676) : <core> [cfg.y:3567]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 279, column 12-19: failed to load module 0(1676) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found 0(1676) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 439, column 51: Can't set module parameter 0(1676) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found 0(1676) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 440, column 97: Can't set module parameter 0(1676) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found 0(1676) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 441, column 92: Can't set module parameter 0(1676) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <tls> found 0(1676) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 442, column 64: Can't set module parameter ERROR: bad config file (5 errors) Cheers, Roberto Fichera.

11 years

Unable to call Extension in Asterisk Server & Vice Versa

by Nishar M.H

Hi All, I have created a Two VMs for Asterisk & Kamailio. Asterisk IP : 192.168.20.196 Kamailio IP : 192.168.20.208 I have installed both successfully. I have created trunk between two and successfully started kamailio. What i am looking for : 1) Asterisk as media server and Kamailio as SIP Server. 2) Call from Asterisk to Kamailio and vice versa. 3) Enable Outbound call for Kamailio User through Asterisk Server. 4) Inbound call from outside through Asterisk to Kamailio Users. The issues i am facing here is : 1) Unable to call Internally. Kamailio SIP to SIP. If i tried to call from 1000 to 1001, "No route to destination. 2) Unable to call from Asterisk to Kamailio and vice versa. Here is the log below: /var/log# tail /var/log/syslog Aug 21 09:37:36 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17507]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Aug 21 09:39:59 kamailio-VirtualBox /usr/local/sbin/kamailio[17504]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1377063599;method=INVITE;from_tag=ea6d986d;to_tag=;call_id=ZDZhMzkxN2RjZjhhYmJhMmJiNTM0OTRlYzA3NzJkMTA.;code=408;reason=Request Timeout;src_user=1000;src_domain=192.168.20.208;src_ip=192.168.20.186;dst_ouser=1000;dst_user=1000;dst_domain=192.168.20.186 One more thing i have to ask is : 1) Is there any command to check Kamailio has a trunk with Asterisk PBX server ? Like in Asterisk SIP show peers. I have tried "kamctl ul show". But it shows kamailio user 1000 and 1001. My kamailio.cfg file : #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - 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. # # 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 ####### 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 fork=yes 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:10.0.0.10:5060 /* 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" 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 (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.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 "mi_rpc.so" loadmodule "acc.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 --------------- # ----- 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) # ----- 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) 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) #!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) 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 setflag(1); # do accouting if (uri=~"sip:4000@192.168.20.196:5080") { route(2); } }* # 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[2] { rewritehostport("192.168.20.196:5080"); # change the IP here with the IP of your Asterisk Server t_relay(); exit; }* 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 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()) { 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); } 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 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } 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 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_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 route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } 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)) { add_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) { 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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($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 # 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; } #!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; } #!endif } -- Thanks & Regards, -------------------------------------------------------------------------------------------- *Nishar Hamsa * --------------------------------------------------------------------------------------------

11 years

Subscribe to a common resource list

by Yufei Tao

Hi I'm trying to work with RLS. Is there a way of allowing user1 and user2 to both subscribe to a common group of contacts, group.common? Seems that each user can only subscribe to its own resource list and won't be able to subscribe to other user's resource list. For example, both user1 and user2's RLS services documents contain resource-list pointing to group1's resource list, like this: <?xml version='1.0' encoding='UTF-8'?> <rls-services xmlns:rl="urn:ietf:params:xml:ns:resource-lists" xmlns="urn:ietf:params:xml:ns:rls-services"> <service uri='sip:group1@mydomain.com'> <resource-list>http://mydomain.com:5060/xcap-root/resource-lists/users/group1/generallist.… </resource-list> <packages> <package>presence</package> </packages> </service> ... ... </rls-services> The problem is the watcher_user in this case would be user1 or user2, which is not 'group1' so the resource list document fails to be retrieved. How can I allow user1 and user2 to retrieve group1's RL document? Thank you! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

11 years

options_reply()

by Alex Balashov

Hello, The options_reply() function does not answer OPTIONS pings that contain a username in the request URI. To its credit, the documentation does say that, too: http://kamailio.org/docs/modules/4.0.x/modules/siputils.html#idp131056 The question is: why not? I do not see anything in RFC 3261 Section 11.1 ("Construction of an OPTIONS Request") that seems to rule out an OPTIONS request with a user part: http://tools.ietf.org/html/rfc3261#section-11.1 Quite a few UAs out there, including, notably, Metaswitch, do send user parts in the OPTIONS RURI. To deal with them, I am forced to simply sl_send_reply("200", "OK"); instead of using options_reply(). What is the underlying theory? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

11 years

CARRIERROUTE + LCR

by Lucky Santiago

I've installed Kamailio v4.0.x packages on my Ubuntu 12.04.1 LTS I would like to configure LCR and CARRIERROUTE modules but I don't find sufficient guide to do it properly. Though i am looking to http://kamailio.org/docs/modules/4.0.x/modules/ for the said modules but it's a bit scattered for me. Can anyone give me resources or guide where I can found : - required modparam for LCR and CARRIERROUTE - SQL examples datas for both modules - example of route logic using both modules Thanks in Advance

11 years

Presence: Get number of active_watchers

by Jan Gaida

Hello, I need to obtain the number of subscribed watchers for a given presentity-uri. I can get it with a SQL Query like "select count(*) from kamailio.active_watchers where presentity_uri = 'test@domain';". But I think there should be a way to get it from within the configuration script. Regards -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

11 years

Enterprise Presence Information

by Ryan C Ash

Good day, Quick question for the group. I am looking at choosing an open source product to potentially use for enterprise presence. The solution would be on the receiving end of SIP NOTIFY messages from Cisco CUCM clusters. This is a very large enterprise environment. I am hoping to stay on the LAMP stack which leads me a bit away from Mobicents. At this time we aren't looking for much else other than designing an highly available solution that can act as the receiving end of these SIP NOTIFY messages and provide that information to consumers as necessary. At this time I don't know exactly what or how the consumers will utilize the data. The enterprise already uses Microsoft Lync as the defacto presence system but it isn't fully integrated with our unified communications deployment. Is Kamilio my best solution? I have also looked at openSIPS but it looks like Kamilio is a more active project with more functionality (much of which I don't immediately need). Should I reconsider Mobicents?

11 years

Is Kamailio's presence modules support pidf-diff+xml format?

by Evgeny Vinnik

Is Kamailio's presence module support pidf-diff+xml (RFC 5262 aka Partial PIDF) format? Did anyone tried to implement parsing of such format? It just I have an issue with an RCS client, which updates presence information in the Partial PIDF format.

11 years

multihomed Kamailio and enable_double_rr

by Steve Davies

Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a phone on 172.16.230.1, talking to Kamailio on 172.16.230.128. On the "outside" Kamailio uses 10.64.5.16 and its talking to 41.221.230.60 I'm using the stock Kamailio 4.0.3 kamailio.cfg, with: WITH_NAT defined mhomed=1 Little change in NATMANAGE to do the rtpproxy_manage with ie or ei as appropriate, coming from my previous post and the response from Alex. Here's the invite from the phone: U 172.16.230.1:3694 -> 172.16.230.128:5060 INVITE sip:7171001@vc2.connection-telecom.com;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.16.230.1:3694 ;branch=z9hG4bK-d8754z-6a91626ae4c3f625-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:2686959@172.16.230.1:3694;transport=udp>. To: <sip:7171001@vc2.connection-telecom.com>. From: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: ...some stuff... Supported: replaces. User-Agent: Bria 3 release 3.5.3 stamp 70600. Content-Length: 256. . v=0. o=- 1377005946728952 1 IN IP4 172.16.230.1. s=Bria 3 release 3.5.3 stamp 70600. c=IN IP4 172.16.230.1. t=0 0. m=audio 52448 RTP/AVP 8 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. Kamailio forwards with double-Record-Route with both of its addresses. I believe this is per SIP OUTBOUND RFC: U 10.64.5.16:5060 -> 41.221.230.60:5060 INVITE sip:7171001@vc2.connection-telecom.com;transport=udp SIP/2.0. Record-Route: <sip:10.64.5.16;r2=on;lr=on;ftag=014e3010;nat=yes>. Record-Route: <sip:172.16.230.128;r2=on;lr=on;ftag=014e3010;nat=yes>. Via: SIP/2.0/UDP 10.64.5.16;branch=z9hG4bKe355.e526ca52.0. Via: SIP/2.0/UDP 172.16.230.1:3694 ;branch=z9hG4bK-d8754z-6a91626ae4c3f625-1---d8754z-;rport=3694. Max-Forwards: 16. Contact: <sip:2686959@172.16.230.1:3694;transport=udp>. To: <sip:7171001@vc2.connection-telecom.com>. From: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: ...some stuff... Supported: replaces. User-Agent: Bria 3 release 3.5.3 stamp 70600. Content-Length: 270. P-hint: outbound. . v=0. o=- 1377005946728952 1 IN IP4 10.64.5.16. s=Bria 3 release 3.5.3 stamp 70600. c=IN IP4 10.64.5.16. t=0 0. m=audio 59194 RTP/AVP 8 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. So that behaviour seems OK. The call does get correctly established and rtpproxy is correctly setup and audio passes in both directions. But when the BYE is sent (from the outside), though, things go wrong: Here's what arrives from upstream. Route: has the two entries per the RR that was sent. U 41.221.230.60:5060 -> 10.64.5.16:5060 BYE sip:2686959@10.64.5.16:5060;transport=udp SIP/2.0. Record-Route: <sip:41.221.230.60;lr=on;ftag=as70703d1c>. Via: SIP/2.0/UDP 41.221.230.60;branch=z9hG4bKbd37.4108b6b2.0. Via: SIP/2.0/UDP 41.221.230.60:5070 ;received=41.221.230.60;branch=z9hG4bK4e6b38bf;rport=5070. Route: <sip:10.64.5.16;r2=on;lr=on;ftag=014e3010;nat=yes>,<sip:172.16.230.128;r2=on;lr=on;ftag=014e3010;nat=yes>. Max-Forwards: 69. From: <sip:7171001@vc2.connection-telecom.com>;tag=as70703d1c. To: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 102 BYE. User-Agent: Enswitch. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. X-Enswitch-RURI: sip:2686959@10.64.5.16:5060;transport=udp. X-Enswitch-Source: 41.221.230.60:5070. . So Kamailio peels off the first route and then sends the BYE actually to itself. With an oddly formed blank Route: header. Tracing through the kamailio.cfg the BYE is processed in WITHINDLG - loose_route() succeeds It logs that 172.16.230.128 "is loose router". U 10.64.5.16:5060 -> 172.16.230.128:5060 BYE sip:172.16.230.128;r2=on;lr=on;ftag=014e3010;nat=yes SIP/2.0. Record-Route: <sip:41.221.230.60;lr=on;ftag=as70703d1c>. Via: SIP/2.0/UDP 10.64.5.16;branch=z9hG4bKbd37.25d16bf3.0. Via: SIP/2.0/UDP 41.221.230.60;rport=5060;branch=z9hG4bKbd37.4108b6b2.0. Via: SIP/2.0/UDP 41.221.230.60:5070 ;received=41.221.230.60;branch=z9hG4bK4e6b38bf;rport=5070. Route: . Max-Forwards: 16. From: <sip:7171001@vc2.connection-telecom.com>;tag=as70703d1c. To: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 102 BYE. User-Agent: Enswitch. X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Content-Length: 0. X-Enswitch-RURI: sip:2686959@10.64.5.16:5060;transport=udp. X-Enswitch-Source: 41.221.230.60:5070. . When Kamailio receives the BYE from itself it sends a 404 Not here. Which is forwarded back upstream. This 404 Not here is generated in WITHINDLG too; looks like loose_route() fails (which makes sense since there is nothing in the Route header), and in that case WINTHINDLG only has code for dealing with SUBSCRIBE and ACK. U 172.16.230.128:5060 -> 10.64.5.16:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.64.5.16;branch=z9hG4bKbd37.25d16bf3.0;rport=5060. Via: SIP/2.0/UDP 41.221.230.60;rport=5060;branch=z9hG4bKbd37.4108b6b2.0. Via: SIP/2.0/UDP 41.221.230.60:5070 ;received=41.221.230.60;branch=z9hG4bK4e6b38bf;rport=5070. From: <sip:7171001@vc2.connection-telecom.com>;tag=as70703d1c. To: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 102 BYE. Server: kamailio (4.0.3 (i386/linux)). Content-Length: 0. . U 10.64.5.16:5060 -> 41.221.230.60:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 41.221.230.60;rport=5060;branch=z9hG4bKbd37.4108b6b2.0. Via: SIP/2.0/UDP 41.221.230.60:5070 ;received=41.221.230.60;branch=z9hG4bK4e6b38bf;rport=5070. From: <sip:7171001@vc2.connection-telecom.com>;tag=as70703d1c. To: "vc2 2686959"<sip:2686959@vc2.connection-telecom.com>;tag=014e3010. Call-ID: ZDQ4YThjNzEzOTBhOTE5NGViNTFhM2Q5MTY2ZmY1ZDc. CSeq: 102 BYE. Server: kamailio (4.0.3 (i386/linux)). Content-Length: 0. . I tried with enable_double_rr as 0 and that did send only one Record-Route with the relayed INVITE, but the record route uses the inside address of the proxy and so we never even receive the BYE from the upstream system in that case. I'm kinda lost about where this is going wrong - so pointers would be welcome! Thanks, Steve

11 years

Problem in call initiate from PC to Mobile phone using sip client thjrough kamailio and rtpproxy

by Manas Mohanty

Hi All, I have installed kamailio-4.0.2 and rtpproxy-1.2.1 in same machine on a private ip (10.150.226.160). First I used PC to PC call by using MicroSip as sip client at both end in LAN. Everything running fine.I can make calls and receive calls on both PC. Then I used SAMSUNG Galaxy star duo and installd CSipSimple as sip client on wifi. When I try call from SAMSUNG Galaxy star (IP - 10.150.212.245:5060) to PC (IP - 10.150.216.84:5060) Its running fine. I can receive call . BUT When I call from PC to SAMSUNG Galaxy star call is not connecting . kamailio sending INVITE message to SAMSUNG Galaxy star repeatedly. No 200 OK comes from SAMSUNG Galaxy star. Can Anybody suggest what could be the problem ? Thanks in advance. Thanks, Manas

11 years

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sr-users August 2013