sr-users October 2014

sr-users@lists.kamailio.org

101 participants
187 discussions

Removing dispatcher entry
by dodul 10 Oct '14

10 Oct '14

Hi, I have a small question. If I remove a server from my dispatcher while it has existing calls, what's gonna happen to the next notifications or requests of the live calls. Will it send them to the same server even though the server is not listed on the dispatcher. Or it will forward them to the other servers listed in the dispatcher. Sent from my Samsung Galaxy smartphone. -------- Original message -------- From: Corey Edwards <tensai(a)zmonkey.org> Date:10-10-2014 10:28 (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: Subject: Re: [SR-Users] Problems configuring Kamailio on Centos 6.5 On Thu, Oct 9, 2014 at 9:54 PM, logikanet <webmaster(a)logikanet.com> wrote: > Good evening! > > I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to > compile and install everything, if I run : service kamailio start it > starts kamailio and if I run service kamailio status I tells me that is > running on PID... which is fine, however, if I tweak the > /usr/local/etc/kamailio/kamailio.conf according to the instructions on > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > , when I run service kamailio start It says "ok" , but when I run service > kamailio status I get " kamailio dead but subsys locked" , I copied and > pasted the settings on the web-site but they don't work, I also tried used > the original cfg file adding the "#!ifdef WITH_ASTERISK" but It still > doesn't work, I'm not an expert on Kamailio I just want to use it as a > register server behind Asterisk , perhaps if you have a valid conf file > that I could use as a template for kamailio 4.0.1 this is what I have : > Kamailio is pretty good about printing out error logs when it fails to start. The trick is to capture them. I would start by running Kamailio from the command line in no-fork mode. # kamailio -DE Corey _______________________________________________ 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

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Problems configuring Kamailio on Centos 6.5
by logikanet 10 Oct '14

10 Oct '14

Good evening! I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to compile and install everything, if I run : service kamailio start it starts kamailio and if I run service kamailio status I tells me that is running on PID... which is fine, however, if I tweak the /usr/local/etc/kamailio/kamailio.conf according to the instructions on http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb , when I run service kamailio start It says "ok" , but when I run service kamailio status I get " kamailio dead but subsys locked" , I copied and pasted the settings on the web-site but they don't work, I also tried used the original cfg file adding the "#!ifdef WITH_ASTERISK" but It still doesn't work, I'm not an expert on Kamailio I just want to use it as a register server behind Asterisk , perhaps if you have a valid conf file that I could use as a template for kamailio 4.0.1 this is what I have : #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK # # 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/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://root:axn0505@200.98.201.4/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://root:axn0505@80.241.222.146/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_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" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "80.241.222.146" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "200.98.201.4" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules" #!else mpath="/usr/lib64/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" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!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) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!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); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif -- *Giuliano Bellini* *Logikanet Webmaster* *Logikanet.com* | webmaster(a)logikanet.com | +55 (41) 3012-0232 /direct/ | +1 206-279-5207 /direct/ www.logikanet.com <http://www.logikanet.com> |

2 1

Re: [SR-Users] Crash Kamailio 4.1.4
by Igor Potjevlesch 10 Oct '14

10 Oct '14

Hello, We updated this morning Kamailio in 4.1.4 with your patch. Kamailio crashed again this afternoon. here an extract from /var/log/messages : Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2] ua=[(innovaphone IP6010/9.00 hotfix24 [9.061271/9061271/300])], INVITE from 'untrusted' host Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SDA for current SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri 0123456789 was rewritten to sip:0123456789@GW, carrier 1, domain 1 Jun 25 13:49:02 /usr/local/sbin/kamailio[20259]: : <core> [mem/q_malloc.c:140]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f12803cb450 (address 0x7f12803cb480) beginning overwritten(abcdefed)! And this is the btfull : #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x0000000000546d3c in qm_debug_frag (qm=0x7f1280275010, f=0x7f12803cb450) at mem/q_malloc.c:142 #3 0x0000000000548b26 in qm_free (qm=0x7f1280275010, p=0x7f12803cb480, file=0x6276a0 "<core>: parser/parse_ppi_pai.c", func=0x627a00 "free_pai_ppi_body", line=102) at mem/q_malloc.c:464 #4 0x000000000056e5e6 in free_pai_ppi_body (pid_b=0x7f12803cb480) at parser/parse_ppi_pai.c:102 #5 0x000000000054fee0 in clean_hdr_field (hf=0x7f1274c3c238) at parser/hf.c:126 #6 0x00007f127cb6dde6 in acc_onreply (t=0x7f1274c157f0, req=0x7f1274c3ac08, reply=0x7f12804a6d70, code=200) at acc_logic.c:501 #7 0x00007f127cb6e30a in tmcb_func (t=0x7f1274c157f0, type=512, ps=0x7fff0b015580) at acc_logic.c:573 #8 0x00007f127ed68478 in run_trans_callbacks_internal (cb_lst=0x7f1274c15860, type=512, trans=0x7f1274c157f0, params=0x7fff0b015580) at t_hooks.c:290 #9 0x00007f127ed6868a in run_trans_callbacks_with_buf (type=512, rbuf=0x7f1274c158b0, req=0x7f1274c3ac08, repl=0x7f12804a6d70, flags=200) at t_hooks.c:336 #10 0x00007f127ed9ac06 in relay_reply (t=0x7f1274c157f0, p_msg=0x7f12804a6d70, branch=0, msg_status=200, cancel_data=0x7fff0b0158e0, do_put_on_wait=1) at t_reply.c:2001 #11 0x00007f127ed9d0b7 in reply_received (p_msg=0x7f12804a6d70) at t_reply.c:2499 #12 0x000000000045d837 in do_forward_reply (msg=0x7f12804a6d70, mode=0) at forward.c:777 #13 0x000000000045e0f8 in forward_reply (msg=0x7f12804a6d70) at forward.c:860 #14 0x00000000004a58e7 in receive_msg (buf=0x924600 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK512b.82b197888826f6b60c0c63b79801294d.0;received=A.B.C.D\r\nVia: SIP/2.0/UDP A.B.C.D:2057;branch=z9hG4bK-129F259C;rport=2057\r\nCall-ID: cb0"..., len=1124, rcv_info=0x7fff0b015c60) at receive.c:273 #15 0x000000000053c9a8 in udp_rcv_loop () at udp_server.c:536 #16 0x000000000046d42b in main_loop () at main.c:1617 #17 0x0000000000470533 in main (argc=7, argv=0x7fff0b015f98) at main.c:2545 It seems to be the same problem but in a different source. Can you help me? Regards, Igor. 2014-06-12 17:46 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > we didn't set the early media parameter . its '0' by default, isn't it? > > regards, > > Igor > > > > > 2014-06-12 17:03 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> if you get a record for 180 response, then you have also the early_media >> parameter set for acc module, isn't it? >> >> In the morning I pushed a patch that should fix this issue. Use latest >> release 4.1.4 and see if works fine. Report the results to know that it is >> gone or not. >> >> Cheers, >> Daniel >> >> >> On 12/06/14 16:55, Igor Potjevlesch wrote: >> >> Hello, >> >> We don't use $ai in xlog nor in other process. only in ACC. >> >> parameters for ACC are : >> modparam("acc", "db_flag", FLT_ACC) >> modparam("acc", "db_missed_flag", 3) >> modparam("acc", "db_url", DBURLWO) >> modparam("acc", "db_extra", >> "src_user=$fU;username=$Au;src_domain=$fd;src_ip=$si;src_pai=$ai;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") >> >> For the 3781-4b1-572014182635-OGNAJ-1-A.B.C.D there is a code 180 >> ringing in the INVITE in ACC table. >> >> regards, >> >> Igor >> >> >> 2014-06-11 23:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Few more things... >>> >>> Are you recording 1xx events? Can you check to see if there is another >>> record in acc table for the same call? You can search by call-id >>> 3781-4b1-572014182635-OGNAJ-1-A.B.C.D >>> >>> Eventually, send also the parameters you set for acc module. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 11/06/14 19:25, Daniel-Constantin Mierla wrote: >>> >>> Hello, >>> >>> so you don't print $ai in xlog() statements nor use it in any >>> assignments or other functions besides acc parameter? >>> >>> Cheers, >>> Daniel >>> >>> On 11/06/14 19:19, Igor Potjevlesch wrote: >>> >>> Hello, >>> >>> We do not access to the P-asserted-identity in our configuration but >>> we added the field PAI in the db base ACC ( for INVITE, ACK and BYE) . >>> I dont know if it's in request_route, failure_route or branch_route . >>> >>> This is the print : >>> >>> (gdb) p mem_block >>> $3 = (struct qm_block *) 0x7f6a6bef1010 >>> (gdb) p shm_block >>> $4 = (struct qm_block *) 0x7f6a5666a000 >>> >>> Regards, >>> >>> Igor >>> >>> >>> 2014-06-11 18:02 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Hello, >>>> >>>> cloning to shm for tm seems ok. Can you tell where you access >>>> P-Asserted-Identity header, via variables? Does it happen in request_route, >>>> failure_route or branch_route? >>>> >>>> Can you print from gdb, any frame: >>>> >>>> p mem_block >>>> p shm_block >>>> >>>> I want to see if parsed filed point to shm or pkg memory. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 11/06/14 17:37, Daniel-Constantin Mierla wrote: >>>> >>>> Hello, >>>> >>>> at least I narrowed it down a bit. It is empty also in the clone stored >>>> in transaction, so it happens either during cloning or before. I will have >>>> to check these parts. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 11/06/14 17:00, Igor Potjevlesch wrote: >>>> >>>> Hello, >>>> >>>> This is the result, always for frame 5 : >>>> >>>> (gdb) p *t->uas.request->pai >>>> $1 = {type = HDR_PAI_T, name = { >>>> s = 0x7f6a60cd34b8 "P-Asserted-Identity: \"0987654321\" >>>> <sip:0987654321@A.B.C.D>\r\nContact: <sip:0987654321@A.B.C.D:5060>\r\nAllow: >>>> INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, >>>> REFER, UPD"..., len = 19}, body = { >>>> s = 0x7f6a60cd34cd "\"0987654321\"<sip:0987654321@A.B.C.D>\r\nContact: >>>> <sip:0987654321@A.B.C.D:5060>\r\nAllow: INVITE, BYE, REGISTER, ACK, >>>> OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE\r\nSupported: >>>> path,"..., len = 43}, len = 66, parsed = 0x7f6a6d81da88, next = >>>> 0x7f6a60cd3f10} >>>> >>>> (gdb) p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>> $2 = {id = 0x0, num_ids = 0, next = 0x0} >>>> >>>> *Did *you find one thing in common between the 2 occurrences? Do you >>>> have any ideas about what is the cause of this pai reset? >>>> >>>> Regards, >>>> >>>> Igor >>>> >>>> >>>> >>>> >>>> 2014-06-11 16:09 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>> : >>>> >>>>> Hello, >>>>> >>>>> in the same frame 5, can yo get: >>>>> >>>>> p *t->uas.request->pai >>>>> p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> On 10/06/14 18:35, Igor Potjevlesch wrote: >>>>> >>>>> Hello, >>>>> >>>>> Here is the results : >>>>> >>>>> (gdb) frame 5 >>>>> #5 0x00007f6a687e9b43 in acc_onreply (t=0x7f6a60d16ff8, >>>>> req=0x7f6a60cd2c10, reply=0x7f6a6c119aa8, code=200) at acc_logic.c:471 >>>>> 471 acc_db_request(req); >>>>> >>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 24

uri2dst: bad_uri
by Yuriy Gorlichenko 10 Oct '14

10 Oct '14

Hello. We use 2 kamailio behind load balanser that have domain name of our system . uac settings like this: modparam("uac","restore_mode","none") modparam("uac","reg_db_url", DBURL) modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 20) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_contact_addr", "sip.service.com:5068") modparam("uac","auth_realm_avp","$avp(s:realm)") modparam("uac","auth_username_avp","$avp(s:uname)") modparam("uac","auth_password_avp","$avp(s:passwd)") We have a trouble with registering trunks at porvider. When UAC try to register trunk We see fails like this: tm [ut.h:272]: uri2dst2(): ERROR: uri2dst: bad_uri: sip.service.com ERROR: tm [uac.c:266]: t_uac_prepare(): t_uac: no socket found ERROR: uac [uac_reg.c:746]: uac_reg_update(): failed to send request for [TRUNKNAME] What we try: advertised_address=sip.service.com alias=sip.service.com alias=kam2.service.com alias=Kam1.extern.ip.addr alias=Kam2.extern.ip.addr alias=load.balanser.ext.ipaddr These settings not resolved our issue. We have another trouble and think that it depended issues: We can call to external numbers through trunks, but call very slow. (10-15 seconds), and some operators (LIKE AT&T dropped our calls to voicemail systems and somethink like that) Our question is how to successfully register providers at UAC with this scheme. thanks for advices.

2 2

uac_from_replace does not restore when changing display name only
by Klaus Darilion 10 Oct '14

10 Oct '14

Hi! It seems that uac_replace_from does restore the original URI only if the >From URI is changed. Only changing the Display-name, eg: uac_replace_from("$avp(pad)",""); does not add the RR-cookie, thus the original will not be restored (e.g. on responses). Is this on purpose or a bug? regards Klaus PS: The workaround is: uac_replace_from("$avp(pad)","$fu");

1 1

Unconditional Call forwarding
by Rajesh Sharma 10 Oct '14

10 Oct '14

Hi . I have installed Kamailio 4.1.6 and basic registration and proxy server functionalities are working fine. Now I want to simulate a Unconditional call forwarding scenario with 181-Call is being forwarded is reported to originator from server. Thanks

3 3

Kamailio With RTP proxy BYE not receiving
by balu 10 Oct '14

10 Oct '14

Hi I am using kamailio with rtp proxy module. I have 2 questions /issues . 1. When caller or callee ends the call the other end call is not disocnnecting . UA is pjsip based and behind NAT router. Present call flow is pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP proxy----> ThridParty SIP Server UA local ip : 192.168.2.11 UA public IP : 89.78.92.23 Kamailio Public ip: 94.50.203.32 Third party Sip server : 76.42.89.25 Here When I disconnect call from either side , it is not disconnecting other side . 2. My second requirement is , how can I define port of third party server . for example if have 3 or 4 sip servers with different sip registration ports other tahn 5060 How can I route registration requests coming from UAs to different ports of third party servers. Please bear my ignorance I am new to kamailio .Hope some experts will help me here . Attached kamailio config and SIP trace taken from kamailio server Thank you

2 1

cr_next_domain not found
by mark＠brightvoip.co.uk 10 Oct '14

10 Oct '14

Hi, I am trying to use the carrierfailureroute table functionality via the cr_next_domain carrierroute function. I am getting the message: "failed to find command cr_next_domain" when I try to start Kamailio I am compiling Kamailio from latest 4.1.6 source, using: make include_modules"db_mysql carrierroute" cfg make group_include="db mysql standard" all make install I have installed libconfuse-devel and the tm module is loaded. Would welcome any helpful suggestions. Thanks, Mark Hall --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

2 1

dlg_set_timeout_by_profile not working on kamailio 4.1.5
by Jose Ferreira 10 Oct '14

10 Oct '14

I'm trying to use dlg_set_timeout_by_profile and it doesn't do nothing on Kamailio 4.1.5 If i try to use the dlg_set_timeout, it also doesn't work, but it gives the following error CRITICAL: dialog [dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f9825ac6880 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1094]: update_dlg_timeout(): failed to update dialog lifetime If i set the timeout with the $avp, it only works before dlg_manage(), but also with a Warning WARNING: dialog [dlg_handlers.c:1245]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f9da0dc7f08 [4066:9857] with clid 'bba73ace3ee94b95bf5b1782406047bd' and tags '6460df6b8b3947ba96b1cf65330bf524' 'm2pmrUpHN06Fc' The dialog module configurations are: modparam("dialog", "db_url", DBURL) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "db_mode", 0) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "user ; domain ; permissions") modparam("dialog", "profiles_no_value", "inbound ; outbound ; all") modparam("dialog", "send_bye", 1) modparam("dialog", "default_timeout", 21600) modparam("dialog", "ka_timer", 0) modparam("dialog", "ka_interval", 0) modparam("dialog", "timeout_avp", "$avp(dlgtimeout)") I'm very interested to use dlg_set_timeout_by_profile, any idea about what is wrong with that ? Regards, *José Ferreira | Technical Manager* M. +351 91 775 7166 | jferreira(a)wavecom.pt Wavecom, Soluções Rádio, SA Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 234 919 190 | F. +351 234 919 191 *www.wavecom.pt <http://www.wavecom.pt/>* [image: WavecomSignature]

2 1

UAC call to AT&T
by Yuriy Gorlichenko 10 Oct '14

10 Oct '14

Hello. we use UAC module for trunk registration at kamailio. We have successfull routing and calling through our trunks to many providers and have just one problem with AT&T endpoints. When callee at AT&T endpoint hook call (I see OK from provider as response to our INVITE from AT&T endpoint) and then kamailio sends CANCEL to endpoint, but then kam sends ACK to OK... So after this strange scenario I successfully hear invite to leave voice mail to AT&T endpoint... What I saw: I look at codecs- at OK i See ulaw codec, so at my INVITE ulaw presents too. With other providers calls succesfully established. Does Anyone have an experiense with calling to AT&T from Kamailio. P.S. calls through our trunk to AT&T from asterisk establishes succesfully too.

2 1

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sr-users October 2014