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37519 discussions

Users being registered without any Authentication
by Faisal Rehman 20 Mar '12

20 Mar '12

Hi, I have the below configuration file kamailio.cfg but all the users are not being registered with authentication, so am I doing some mistake in it? #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - 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://faisal:faisal123@localhost/kamailio" #!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_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 */ /* 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:192.162.15.23:5161 listen=tcp:129.162.15.23:5161 alias="xmpp.kamailio.org" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5161 #!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 ####### 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 "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" loadmodule "xmpp.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) #--------XMPP Module Params--------------- modparam("xmpp", "domain_separator", "%") modparam("xmpp", "xmpp_domain", "192.168.15.23") modparam("xmpp", "xmpp_host", "192.168.15.23") modparam("xmpp", "gateway_domain", "192.168.15.23") modparam("xmpp", "backend", "server") modparam("xmpp", "xmpp_port", 5299) modparam("xmpp", "xmpp_password", "casilla233") # ----- 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) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); #authentication route(AUTH); ###############-----------------------------------XMPP Server Configuration----------------------------------------################# if (uri =~ "sip:.+@sip-xmpp\.kamailio\.org") { #absorb transmission if (!t_newtran()){ sl_reply_error(); exit; } #Handling Instant Messaging if (is_method("MESSAGE")){ xlog("message to XMPP: from <$fu> ru>\n"); if (xmpp_send_message()){ sl_send_reply("202", "Accepted"); } else { sl_send_reply("404", "Not found"); } exit; } #Un-supported Type of SIP Message xlog("message to XMPP: method [$rm] not supported yet\n"); sl_send_reply("503", "Service unavailable"); exit; } ##################--------------------------------XMPP Server Configuration-----------------------------------------------############### # 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(); 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 $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 if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!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; } #!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 } Regards, Faisal

3 5

Presence question
by David 19 Mar '12

19 Mar '12

Hello, I am running a presence server. I am searching for a problem with my presence, and I have traced it down to this query : delete from active_watchers where presentity_uri='sip:test109@domain.tld' AND event='dialog' AND to_tag='f0253efb8005057b3cc0ab8fa35ff1d9-c961' This query is being sent to the database a few seconds after the initial subscribe is received. So the subscription only lasts in the database for a few seconds. I tried running level 3 error reporting ( hundreds of DEBUG ). I did not see any justification for query. Where is this query coming from ( source code/ config file )? Why is this subscription being deleted almost immediatly after it is setup ? Thanks, David Here is a sample request ( not the same as the one above, I did not capture the SIP at the same time as the SQL ) : U PROXYIP:5060 -> PRESENCEIP:5060 SUBSCRIBE sip:test109@domain.tld SIP/2.0. Record-Route: <sip:PROXYIP;lr=on;ftag=db911ce615f1598c>. Record-Route: <sip:TCPPROXYIP;r2=on;lr=on>. Record-Route: <sip:TCPPROXYIP;transport=tcp;r2=on;lr=on>. Via: SIP/2.0/UDP PROXYIP;branch=z9hG4bK9543.f0668033.0. Via: SIP/2.0/UDP TCPPROXYIP;branch=z9hG4bK9543.d2e18073.0;i=0a3. Via: SIP/2.0/TCP USERIP:61334;branch=z9hG4bK-6ee64c28;rport=61334. To: sip:test109@domain.tld. From: "103 Zone T" <sip:test103@domain.tld>;tag=db911ce615f1598c. Call-ID: 3f55db6e-b2ec6ca4(a)192.168.219.109. CSeq: 53944 SUBSCRIBE. Max-Forwards: 68. Contact: "103 Zone T" <sip:test103@USERIP:8280;transport=tcp>. Accept: application/dialog-info+xml. Expires: 1800. Event: dialog. User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. . # U PRESENCEIP:5060 -> PROXYIP:5060 SIP/2.0 202 OK. Record-Route: <sip:PROXYIP;lr=on;ftag=db911ce615f1598c>. Record-Route: <sip:TCPPROXYIP;r2=on;lr=on>. Record-Route: <sip:TCPPROXYIP;transport=tcp;r2=on;lr=on>. Via: SIP/2.0/UDP PROXYIP;branch=z9hG4bK9543.f0668033.0. Via: SIP/2.0/UDP TCPPROXYIP;branch=z9hG4bK9543.d2e18073.0;i=0a3. Via: SIP/2.0/TCP USERIP:61334;branch=z9hG4bK-6ee64c28;rport=61334. To: sip:test109@domain.tld;tag=f0253efb8005057b3cc0ab8fa35ff1d9-8c10. From: "103 Zone T" <sip:test103@domain.tld>;tag=db911ce615f1598c. Call-ID: 3f55db6e-b2ec6ca4(a)192.168.219.109. CSeq: 53944 SUBSCRIBE. Expires: 1800. Contact: <sip:PRESENCEIP:5060>. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0.

2 1

Re: [SR-Users] XMPP module not passing subscribes coming from xmpp side
by Dan-Cristian Bogos 19 Mar '12

19 Mar '12

Hi Pavel, Many thanks for your reaction. Good to know about the issues, saved me some important time with troubleshooting. Will try to find alternatives in this case since I already found some other limitations which stop me in having an enterprise solution. Have a good one! DanB > > Hi Dan, > > student of mine (Martin) realized that, but it was using Jabberd2 not > opernfire XMPP server, because the Openfire did not register domain from the > XMPP component correctly and did some other strange things. > Here are parts of Martin communication > > http://comments.gmane.org/gmane.comp.voip.openser.user/31713 > > > palo73 > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Dan-Cristian Bogos >> Sent: Friday, March 02, 2012 11:22 AM >> To: sr-users(a)lists.sip-router.org >> Subject: [SR-Users] XMPP module not passing subscribes coming from xmpp >> side >> >> Hey Guys, >> >> I was wondering if anyone having experience with using xmpp module as >> presence gateway between SIP and XMPP. I started evaluating it and till >> now got some weird results, not sure if it is just my setup which does >> not match the one widely used or the module is incomplete. >> >> Regarding my setup, I use OpenFIRE 3.7.1 on one side and Kamailio 3.2.0 >> out of debian packages on the other. >> >> The issue I got right now is that the subscribe coming from XMPP side >> produces no action/reply on SIP side. Bellow you can see the packet >> coming from OpenFIRE: >> # >> T 2012/03/02 09:49:57.743750 127.0.0.1:5275 -> 127.0.0.1:49965 [AP] >> <presence id="4h8F6-19" to="dan4(a)gw.mydomain.com" type="subscribe" >> from="dan(a)mydomain.com"><c xmlns="http://jabber.org/protocol/caps" >> ext="voice-v1 video-v1 camera-v1 " hash="sha-1" >> node="http://jitsi.org" >> ver="ohjz8WKq0ZQRiNWRNcapX4BBHLk="/></presence> >> """ >> >> Anyone here got this working somehow? >> >> Ta, >> DanB >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi- >> bin/mailman/listinfo/sr-users > > > > > ------------------------------ > > Message: 5 > Date: Fri, 2 Mar 2012 11:44:12 +0100 > From: Reda Aouad <reda.aouad(a)gmail.com> > Subject: [SR-Users] Registration Limits > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAA30pc7pkPcBSfMSdRpvCSHNm0kiSw9r13huUe4jikiEPdcr1Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Is there a way to ensure single-registration per user-agent for a user, > which overwrites previous registration ? > Or is there a way to limit the number of registrations per user, but > overwriting the earliest registration for each new one ? > > Thanks, > Reda >

3 4

Unsuccessful call using DRouting - Request Timeout
by Fatima Chahrour~Vanrise Support 19 Mar '12

19 Mar '12

Dears, I started my lab on VMware (debian lenny) where I installed Kamailio 3.2 and started drouting configuration. I'm using 2 PCs each having an x-lite softphone and the Kamailio on the vmware all connected locally. My primary test is to make a call from x-lite on PC (192.168.111.11) to x-lite on PC (192.168.111.226) through Kamailio\ drouting modules (192.168.111.139). What is happening is that when I call 9611558800, for example, am getting 'Calling' message on PC x-lite followed by 'Call Failed: Request Timeout' after a certain time. Attached is my routing logic script, and ngrep for the call that is failing. And my dr_tables data is : dr_rules: 4, '0', '9611', '', 0, '', '#3', 'Lebanon Fix' dr_gw_lists: 3, '5', 'Test Call' dr_groups: 3, '.*', '.*', 0, 'Everyone' dr_gateways: 5, 1, '192.168.111.226', 0, '', '', 2, 'PC1' I tried adding port number to the gateways, remove '#' and add 5 in the dr_rules, remove the "0" from do_routing in the script, but with no success. Reading some related documents and books, I found that all scenarios using drouting have gateways as media servers (Asterisks) . So my question: is this a must, to have a media proxy even using my above primary enviroment? If not, can you please help me finding my mistake that prohibit the call's success(ringing)? Thanks in advance. F.Chahrour

1 0

message offline store questions...
by Krishna Kurapati 19 Mar '12

19 Mar '12

Hi, I am using msilo module to store offline message. I am sending back "202 pending" when the message is stored (m_store) However, when I message delivered (m_dump), how do I inform the sender that the message is successfully delivered? Also, how do I handle if the sender if offline when I try to inform about the delivery of the message. I think I need offline notify store to handle this condition. Instead of removing the message from the silo table upon receipt of 200 OK from the receiver, I am thinking keep the message there until the NOTIFY is successfully delivered to the sender. In effect, the silo table in the database function as offline message and notify store. I reviewed module_k/msilo source code, and the functionality that I need seems missing. I am thinking of two approaches. 1) Extend the msilo module to implement the feature, if more people need it, then I can submit the changes. 2) Write sqlops + lua script to address the missing function. In either case, I am thinking of storing the callid of the original message (possibly in silo table?) in the database and sending the NOTIFY with the callid to the sender. I would like to extend the silo table with two columns: 1. the callid of the message 2. flags to indicate status of the delivery of the message. Appreciate your comments, suggestions and alternatives. Thanks Krish Kura

2 2

Re: [SR-Users] DRouting cause Kamailio 3.2 to stop
by Fatima Chahrour~Vanrise Support 19 Mar '12

19 Mar '12

Dear Daniel, Thanks for your reply.. I'd like to ask you too if my script in my first post that am using in the routing area is correct? I found file named core in the '/' location as you mentioned. When I open it using #cat core, the opened file go crazy and writes strange characters and it ending with the below: /usr/local/sbin/kamailio-P/var/run/kamailio.pid-f/usr/local/etc/kamailio//ka mailio.cfgSSH_AGENT_PID=3234GPG_AGENT_INFO=/tmp/seahorse-jKaZEU/S.gpg-agent: 3246:1TERM=xtermDESKTOP_STARTUP_ID=SHELL=/bin/bashGTK_RC_FILES=/etc/gtk/gtkr c:/home/vanrise/.gtkrc-1.2-gnome2WINDOWID=33580848USER=rootGNOME_KEYRING_SOC KET=/tmp/keyring-ZKxFck/socketSSH_AUTH_SOCK=/tmp/keyring-ZKxFck/sshUSERNAME= vanriseSESSION_MANAGER=local/debian:/tmp/.ICE-unix/3188MAIL=/var/mail/rootPA TH=/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/DESKTOP_SES SION=defaultGDM_XSERVER_LOCATION=localPWD=/usr/local/etc/kamailioLANG=en_US. UTF-8GDM_LANG=en_US.UTF-8GDMSESSION=defaultSHLVL=4HOME=/rootGNOME_DESKTOP_SE SSION_ID=DefaultLOGNAME=rootDBUS_SESSION_BUS_ADDRESS=unix:abstract=/tmp/dbus -sH8atcWdX4,guid=f0dcba443f8fd1d5267161364f4dd99dXDG_DATA_DIRS=/usr/local/sh are/:/usr/share/:/usrdebian:/# 62;9;c62;9 Sorry, I've just realized an info that I should mention and would be helpful, am running this Kamailio on a debian Lenny VMware as a trial before migrating it to its server. Thank you, F.Chahrour From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, March 01, 2012 10:15 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Fatima Chahrour~Vanrise Support; sr-users-bounces(a)lists.sip-router.org Subject: Re: [SR-Users] DRouting cause Kamailio 3.2 to stop Hello, have you found the core file in /var/log/? It is a strange location unless you started with -w /var/log. The core file is named usually 'core' or core.pid (replace pid with same number, like, 'core.1234'). The default location in in '/' unless there is a different location set via -w or enforced by operating system (e.g., mac os x puts it in '/corefiles'). Hope it helps to locate the core file and send the backtrace. Cheers, Daniel On 3/1/12 9:05 AM, Fatima Chahrour~Vanrise Support wrote: I also tried : #gdb /usr/local/sbin/kamailio /var/log/core.file (gdb) bt And got No Stack.. BR, F.Chahrour

2 9

ser_mysql.sh
by Marwan Idriss 19 Mar '12

19 Mar '12

Greeting, I need the ser_mysql.sh for the ser version ser 2.0.0-4ubuntu2 Regards Marwan

1 0

Best approach for TCP/TLS connection re-use for nated Contacts
by Giacomo Vacca 19 Mar '12

19 Mar '12

Hi all, I have a scenario where it's not clear to me what the best approach should be (Kamailio 3.2.0 on Squeeze, although I think it's a matter of routing logic only). I have a client behind NAT that sends an initial REGISTER request with a Contact containing private ip:port (e.g. Contact: "GV" <sip:12345@192.168.1.2:54321;transport=tcp> ) However, after it's challenged for authentication with a 401, it sends the following REGISTER request with Authorization header and with a Contact containing ip:port equal to the 'received' SIP URI (with a combination of 'rport' and 'received' parameters (e.g. Contact: "GV" <sip:12345@[receivedIP]:[rport];transport=tcp> ) In this case kamailio's routing logic is not marking the REGISTER as natted (no Received field in location and no NAT flags set). You can imagine I have a fairly standard NAT handling: route[NATDETECT] { force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } return; } FLT_NATS won't be set, and so it won't FLB_NATB in the following processing. This client receives a call: Kamailio relays the INVITE to it using the stored Contact. The INVITE is correctly received by the client, because the stored Contact refers to the connected socket ([received]:rport). The trouble starts when then the client replies with a 200 OK containing the private ip:port in the Contact header field (again 192.168.1.2:54321). The ACK will then have that private ip:port in the R-URI, and its relaying will fail. One of the solutions I've found is using always add_contact_alias() in onreply_route when handling the 200 OK, and then use handle_ruri_alias() when defining the destination for the ACK. I know that without traces and the full .cfg this is quite a broad question, but I wonder if you have already had to deal with this kind of client behaviour and can provide advice on the best practices. This is in particular useful for the re-use of TCP-based connections. Thanks in advance, Giacomo Truphone Limited is a limited liability company registered in England & Wales, registered office: 4 Royal Mint Court, London, EC3N 4HJ. Registered No. 04187081. VAT No. GB 851 5278 19. Tru is a brand name of Truphone and is a Truphone Communications Service. Truphone is a trading name for a number of distinct legal entities that operate in combination. www.truphone.com<http://www.truphone.com>. This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

3 3

register_udomain(): Invalid table version (use ser_mysql.sh reinstall)
by Ryan Gholam 19 Mar '12

19 Mar '12

Hello , I hope someone encountered this problem on UBUNTU , when connecting to the database i get this error in my log file :var/log/sys Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: register_udomain(): Invalid table version (use ser_mysql.sh reinstall) Mar 19 14:09:46 rayan-desktop /usr/sbin/ser[21807]: domain_fixup(): Error while registering domain Any help is appreciated . Thank you .

1 0

add_contact_alias and handle_ruri_alias vs fix_nated_contact
by yufei.tao 19 Mar '12

19 Mar '12

Hi I have been using fix_nated_register/fix_nated_contact for NAT traversal and all worked fine. I've come across add_contact_alias and handle_ruri_alias: http://sip-router.org/wiki/tutorials/alias-example where it says: "The benefits of using add_contact_alias() and handle_ruri_alias() functions instead of conventional NAT traversal solutions are: * Request-URI in requests to UAs behind NATs is always what UAs expect * Re-use of tcp/tls sessions between proxy and UAs " I assumed the 'conventional NAT traversal solution' here means fix_nated_contact? I understand the first point, in that the R-URI always contains what the client puts in the contact hf. But for the second point, about tcp/tls reuse, does fix_nated_contact do the same trick, or does add_contact_alias and handle_ruri_alias give you more benefit over fix_nated_contact? So far I feel they achieve the same thing but would like this confirmed/corrected. Thank you very much! Yufei

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