- sr-users - kamailio.org

multiple endpoints call
by Yuriy Gorlichenko 12 Oct '14

12 Oct '14

Hello. I have multiple endpoints registered at my kamailio with one account (for example user1 registerd from norhway, USA, and Russia at one time), so when I call from user2 to user1 I want to ring all endpoints registered by user1 account. Now I can ring only one, first entry at location table. What setting may I use for this? Thanks

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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

3 2

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

Serial forking & q value
by Marino Mileti 10 Oct '14

10 Oct '14

Hi guys, I've tried with success to implement serial forking based on q value (i've followed documentation) with the istruction: append_branch("sip:b@example.com", "0.5"); Now my questions are: is it possible to import URI from db? is it possible to import q value from db too? I would like to change the q value based, for example, on "night & day"; so it would be nice if I can store it on DB and then load it dinamically when i need.For the first question I think that it's possible with dbaliases for example...but I don't know how get a q value for each contact :) Marino

2 1

Best way to handle 100 but no 180/183?
by Tobias 09 Oct '14

09 Oct '14

Dear Kamailio users, I have an issue with some upstream carriers quickly replying with 100 Trying but then basically times out. I'm looking for a good solution to distinguish between the 100 reply and any other replies, and alter the timeout values. My goal is to be able to re-route calls where the 180/183 is not received within a few seconds, and if the 180/183 is received I'd like to give the call a much longer timeout. I realize there is a "fr_inv_timer" that would do this, but I would not like to set that to low since I fear it would affect many other call types, i.e. if a carrier would send 183 once but for some reason does not resend on regular intervals. I'm on Kamailio 4.1 and my current settings are:fr_timer : 2000fr_inv_timer : 120000restart_fr_on_each_reply : 1 The only solution I can think of is to update the fr_inv_timer in an on_reply-route if the reply code is 180/183 but I'm not sure that would even work? Is that the way to go or is there another way? Thanks in advance!/Tobias

3 4

Setting Kamailio User-Agent dynamically
by Errol Samuels 09 Oct '14

09 Oct '14

Hello All, I am using Kamailio in front of Freeswitch and Kamailio is handling the registrations then forwarding to Freeswitch and everything is working for the most part. However, there is niggling pain in the neck issue where the user's User-Agent shows in Freeswitch as kamailio (4.0.6 (x86_64/linux)) instead of the user's Real User-Agent I am trying to get Kamailio to pass the real User-Agent of the user without success. I am able to set a static User-Agent by doing this in the global section. *user_agent_header = "User-Agent: We love beer";* However, I am unable to set it dynamically for example: *user_agent_header = "User-Agent: " + $ua;* With is one Kamailio doesn't even start so my syntax is probably wrong. I also tried playing around with the textops module so I have ran out of ideas Any ideas? Thanks in advance. regards, Errol

4 17

KAMAILIO not updating Registration status to DB
by Varghese Paul 09 Oct '14

09 Oct '14

Hi all, We are experiencing a strange problem in our Kamilio server. Some of the webrtc endpoint registrations are not updated to the Database and there is no issue with the registration. Once we flushed the registration for this endpoint then it is updating to the DB for the new Register Packet. Flushed the Registration for the endpont : kamctl ul rm <USERNAME> Kamalio Version: 4.1.2 modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 1)* modparam("usrloc", "nat_bflag", 3) modparam("usrloc", "use_domain", 1) Is there any other settings to avoid this issue ? Regards Varghese Paul

2 2

duplicate information in siptrace table
by Fred 09 Oct '14

09 Oct '14

Hello, I use Kamailio 4.1.3 with siptrace module. In the sip_trace table I see duplicate information, one line with empty and one line with filled traced_user column. The empty lines are useless for me, I would like only entries with traced_user information. Here the siptrace part of kamailio.cfg: #### Parameters for Siptrace module modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_sl_acks", 0) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_delayed", 0) ###### SIP Trace Config in main route if (!is_method("OPTIONS")) { if (!ds_is_from_list("2")) { $avp(s:user) = $fU; } else { $avp(s:user) = $rU; }; if (is_avp_set("$avp(s:user)")){ sip_trace(); #setflag(22); }; }; Has anyone a tip for me how I can get rid of the empty traced_user lines? Regards Fred

2 1

Re: [SR-Users] Segfault in 4.1.6
by David Wilson 09 Oct '14

09 Oct '14

Hi Daniel, Thanks for the response. Reviewing the logs again, the first occurrence was on 25Sep, after upgrading to 4.1.6+precise on 24Sep. Previous upgrades were to: 4.1.5+precise 18Aug 4.1.4+precise June Traffic patterns are fairly uniform over that period. I've installed the nightly (4.1.6+0~20141004025841.74+precise) and we'll see how that goes until the next release. Cheers, Dave > Date: Thu, 02 Oct 2014 09:02:14 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Segfault in 4.1.6 > Message-ID: <542CF876.6060900(a)gmail.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello, > > I see it crashed in dialog, it was a fix for it after 4.1.6 for the > cleanup timer procedure, affecting some particular cases with dialog > states (e.g., dialogs not answered even after many minutes since > creation). It was there before 4.1.6 (most probably from 4.1.0), not > introduced by that version. What you were using before 4.1.6? > > You can get latest version from branch 4.1 either via GIT or via debian > nightly builds of 4.1 (if using debian). > > Otherwise, yes, a coredump with a gdb backtrace will be very helpful. > > Cheers, > Daniel > > On 02/10/14 08:31, David Wilson wrote: > > Hi all, > > > > I?ve had Kamailio running nicely for nearly a year, but have hit a segfault twice since upgrading to 4.1.6 earlier this week. > > > > The log shows: > > > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1222]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 > > Oct 2 11:28:03 localhost kernel: [11259.904645] kamailio[1206]: segfault at 58 ip 00007fa56fba0f86 sp 00007fffea594ca0 error 4 in dialog.so[7fa56fb63000+54000] > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:777]: handle_sigs(): child process 1206 exited by a signal 11 > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:780]: handle_sigs(): core was not generated > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: INFO: <core> [main.c:792]: handle_sigs(): INFO: terminating due to SIGCHLD > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1220]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received > > ?. > > > > The first two lines above seem to be the important parts. Nothing was logged for about 10s prior to this extract. An earlier instance of the issue showed almost the same details: > > > > Oct 1 20:39:12 localhost /usr/sbin/kamailio[1219]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 > > Oct 1 20:39:12 localhost kernel: [554502.639227] kamailio[1207]: segfault at 58 ip 00007fd62c5c6f86 sp 00007fffd77c9890 error 4 in dialog.so[7fd62c589000+54000] > > > > I?m trying to catch a core next time it occurs, no core is available yet. > > > > Is this a new issue in 4.1.6? > > Any suggestions about the likely cause? > > > > Cheers, > > Dave. > >

2 1

GIT branch 4.2 has been created
by Daniel-Constantin Mierla 08 Oct '14

08 Oct '14

Hello, short note to say that the branch 4.2 has been created, as announced on mailing list a while ago. This is the official branch to be used for releasing any of the 4.2.x versions. The first one, v4.2.0, is expected to be out in about one week. The version used now in master branch is prefixed with 4.3.0, which is expected to be the next future major release. With this change, practically commits with new features to be part of 4.3.0 can be pushed to master branch from now on. Any fix that is applicable for both master and 4.2 branches has to be committed first in master branch, then cherry-picked or merged to branch 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Multiple kamailio sharing single Registration DB
by Varghese Paul 08 Oct '14

08 Oct '14

Hi all, How i can setup multiple kamailio servers should share the single registration DB. Requirement: I have two kamailio instances Kamailio-A and Kamalio-B ,both are sharing the same DB which is postgre*SQL *which is hosted in some other server. When a USER AGENT A send a register packet to Kamailio-A it is updating the registration status in the DB. But in kamailio-B if am doing a lookup(location) i am getting -1 as return value. kamctl ul show <USER_AGENT> 404 AOR not found 1. How kamailio retrieving the registrations status ? Is it from the DB or from the internal cache ? 2. How can i read the registration status from the DB always ? Let me know if any one need any more information. Regards Varghese Paul

2 1

chane source IP for REGISTER to provider
by Yuriy Gorlichenko 08 Oct '14

08 Oct '14

Hello. We have multiple kamailio behind load balancer. We use UAC to send REGISTER procedure to provider. Some providers drop wrong ip. So We have loadbalanser with external IP and have kamailio servers with external IPs too. So we need to send REGISTER packets to porovide with source IP of our Load balancer. Now kamailio send REGISTER pakets with its external IP. Question is how to send REGISTER with ip adress of loadbalanser? (allias with it dn and ip already exists at config) (listet already had it ip too) thanks/

2 1

Kamailio -topoh- and Via headers
by Gonzalo Gasca 08 Oct '14

08 Oct '14

I'm using Kamailio as SIP Registrar using Websockets. My topology looks like this: Sip client (sipml5) ---> wss ---> Nginx ---> ws ---> Kamailio 4.1.5 When I look into my SipMl5 application in the Register Message 200 OK from Kamailio I see the Nginx private IP address 172.31.22.2 Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=37111;received=172.31.22.2;branch=z9hG4bKtv75otkzmPVsdNWevweLt4TN9JnLnQ0p How can I remove private IP Address in Via header to achieve topology hiding? >From Kamailio logs: Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sips:gogasca@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200#015#012 Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(172.31.22.2, df7jal23ls0d.invalid, 0) O Version: kamailio 4.1.5 (x86_64/linux) # ------ topoh -------- modparam("topoh", "mask_key", "opencall") modparam("topoh", "mask_ip", "<Public IP Address of Kamailio>") modparam("topoh", "vparam_prefix", "llamato") modparam("topoh", "mask_callid", 1) modparam("topoh", "sanity_checks", 1)

2 6

Re: [SR-Users] Presence PUBLISH error
by Marino Mileti 08 Oct '14

08 Oct '14

Thanks Daniel, I'm trying to use sqlite (mysql is too "heavy" for my environment) but usually I receive a lot of commit failed because database is locked (especially in presentity.c and publish.c) Is there any way to manage, maybe i've missed some sqlite configuration parameters? ----Messaggio originale---- Da: miconda(a)gmail.com Data: 6-ott-2014 18.17 A: "Marino Mileti"<marino.mileti(a)alice.it>, "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] Presence PUBLISH error hello, It seems that the column type and value type are mismatching, but they can still be compatible. The fact is that db_text was not designed to handle large data values, so not tested with presence. Eventually you may want use sqlite (or other db module) instead or go into db_text and update as needed. Cheers, Daniel On 06/10/14 15:14, Marino Mileti wrote: Hi guys, with Kamailio 4.1.6 standard configuration and dbtext as db module, when server receives a publish message on the log there's: ERROR: db_text [dbt_base.c:391]: dbt_insert(): incompatible types v[5] - c[7]! ERROR: presence [presentity.c:402]: update_presentity(): inserting new record in database ERROR: presence [publish.c:464]: handle_publish(): when updating presentity and the server reply with 500 Server Internal Error. My conf is standard I've only used DBURL to point to my dbtext files: I've enable presence and presence_xml module # ----- presence params ----- modparam("presence", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence", "db_table_lock_type", 0) modparam("presence_xml", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence_xml", "force_active", 1) Any idea? Marino _______________________________________________ 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

rtpengine with rejected re-invites to new RTP ports
by Jeff Pyle 08 Oct '14

08 Oct '14

Hello, Given the following scenario with Kamailio and rtpengine in the middle: - call establishes with G.711 RTP - b-leg re-invites to T.38, indicating a different port number then he is using for G.711 - a-leg refuses the re-invite with a 488 - call continues using G.711 on original port numbers as if re-invite never happened Will Kamailio and rtpengine handle it? Today I have a functional configuration with rtpproxy on another SIP proxy. It handles every scenario I have tried except this one. rtpproxy sees the new ports in the re-invite and adjusts its session accordingly. If the re-invite is rejected, the old media ports are no longer valid through rtpproxy, and the call fails. Is there an approach I can take with Kamailio and rtpengine to allow this scenario to succeed? Regards, Jeff

2 2

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

Thx, exactly what I was looking for!

1 0

tcp_read_data() errors in syslog
by Petr.Wozniak＠seznam.cz 07 Oct '14

07 Oct '14

Hello All, I have installed kamailio with TLS module together with siremis on server and all seems works fine except that sometimes appear in syslog file the following errors: /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection timed out (110) /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading /usr/sbin/kamailio[3268]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection reset by peer (104) Some part of syslog please you find enclosed. I don't know where I should search for the cause of these errors and how to debug and remove them. I don't know if these errors are caused by incorrect kamailio configuration or are caused on client's side. Thank you for your opinions and help in advance Regards, Petr

3 2

Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

I'm using Kamailio 4.15 with Siremis 4.1.0 both are great BTW! I'd like to add management of the Kamailio Dynamic Routing Module to Siremis web manager Any pointers on how to do this; I'm not familiar with OpenBiz Thx in advance

2 1

multiple endpoints call
by Marino Mileti 07 Oct '14

07 Oct '14

Take a look at : modparam("registrar", "max_contacts", x) to have up to "x" contacts for AOR and in your REGISTRAR route if your save("location") has some extra parameters (for example the 0x04 used to restrict one single record in loaction table ----Messaggio originale---- Da: ovoshlook(a)gmail.com Data: 6-ott-2014 22.02 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: [SR-Users] multiple endpoints call Hello. I have multiple endpoints registered at my kamailio with one account (for example user1 registerd from norhway, USA, and Russia at one time), so when I call from user2 to user1 I want to ring all endpoints registered by user1 account. Now I can ring only one, first entry at location table. What setting may I use for this? Thanks

1 0

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

ok thanks

1 0

tcp_read_data() errors in syslog
by Petr.Wozniak＠seznam.cz 06 Oct '14

06 Oct '14

Hello All, I have installed kamailio with TLS module together with siremis on server and all seems works fine except that sometimes appear in syslog file the following errors: /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection timed out (110) /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading /usr/sbin/kamailio[3268]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection reset by peer (104) Some part of syslog please you find enclosed. I don't know where I should search for the cause of these errors and how to debug and remove them. I don't know if these errors are caused by incorrect kamailio configuration or are caused on client's side. Thank you for your opinions and help in advance Regards, Petr

1 0

Presence PUBLISH error
by Marino Mileti 06 Oct '14

06 Oct '14

Hi guys, with Kamailio 4.1.6 standard configuration and dbtext as db module, when server receives a publish message on the log there's: ERROR: db_text [dbt_base.c:391]: dbt_insert(): incompatible types v[5] - c[7]! ERROR: presence [presentity.c:402]: update_presentity(): inserting new record in database ERROR: presence [publish.c:464]: handle_publish(): when updating presentity and the server reply with 500 Server Internal Error. My conf is standard I've only used DBURL to point to my dbtext files: I've enable presence and presence_xml module # ----- presence params ----- modparam("presence", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence", "db_table_lock_type", 0) modparam("presence_xml", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence_xml", "force_active", 1) Any idea? Marino

2 1

Problem when I want install kamailio on CENTOS 7 x64
by ANTHONY HESNAUX 06 Oct '14

06 Oct '14

Hello all, I want to install kamailio on CENTOS 7 x64 but I have this problem when I execute "kamdbctl create » Can you help me ?? Thank YOU Anthony Console display: [root@localhost kamailio-4.0.1]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. INFO: creating database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: granting privileges to database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: creating standard tables into kamailio ... Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. ERROR 1101 (42000) at line 2: BLOB/TEXT column 'body' can't have a default value ERROR: Creating core tables failed at msilo! [root@localhost kamailio-4.0.1]#

2 1

BYE issue - Kamailio +RTP proxy
by balu 06 Oct '14

06 Oct '14

Dear experts , 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 INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 60822 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:60823 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 12140 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.testcalls.com", nonce="141110871219020226209383237537" Content-Length: 0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> ACK sip:18792356789@76.42.89.25:5060 SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 ACK Content-Length: 0 INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.4f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Proxy-Authorization: Digest username="test", realm="sip.testcalls.com", nonce="141110871219020226209383237537", uri=" sip:18792356789@76.42.89.25:5060", response="e98f243028e20a3d864dc54149db8ab1" Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 47574 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:47575 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv ACK sip:76.42.89.25:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.3f67e202ad7510de3a935364e562755c.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjpDeX-U.AQN7jSbWJ7Tqg4Sbl0Lby6nkE Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 ACK Content-Length: 0 BYE sip:test@192.168.2.11:58429;ob;alias=89.78.92.23%7E58429%7E1 SIP/2.0 Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060;branch=z9hG4bK190938140255193499457203 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Content-Length: 0 Max-Forwards: 70 SIP/2.0 200 OK CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060 ;branch=z9hG4bK190938140255193499457203;rport=5060 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 ----------- config file below ----------- #!KAMAILIO # check_via=no rev_dns=no dns=no #!define WITH_NAT #!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 ######### #!ifdef WITH_MYSQL #!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 #!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 #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=2 memlog=2 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes #sreekanth commented no auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" listen=udp:94.50.203.32:8304 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=3074 #!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/" #!else mpath="/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" loadmodule "mangler.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", 1) #sreekanth changed above value from 0 to 1 # ----- 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(i:42)") #sreekanth changed above $avp(RECEIVED) to $avp(i:42) 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 #added by sreekanth #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) loadmodule "rtimer.so"; #!ifdef CNXCC_CHANNEL modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") #!endif loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second #sreekanth-End ####### 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 { #setflag(DLG_FLAG); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if (is_method("INVITE")) { route(RELAY); } # 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")) { sl_send_reply("100", "Trying"); 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==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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); #t_newtran(); #t_reply("200", "OK"); # exit; } 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; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # 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 { if(is_first_hop()) set_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("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} #from here sreekanth commented #if (is_request()) { #rtpproxy_manage("co"); } #if (is_reply()) { #rtpproxy_manage("z50"); } #if (is_request()) { # if (!has_totag()) { # if(t_is_branch_route()) { # if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { # add_rr_param(";nat=yes"); # } # } # } # } #sreekanth till here if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { 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|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

1 0

Simplest config possible [cont]
by White, Phil 06 Oct '14

06 Oct '14

Hi All, I've had an 'interesting' weekend reading RFCs and playing with kamailio. I'm making progress - however, I would be grateful for a little clarification. I have sorted out a very basic stateless redirect proxy (config file below). I have cut it down to the absolute basic minimum - and it mostly appears to do what I would expect it to do. It does allow me to call any registered extension from my SIP softphone. Problem 1: I have a Grandstream Budgetone 200, which will happily receive calls, but will not initiate any. Looking at a ngrep, it is ignoring the redirect, and sends out a flood of UDP INVITEs (also see below for details) Question: Is this the fault of my configuration, or the Grandstream behaving incorrectly? Problem 2: I am having difficulty in working out which module and command I require to perform a function. Could someone please indicate the INVITE route in this file for a stateful proxy? Many thanks, Phil --- ## WARNING - DO NOT USE THIS IN A PRODUCTION SERVER. FOR TESTING & EDUCTION ONLY debug=2 fork=no log_stderror=yes listen=192.168.1.254:5060 ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "sl.so" loadmodule "usrloc" modparam("usrloc", "db_mode", 0) loadmodule "xlog.so" loadmodule "registrar.so" ####### Routing Logic ######## route{ if(method=="REGISTER") { if (!save("location")) { xlog("L_ERR", "**REGISTER: save failed \n"); sl_reply_error(); } else { xlog("L_NOTICE", " REGISTER rcvd $si:$sp \n"); }; }; if(method=="INVITE") { lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; sl_send_reply("302","Redirect"); }; if(method=="PUBLISH" || method=="SUBSCRIBE"){ xlog("L_INFO","REJECTION. We do not process $rm requests. \n"); route(No); }; } route[No] { sl_send_reply(503, "Service Unavailable"); } --- U 2014/10/05 16:38:04.925388 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. <...SNIP...> U 2014/10/05 16:38:04.925895 192.168.123.254:5060 -> 192.168.123.16:5060 SIP/2.0 302 Redirect. Via: SIP/2.0/UDP 192.168.123.16:5060 ;branch=z9hG4bKaf202530a743a053;rport=5060. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. Contact: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>. Server: My SIP Server. Content-Length: 0.U 2014/10/05 16:38:04.954405 192.168.123.16:5060 -> 192.168.123.254:5060 ACK sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: path. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 ACK. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Length: 0. . U 2014/10/05 16:38:04.975398 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKa55f284fa742ad32. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=9a25392722276c54. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: f2989990cfa8525c(a)192.168.123.16. CSeq: 1445 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. . . . etc...

4 5

how to always use rtpproxy-ng in kamailio
by andrew 05 Oct '14

05 Oct '14

Hi, I have one question about kamailio. rtpproxy-ng is installed with kamailio. When client A initiates one call to client B, SDP in invite/200ok are rewritten, where ice candidates for rtpproxy-ng are added. Sometimes rtp packets are relayed through rtpproxy-ng, but in some cases rtp packets pass through route created by ICE algorithm. How to set kamailio, then rtpproxy-ng is always used? B.R. andrew

2 1

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 04 Oct '14

04 Oct '14

Anybody have some insight or pointers? Thx

2 1

channel parameter on SDP
by Kelvin Chua 03 Oct '14

03 Oct '14

Hi guys, my provider is being a dick. they do not accept doubango's a=rtpmap:0 PCMU/8000/1 what they wanted was a=rtpmap:0 PCMU/8000 is there a way of changing this on kamailio? Kelvin Chua

2 1

Routing calls to a subscribed softphone
by Will Ferrer 03 Oct '14

03 Oct '14

Hi I was wondering if any one had any advice or examples for me of how to get a call to be routed to a subscribed softphone. We have 2 boxes in our testing deployment, a load balancer / sbc and a call processing box. Calls come in to the sbc, and then are passed to the call processing box. The call is analyzed and the branch uri is rewritten to a destination address when applicable for the call (this is how we handle routing of calls to certain numbers -- we do this utilizing custom code and a custom db). This works just fine when the destination sip uri is phone number (in which case we do lcr) or if the destination goes to a remote address. However when the destination is a subscriber: sip:[subscriber_name]@[our_domain_name], the call is passed back to the sbc, which passes it to the callprocessing box, back and forth until a too many hops error occurs. The subscriber I am trying to send the call too does show up under "kamctl ul show". I feel like there is something basic I must be missing here. Does any one have any advice for me? Thank you very much in advance. All the best. Will Ferrer

2 5

Enable SipTrace
by Bruno Emer 03 Oct '14

03 Oct '14

Hello! I just got some kamailio servers working with websockets using jssip in my clients and now I want to get the SIP Messages to better understand how it flows between my servers. I have enabled the "siptrace" module, and I can now save information on my database, but the problem is that I can't read the SIP Messages there, as it saves some strings in my database. The SIP messages appears like this store in the sip_trace table: \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 I am currently using postgres as my database, and to get this I just have: - Define "WITH_SIP_TRACE" in the beginning of my configuration file: #!KAMAILIO #!define WITH_PGSQL ##!define WITH_DEBUG #!define WITH_SIP_TRACE - loadmodule: #!ifdef WITH_SIP_TRACE loadmodule "siptrace.so" #!endif - Configure SIP_TRACE params: #!ifdef WITH_SIP_TRACE # ---- sip_trace params ---- modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost /kamailio") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) #!endif - Get the messages: request_route { sip_trace(); setflag(22); route(REQINIT); ... Does anyone knows if is there something wrong in this configuration, or what can I do to fix this and have the SIP messages stored in my database? Thanks! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 2

Re: [SR-Users] Kamailio & queue
by Robert Boisvert 03 Oct '14

03 Oct '14

Only with rtpproxy. On Fri, Oct 3, 2014 at 5:25 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > mohqueue module is also compatible with rtpproxy-ng or only with rtppoxy? > > Because during startup i've noticed this error: > mohqueue : Unable to load rtpproxy_stream2uac > > > > > >There was a typo in the parameter name. If you use default values, just > remove the modparam. > >Otherwise, I pushed the patches to master and 4.1 branches. So you would > need to upgrade to latest git version. > >Cheers, > >Daniel > >>On 02/10/14 10:15, Marino Mileti wrote: > >>I don't need any IVR so I would like to try this module and following doc > i've setup dbtable name parameters: > >>modparam ("mohqueue", "db_qtable", "mohqueues") > >>modparam ("mohqueue", "db_ctable", "mohcalls") > >>but when start Kamailio says: > >>ERROR: <core> [modparam.c:156]: set_mod_param_regex(): > set_mod_param_regex: parameter <db_qtable> of type <1> not found in module > <mohqueue> > >>What's wrong? > > 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 > > > > --- > Questa e-mail è priva di virus e malware perché è attiva la protezione > avast! Antivirus. > http://www.avast.com > > > _______________________________________________ > 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 >

1 0

[OT] About next Kamailio World, May 27-29, 2014, in Berlin
by Daniel-Constantin Mierla 03 Oct '14

03 Oct '14

Hello, we have preliminarily booked May 27-29, 2014 for the next Kamailio World event, hosted again by FhG Fokus Institute in Berlin. It was really hard to find a time frame during April and May 2015, because of different dates for Catholic and Orthodox Easters, 1st of May as well as quite some many public holidays in Germany. I am writing to see if anyone is aware of any major blocker for that time frame. Write me directly and we will see if anything can be done, selected the best options based on constraints. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio & queue
by Marino Mileti 03 Oct '14

03 Oct '14

I don't need any IVR so I would like to try this module and following doc i've setup dbtable name parameters: modparam ("mohqueue", "db_qtable", "mohqueues") modparam ("mohqueue", "db_ctable", "mohcalls") but when start Kamailio says: ERROR: <core> [modparam.c:156]: set_mod_param_regex(): set_mod_param_regex: parameter <db_qtable> of type <1> not found in module <mohqueue> What's wrong? ----Messaggio originale---- Da: miconda(a)gmail.com Data: 2-ott-2014 9.44 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org>, "Marino Mileti"<marino.mileti(a)alice.it> Ogg: Re: [SR-Users] Kamailio & queue Thats true if you need to answer and do some IVR, otherwise you can hold the INVITE or other request (until it is cancelled or times out). Specific on a call queue, may worth checking the module: - http://kamailio.org/docs/modules/stable/modules/mohqueue.html Holding the invite in config file is possible with: - t_suspend() and then the (index, label) identification of transaction -- see tmx module - add (index,label) to mqueue - when the routing of invite has to be resumed, fetch from the queue and do t_continue() Cheers, Daniel

2 2

simplest config file possible - not working!
by White, Phil 02 Oct '14

02 Oct '14

Sorry Guys, I'm very new to this. Taking a dive 'under the hood', to try to learn what is going on, and to see what I can achieve. I have 1 sip device ( voip telephone) on the network, constantly firing out REGISTER requests (I have turned off my asterisk server!). That's good. I have a terminal window open, running tcpdump. That's good too - I can see what is going on. I also have kamailio - giving me absolutely nothing. It's not complicated config file. In fact it is the simplest I could do. However, it logs nothing - and i am expecting it to. Can someone put me out of my misery, and point out the glaring error I have? How can I send messages to STDERR while testing? Cheers, Phil ---- ## WARNING - FOR TESTING & EDUCTION ONLY ####### Global Parameters ######### debug=5 fork=no log_stderror=yes port=5060 ####### Modules Section ######## # set module path mpath="/usr/local/lib/kamailio/modules/" loadmodule "sl.so" ####### Routing Logic ######## route{ if(method=="REGISTER") { log(1,"this SIP request is a REGISTER message\n"); sl_send_reply("404", "No registrar"); }; if (af!=INET6) { log(1,"Message received over IPv4 link\n"); }; if(proto==UDP) { log(1,"SIP message received over UDP\n"); }; if(status=="200") { log(1,"this is a 200 OK reply\n"); }; if(uri==myself) { log(1,"the request is for local processing\n"); }; }

3 5

RFC: updating default values
by Daniel-Constantin Mierla 02 Oct '14

02 Oct '14

Hello, I am opening this discussion to decide if there is need to adjust some of the default values we have in Kamailio. Many of them were set even like 10 years ago, so they might not be very actual anymore. The main goal is to get the best possible settings for common usage. To start with, here are some values that should be reviewed: - default private memory is 4MB - if the config is not that small, it might not be sufficient free pkg to play with (e.g., for sql_query() result, storage of $var(...) values). Should it be increased to 8MB or other value? Debian/Centos have startup script that sets its value to 8 via command line - default shared memory is 32MB - for a decent deployment with tm, location, lcr/dispatcher, permissions, and anti-flood, it might leave not much free space. Should we double it or set to a different value - usrloc - db_ops_ruid should enabled (1) - seems stable, without it there are problems updating/deleting location records when UA changes the call-id for same contact address. - usrloc - hash_size - now is 9, which results in 512 slots, meaning is ok for few thousands of registered users, for more, performance will decrease when doing save/lookup location -- should it be made 10 (1024 slots for internal hash table)? - auth_db - load_credentials defaults to 'rpid', meaning that the query to get the password will retrieve also the rpid column. I haven't see rpid being used that much lately (replaced by PAI/PPI). I would make this parameter empty by default to avoid querying for an extra column that is likely to be empty. Perhaps there are more, I just wanted to get started. Reply with your comments to above list as well as add new items you thinks their default values should be adjusted. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 6

Problem with locally generated BYE in dialog timeout.
by Julia Boudniatsky 02 Oct '14

02 Oct '14

Hello, We use dialog module in our configuration. uac_replace_from/to is called after dlg_manage(). restore_mode is auto. URIs are modified automatically in all subsequent local requests, exclude BYE generated in dlg timeout. In this case original URIs are sent to both caller and calee sides. If call going through only one kamailio server, all works fine. Problem starts with a chain of servers with from/to manipulation. When timeout occurs in one kamailio server, calee kamailio server receives BYE with original (not modified) URIs and URI restore gets wrong parsing result. Call clears in caller side and stay connected in caller. Whether there is any solution? BR , Julia.

3 4

Kamailio & queue
by Marino Mileti 02 Oct '14

02 Oct '14

Hi, Is it possible to use kamailio to manage a very simple queue? All calls inbound go directly in queue and then the system to to extract calls and route to agent. A very simple call center. I've seen that there's a mohqueue module...is there any conf example? Many thanks

3 2

event_route[core:receive-parse-error] question
by Julia Boudniatsky 02 Oct '14

02 Oct '14

Hello, I try to use event_route[core:receive-parse-error] in 4.1.6 event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } When core parse error occurs , *log message is not printed*. What is wrong? Thank you, Julia

2 3

Segfault in 4.1.6
by David Wilson 02 Oct '14

02 Oct '14

Hi all, I’ve had Kamailio running nicely for nearly a year, but have hit a segfault twice since upgrading to 4.1.6 earlier this week. The log shows: Oct 2 11:28:03 localhost /usr/sbin/kamailio[1222]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 Oct 2 11:28:03 localhost kernel: [11259.904645] kamailio[1206]: segfault at 58 ip 00007fa56fba0f86 sp 00007fffea594ca0 error 4 in dialog.so[7fa56fb63000+54000] Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:777]: handle_sigs(): child process 1206 exited by a signal 11 Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:780]: handle_sigs(): core was not generated Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: INFO: <core> [main.c:792]: handle_sigs(): INFO: terminating due to SIGCHLD Oct 2 11:28:03 localhost /usr/sbin/kamailio[1220]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received …. The first two lines above seem to be the important parts. Nothing was logged for about 10s prior to this extract. An earlier instance of the issue showed almost the same details: Oct 1 20:39:12 localhost /usr/sbin/kamailio[1219]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 Oct 1 20:39:12 localhost kernel: [554502.639227] kamailio[1207]: segfault at 58 ip 00007fd62c5c6f86 sp 00007fffd77c9890 error 4 in dialog.so[7fd62c589000+54000] I’m trying to catch a core next time it occurs, no core is available yet. Is this a new issue in 4.1.6? Any suggestions about the likely cause? Cheers, Dave.

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Relaying ACK to Asterisk
by Igor Potjevlesch 01 Oct '14

01 Oct '14

Hello, I'm having trouble with this scenario (Kamailio and Asterisk are working on the same server, Asterisk listens on 4060 instead of 5060): the UAC sends an ACK request with the following R-URI: sip:955*95%23@ <sip:955*95%23@%3cIP_ASTERISK/KAMAILIO%3e:4060> <IP_ASTERISK/KAMAILIO>:4060. When I'm doing a capture on loopback interface, I just see an ACK request from IP <IP_ASTERISK/KAMAILIO>:5060 to IP <IP_ASTERISK/KAMAILIO>:5060. So the ACK seems to loop inside Kamailio. What could explain that the good port defined by the UAC is deleted? Regards, Igor.

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rtpengine alt stream question
by Alex Balashov 30 Sep '14

30 Sep '14

Hello, I recently encountered SDP attributes that look like this: Media Attribute (a): alt:1 3 : njfxofkg bavpjfxg 192.168.2.59 41978 Media Attribute (a): alt:2 2 : fpukfyaj dqmiarjx 192.168.111.1 41978 Media Attribute (a): alt:3 1 : euvrwenk jctmhavh 192.168.238.1 41978 I was not familiar with these. However, RFC 4796 says: alt: the media stream is taken from the alternative source. A typical use case for this is an event where the ambient sound is separated from the main sound. The alternative audio stream could be, for example, the sound of a jungle. Another example is the video of a conference room, while the main stream carries the video of the speaker. This is similar to the 'live' role in H.239. RFC 6064 http://tools.ietf.org/html/rfc6064#section-4.4 speaks to this a little as well. This does not really make any clearer to me what these mean in this scenario, and unfortunately I cannot find out from the user. My question is: I assume that rtpengine (both Kamailio and daemon-side) does not support this, correct? Are there any plans to in the future? Can anyone with more experience with video, perhaps, give some insight into the meaning and application of this attribute? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 30 Sep '14

30 Sep '14

Unfortunately rtpengine doesn't work in this way. At the end of the calls this is the output log: Final packet stats: Tag 'Fw3D7R0', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30224 <> 192.168.10.20:7078 , 540 p, 92880 b, 0 e Media #1, port 30225 <> 192.168.10.20:7079 (RTCP), 3 p, 324 b, 0 e Media #2, port 30256 <> 192.168.10.20:9078 , 0 p, 0 b, 0 e Media #2, port 30257 <> 192.168.10.20:9079 (RTCP), 3 p, 264 b, 0 e Tag 'qWE6Gsh', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30140 <> 192.168.10.50:7078 , 533 p, 91068 b, 0 e Media #1, port 30141 <> 192.168.10.50:7079 (RTCP), 5 p, 444 b, 0 e Media #2, port 30170 <> 192.168.10.50:9078 , 0 p, 0 b, 0 e Media #2, port 30171 <> 192.168.10.50:9079 (RTCP), 1 p, 88 b, 0 e Tag 'TTPyT~Hdw', created 0:41 ago, in dialogue with 'Fw3D7R0' Media #1, port 30206 <> 172.20.11.208:7078 , 1070 p, 183736 b, 0 e Media #1, port 30207 <> 172.20.11.208:7079 (RTCP), 4 p, 496 b, 0 e Media #2, port 30240 <> 172.20.11.208:9078 , 4188 p, 1435946 b, 0 e Media #2, port 30241 <> 172.20.11.208:9079 (RTCP), 4 p, 400 b, 0 e 192.168.10.x clients are natted..it seems that rtpengine open 2 ports (for example video) for each receiver (30256 & 30170) and 1 port for the caller (30240). But on the INVITE of Kamailio only video port 30170 is offered to receivers, instead on caller side there are 2 distinct 183s message that offer 30190 & 30240. It's a little bit strange because some of these port doesn't appear in the log of rtpengine. At the end I can see video only on one receiver I don't know if the problem is on Kamailio (rtpproxy-ng module) or in the rtpengine :) > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers _______________________________________________ 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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Call Group versus MAX_BRANCHES limit
by João Vitor Arruda 30 Sep '14

30 Sep '14

Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a parallel forking when lookup() is used) in sequence (similar to serial forking). Currently I have written code that is similar to the one described here: http://www.kamailio.org/dokuwiki/dokuphp/tutorials:avpops#serial_forking The only difference is that for each member I call lookup() and then in the failure_route(), I pull the next member from the stack and call lookup() again. The code works fine when used for groups with less than 12 members. However, call groups with 12 or more members fails as I can never reach remaining extensions once MAX_BRANCHES limit is reached. I've tried another approach using the functions described here: http://kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.serial_forking but it too uses a new branch for every group member until the 12 limit is reached out. I also tried to use the functions "remove_branch(index)" and "clear_branches()" (both of which are poorly documented. In fact the only reference I ever found for these functions was here: http://www.kamailio.org/wiki/features/new-in-3.2.x#functions). Unfortunately, I wasn't able to prevent the MAX_BRANCHES limit from being reached. Ultimately, my goal is to have a limitless Call Group. Do you have any suggestions? (other than increasing the hard coded MAX_BRANCHES limit) Thank you, Joao Arruda

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Re: [SR-Users] Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hi,I ran the code on my server (as Daniel suggested) and then tried to send message from one user to other but no traffic was detected.ngrep d any qt W byline "sip" port 5060interface: anyfilter: (ip or ip6) and ( port 5060 )match: sipAlso, after I tried to install the PRESENCE module and having restarted the systems today, I noticed that the response that I am getting on Jitsi is "403 Not Relaying" and not "403 Not Allowed". I am not sure what caused the change.I hope someone can help me figure it out.Cheers!Gaurav Original message From:"DanielConstantin Mierla"< miconda(a)gmail.com >Date: 30 Sep 14 16:19:30Subject: Re: [SRUsers] Kamailio with Jitsi Presence troubleTo: ; "Kamailio (SER) Users Mailing List"Hello,you should grab the SIP traffic on kamailio server and send it to the mailing list. In this way we can spot why you get the 403 Not Allowed. You can use:ngrep d any qt W byline "sip" port 5060Cheers, DanielOn 30/09/14 08:50, Gaurav Kumar wrote:Hello all!I tried sendin g this earlier too but since I was not subscribed tothe mailing list, it bounced back.I am trying to setup a secure videoconferencing setup for mysmall office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very fewusers (less than 10, at most) to use it through Jitsi. I am tryingto follow the tutorials available at: http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://www… am able to log into Jitsi on independent machines, both behindand without a firewall. But the trouble I am facing is that theusers do not appear online to each other. If I try to send a textmessage nonetheless, I get the "403 Not allowed" error.At first, I tried the config file for Kamailio provided on thefirst link. It had some problems due to it being an old versionand the config file on the 2nd link help ed me out. I suspect thereis some problem with the PRESENCE module but I do not know whatsince it does not give any error whatsoever. I added the "#!defineWITHPRESENCE" line and installed the Presence module but to noavail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00bfor Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a weeknow. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now! SIP Express Router (SER) and Kamailio (OpenSER) srusers mailing list srusers(a)lists.siprouter.org http://lists.siprouter.org/cgibin/mailman/listinfo/srusersDanielConstantin Mierla http://twitter.com/#!/miconda http://www.linkedin.com/in/micondaGet Yourself a cool, short @in.com Email ID now!

1 0

Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hello all!I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back.I am trying to setup a secure videoconferencing setup formy small office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very few users(less than 10, at most) to use it through Jitsi. I am trying to followthe tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… able to log into Jitsi on independent machines, both behind andwithout a firewall. But the trouble I am facing is that the users do notappear online to each other. If I try to send a text messagenonetheless, I get the "403 Not allowed" error.At first, I triedthe config file for Kamailio provided on the first link. It had someproblems due to it being an old version and the config fil e on the 2ndlink helped me out. I suspect there is some problem with the PRESENCEmodule but I do not know what since it does not give any errorwhatsoever. I added the "#!define WITHPRESENCE" line and installed thePresence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now!

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Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
by Julia Boudniatsky 30 Sep '14

30 Sep '14

Hello, Some Uri transformations stopped working after upgrade from 3.3 to 4.1.6. *To: <sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060 <http://noind@10.25.153.150:5060>;user=phone>* *In v3.3* $tu({uri.params}) return: cic=012;csel=noind $tu({uri.param,cic}) return: 012 *In v4.1.6* $tu({uri.params}) return : user=phone $tu({uri.param,cic}) return : Best regards, Julia

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Re: [SR-Users] Questions about multiple contacts in kamailio
by Federico Cabiddu 30 Sep '14

30 Sep '14

Hi, you can achieve this by using the flag 0x04 when calling save() function from registrar module: http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1965144 Regards, Federico On 30 Sep 2014 10:54, "qw" <applemax82(a)163.com> wrote: Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew _______________________________________________ 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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Questions about multiple contacts in kamailio
by qw 30 Sep '14

30 Sep '14

Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew

1 0

Kamailio -PSTN Gateway
by Zaka Ul Isam 30 Sep '14

30 Sep '14

Hello Folks: Please help with above, I have browsed and tried various suggestions on this list without much luck! I think problem can be reduced to three questions ;) 1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be compiled? (DIALPLAN, CARRIERROUTE, LCR aso) If yes, then we ought to put up a list of modules required for each task/ activity..... 2: How Kamailio core decides the call routing? i.e: based on dialed digits as specified in dialplan . # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00355)[0][4-9]{3,20}$")) return; && #!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 = "10.1.6" desc "PSTN GW Address" pstn.gw_port = "5080" desc "PSTN GW Port" #!endif 3: If Kamailio is listening on say wlan0 interface 10.42.0.1:5060 (Ubuntu HOT SPOT) can it route to PSTN Gateway/ Softswitch on P2P1 (Eth0) without Bridging? Put another way, does kernel route towards the specified interface or Kamailio is capable of routing based on active routing cache? Thanking you in advance, Zak ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

2 2

kamailio call limit
by Yuriy Gorlichenko 30 Sep '14

30 Sep '14

Hello. I try to test with SIPp my stak of kamailio->asterisk. I run SIPp with 200 calls/sec and see only 68 at maximum active calls at server. When I set 500 calls/sec with limit 1000 I see 68 active connections again. So when I try test SIPp to asterisk without Kam i see wright maximum of active channels/calls. My question - does Kamailio have limits ffor calls? If no- help me to understand please what is going on with my server. Thanks.

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Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:51, Marino Mileti wrote: > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers

1 0

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:23, Marino Mileti wrote: > The problem isn't on 183s but on the multiple INVITE that Kamailio sends to > clients behind rtpengine. Rtpengine open new ports for answer but on INVITE > the rtpengine ports are the same...This happens because for all these > clients the callid is still the same..so for rtpengine there's no > difference...for this reason I can see early media only on one of the > receivers I still don't understand what difference an RTP proxy is supposed to make for the invite. Your offering client only opens one endpoint and sends that out. Without RTP proxy, all receiving clients see the same endpoint and they will all send their early media to that same endpoint. With RTP proxy, the same thing happens. What's the difference? cheers

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:29, Marino Mileti wrote: > Wow! Do you have an example of how to do that? How I have to modify my > kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in > this way? You don't have to do anything, tags are already included in all the messages. cheers

1 0

Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:08, Marino Mileti wrote: > But with from-tag and To-tag it's possible to instruct rtpengine to generate > new couple of ports for each branch of a call? In the source code of > rtpengine it seems that it check only the callid parameter Yes it will. The call-id is only a vague umbrella under which the from/to tags are collected. All media parameters, ports etc are associated with the tags, and a single call can contain any number of different from/to tags, all with their own separate parameters. cheers

2 1

R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

No no. The video will be sent by the caller user to all the callees. I'l try to explain better. My scenario is: - A make a call to a group... B & C are group member...so Kamailio is able to call them in parallel using alias.. - B & C receive the INVITEs & replies with two separate 183 with SDP (in SDP they specified where they are able to receive audio&video) - A receives two 183...& starts to send its RTP video stream to B & C (early media) - Once B or C answers the call the other leg is cancelled.. Everything is working fine but if B & C are behind NAT rtpproxy is activated and during INVITE for B &C rtprpoxy doesn't generate a couple of new ports for each of them but it uses always the same ports. So the only fastest client (B or C) get the video. I don't want to change IP between 183 & 200, i would like only that rtpptoxy sends INVITE for B & C with differents port.Is it possible to implement this scenario or there's some turnarund? ----Messaggio originale---- Da: frank(a)carmickle.com Data: 25-set-2014 16.22 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] R: Re: RTPPROXY & BRANCH On Sep 25, 2014, at 10:09 AM, Marino Mileti <marino.mileti(a)alice.it> wrote:Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) I don't understand. Are you saying that you have clients that when they receive an invite sent video with 183? How do you want to composite the video to show to the caller? It is not RFC3261 compliant to change IP and port from 183 to 200. Of course you can reinvite after the 200. Most B2BUAs require you to ignore early media and generate something locally to send to the caller or just send them 180. Maybe if you explain your use case someone can help you. --FC

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/26/14 16:57, Marino Mileti wrote: > Hello, > >> On Friday 26 September 2014 16:44:44 Marino Mileti wrote: >>> Hi guys, >>> I've seen that setting the parameter extra_id_pv, every branch should >>> be a different callid.. >>> How can i set this parameter? I've tried with : >>> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") >>> >>> but in the INVITE message the callid is still the same for all >>> branches. Any suggest? > >> Did you assign a value to $avp(extra) in the script, before calling any of > the rtpproxy functions? > >> Did you use the 'b' parameter in the call to rtpproxy_*() functions? >> -- >> Alex Hermann > > Yes, i've assigned a value in the script, just to try I've assigned $rU and > on the log of rtpengine I can see the value of "via-branch" valorized > correctly. I've also used the 'd' parameter in rtpproxy_manage() function. > > I've also checked on the source code of rtpengine and the part regarding the > "parse & get" of via-branch value is commented...I'm using the master branch > of rtpengine. > > This is the piece of code of call_interfaces.c where the via-branch > parameter is commented: > //bencode_dictionary_get_str(input, "via-branch", &viabranch); The via-branch is currently unused in rtpengine, even though it does appear in the control messages. From-tag and to-tag are sufficient to identify a unique dialogue within a call. cheers

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R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ 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

3 3

rtpproxy_offer and fix_nated_sdp in one route
by Sebastian Damm 29 Sep '14

29 Sep '14

Hi, I switched from rtpproxy module to the rtpproxy-ng module lately, and noticed a strange behavior. In my branch route to the device, I have two statements: fix_nated_sdp("1"); rtpproxy_offer(); The first command appends a line with "direction:active" to the SDP. The second one puts the RTP proxy in the stream. This worked all well with the old rtpproxy module. But with the new rtpproxy-ng module, I get an empty line after the SDP body, just before the "direction:active" line in the SDP, which makes the packet invalid. I tried changing the order of both statements, but without any difference. What I saw is, that the old module just sent some basic parameters to the rtpproxy and got only IP and port back. The new module sends the complete SDP to the rtpengine and gets back the "fixed" SDP. Has anyone ever seen this? Is there a way to fix it? Best Regards, Sebastian

3 4

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } corelog=1 debug=0 kamailio 4.1.6 from source Wrong header "From" is simulated by SIPP. In log I receive only ERROR:<core> [parser/parse_from.c:79]: parse_from_header() ...... Thank you, Julia

1 0

Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 29 Sep '14

29 Sep '14

Hello all!I am trying to setup a secure videoconferencing setup for my small office. After a lot of convincing, my bosses have allowed me to create a setup and have given me a live IP. I have used it on a Ubuntu 12.04 setup and want to setup a SIP server for very few users (less than 10, at most) to use it through Jitsi. I am trying to follow the tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… am able to log into Jitsi on independent machines, both behind and without a firewall. But the trouble I am facing is that the users do not appear online to each other. If I try to send a text message nonetheless, I get the "403 Not allowed" error.At first, I tried the config file for Kamailio provided on the first link. It had some problems due to it being an old version and the config file on the 2nd link helped me out. I suspect there is some problem with the PRESENCE module but I do not know what since it does not give any error whatsoever. I added the "#!define WITHPRESENCE" line and installed the Presence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now!

1 0

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); }

1 0

Does kamailio reloads the modules per sip message ?
by Rahul MathuR 29 Sep '14

29 Sep '14

Hello, We have observed a strange behavior in corex module that it gets loaded at every sip packet which arrives to kamailio. We put a static variable and saw that it gets re-initialized to 0 everytime any sip packet comes to it. Could you please tell me how to stop it and load it just once. Thanks in advance ! -- Warm Regds. MathuRahul

2 1

DBTEXT Engine & ALIAS_DB
by Marino Mileti 29 Sep '14

29 Sep '14

Hi to all, I've setup FS with dbtext module, and now I'm trying to make a alias_db_lookup(dbaliases) but it doesn't find any alias.Same conf with mysql db works great. Is it possible to have alias with dbtext engine? Marino

2 1

tm.t_uac_wait documentation?
by Juha Heinanen 29 Sep '14

29 Sep '14

i could not find any documentation about tm.t_uac_wait rpc command. is it documented somewhere? -- juha

2 3

tm.t_uac_wait does not use advertise address
by Juha Heinanen 29 Sep '14

29 Sep '14

based on more tests, it appears that tm.t_uac_wait does not use advertise address (if given on listen line) when it substitutes !! (SUBST_CHARs) in request headers. i'll open a bug report on it. -- juha

2 1

cnxcc serial fork
by Kelvin Chua 29 Sep '14

29 Sep '14

hi carlos, i have a serial forking scenario to several providers as failover. and i used cnxcc_set_max_credit to control the session credit/time. just wanted to make sure it does what i think it should do so i made a little experiment. the logic goes this way. 1. cnxcc_set_max_credit -> $100, cost per second -> $1 2. call will fail, try the next provider 3. cnxcc_set_max_credit -> $200, cost per second -> $2 what i am expecting to see is, if i do kamctl kamcmd cnxcc.active_clients i would see $200 as the max_amount. but in kamailio 4.1.5, i still see $100, should it be updated to $200? is there an override in cnxcc_set_max_credit when called twice or after failure_route? Kelvin Chua

2 3

rtpproxy & extra_id_pv
by Marino Mileti 26 Sep '14

26 Sep '14

Hi guys, I've seen that setting the parameter extra_id_pv, every branch should be hae different callid.. How can i set this parameter? I've tried with : modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") but in the INVITE message the callid is still the same for all branches. Any suggest?

2 2

Kamailio server crashing
by aawaise 26 Sep '14

26 Sep '14

Hello, I installed kamailio and replaced kamailio.cfg with a customized kamailio.cfg and replaced kamctlrc file also. And restarted the kamailio server. Server gets started ok and after a minute or two, I get the error, " '/usr/local/kamailio/sbin' doesnot belong to any package ". I am facing this problem for quite some time now. Any solution to this will be very helpful. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-server-crashing-tp130923.h… Sent from the Users mailing list archive at Nabble.com.

2 1

Bash Code Injection and 'exec' module
by Seudin Kasumovic 25 Sep '14

25 Sep '14

Hi kamailio users, we are witnesses of new discovered bug in bash: Bash Code Injection Vulnerability via Specially Crafted Environment Variables (CVE-2014-6271) https://access.redhat.com/node/1200223 As exec module exports all SIP headers in environment so it's was easy to push bash command. There is attached simple kamailio test config file. With sipp we sent header to output 123 into file /tmp/123 like this: User-Agent: () { :;}; echo 123 > /tmp/123 Debug output from kamailio is: 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_LENGTH=135 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_TYPE=application/sdp 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH * 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_USER_AGENT=() { :;}; echo 123 > /tmp/123* 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_SUBJECT=Performance Test 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_MAX_FORWARDS=70 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTACT=< sip:T00157@198.51.100.2:5060> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CSEQ=1 INVITE 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CALLID= 1-5394(a)198.51.100.2 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_TO=+442033998806 <sip:+442033998806@orange.voip> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_FROM=+442033998833 <sip:T00157@orange.voip>;tag=5394SIPpTag001 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_VIA=SIP/2.0/UDP 198.51.100.2:5060;branch=z9hG4bK-5394-1-0 5(30147) DEBUG: exec [exec_mod.c:175]: w_exec_msg(): executing [/bin/true] ls /tmp shows new created file !!! I created simple patch to fix this issue in exec module based on suggestion from RedHat until you fix your bash what is recommended. -- Seudin Kasumovic

2 4

Re: [SR-Users] Kamailio - Separated Servers
by Bruno Emer 24 Sep '14

24 Sep '14

Hello Guys! Good news about my tests here! I found some configurations relative to Outbound proxy and I get this working. Now, I can have two outbound proxies, one registrar and one database. I can register one user in the Outbound Proxy1 and one user in Outbound Proxy 2 and place a call between these users. But now I'm facing another problem! The ACK messages are not being delivered, so the session is being dropped after a few seconds. I think that the problem is in the Contact field. Trying to investigate it now! I'm not sure about how to change the message to deliver the ACK. Did someone had this problem before? In anycase, here is the routing configuration that I'm using in my Oubound proxy: request_route { if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } route(REQINIT); if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); t_check_trans(); if (is_method("REGISTER")) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { if (is_method("INVITE|SUBSCRIBE|ACK")) record_route(); if (@via[2] == "") { # From client so route to registrar... if ($rU == $null) { sl_send_reply("484", "Address Incomplete"); exit; } remove_hf("Route"); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { # From registrar so route using "Route:" headers... if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: xlog("L_ERR", "in request_route\n"); sl_reply_error(); exit; } } t_on_failure("FAIL_OUTBOUND"); } } route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { 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; } } route[WITHINDLG] { if (has_totag()) { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: if (is_method("ACK")) { if ( t_check_trans() ) { record_route(); route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); } } else { if (is_method("NOTIFY")) { record_route(); } route(RELAY); } exit; } } onreply_route { if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (!t_check_trans()) { drop; } if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299) { remove_hf("Flow-Timer"); if ($(hdr(Require)[*])=~"outbound") insert_hf("Flow-Timer: FLOW_TIMER\r\n", "Call-ID"); } } failure_route[FAIL_OUTBOUND] { if (t_branch_timeout() || !t_branch_replied()) { send_reply("430", "Flow Failed"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } Thanks again!!! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2014-09-22 12:40 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hi, Daniel and Frank! > > Thanks for your support! > Now, I changed my scenario just a lil bit to make it easy to understand > and deploy. I'm doing the following: > > 1. Created a server using the "Outbound" module configuration example, > but add WebSocket support in it. (IP Address 192.168.1.16) > 2. Created a server running as Proxy and Registrar with WebSockets > support. (IP Address 192.168.1.15) > 3. Registered two users (bob and alice(a)enterprise.com) using WebRTC. > These users sent they Register requests to the Oubount server > (192.168.1.16). > 4. I'm also using an external postgres database to store the user > informations. In my proxy/registrar server, I've configurated > "modparam("usrloc", "db_mode", 3)", so it will not store data in memory. > > > After register the users in my scenario, I can see the following > information stored in my database (output of kamctl db show location): > > id | ruid | username | domain | > contact | received | path | > expires | q | callid | cseq | last_modified | > flags | cflags | user_agent | socket | methods | > instance | reg_id > > -----+----------------------+----------+----------------+------------------------------------------------+-----------------------+------+---------------------+----+------------------------+------+---------------------+-------+--------+-------------------+---------------------+---------+-------------------------------------------------+-------- > 277 | uloc-54203f71-10bd-2 | alice | enterprise.com | > sip:v4s0prah@o35ctmdnnruv.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:31:21 | -1 | udb4shpe4dnm1jsl6016s5 | 82 | > 2014-09-22 15:26:21 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:75c837d1-740a-4c62-baa0-56a06149699e> | 1 > 278 | uloc-54203f71-10bc-2 | bob | enterprise.com | > sip:lr78s270@0mkf6fgtprlh.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:34:02 | -1 | 5mpl9q4bkuqsnpk8r316j5 | 82 | > 2014-09-22 15:29:02 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:e92ef911-1ae4-45b1-bfc9-b5752b0f2fe1> | 1 > > > So, I can assume that both users are registered and ready to start > sessions. > > Now, when I try to start a new session between then (let's say bob calling > alice) what I get in the endpoint is the following message: > > SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) > Via: SIP/2.0/WS > 0mkf6fgtprlh.invalid;rport=33154;received=192.168.1.11;branch=z9hG4bK7711221 > To: <sip:alice@enterprise.com>;tag=94c1b96a0e6eb467885b5057e1e49f9e.1970 > From: <sip:bob@enterprise.com>;tag=on056iipql > Call-ID: 5e1pmj2fppbgg13doi4k > CSeq: 3989 INVITE > Server: kamailio (4.1.5 (x86_64/linux)) > Content-Length: 0 > > What I understood about this problem is that the Proxy server is trying to > forward the INVITE to the Outbound server and this server does not know > that to do with this message - like forward it to alice. > > I'm stucked in this step. > > Here you can find the configuration files that I'm using: > > Outbound Server: > https://mega.co.nz/#!Wlw2RL7S!G2YOP-1QbSCeNQddaAe3kyQMqz8yteZlZsAUpjZX4v4 > > Proxy Server: > https://mega.co.nz/#!qxAXTbbb!4zBjRtmADdl3rPCN2Yidd3xnZEjFahW3GSMb00w41xc > > > > Thanks again! > > Bruno Emer > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2014-09-18 18:04 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > > Ok, Daniel! >> >> I am trying exactly what you said, but for some reason the proxy server >> can't forward the message back to the WebSocket server. I am getting >> "487 Unresolvable destination". Appears that my Proxy Server can't get data >> from usrloc. >> >> When I register an user using WebRTC I can see that the system stores the >> information in kamailio database and in the field "contact" it puts a >> string. The Proxy server is looking in database and just find this >> information. Actually, I'm trying to understand the entire process that >> envolves passing information from an Edge Server to a Proxy server that >> sends the Register information to another Registrar server (as I described >> before). The database is shared between all the servers, and they are in >> the same network. >> >> I just read the information in another topic, as you said, but I think >> that my case is a lil bit different from it. I you agree, I can post my >> configuration files here. Maybe this can help another users with the same >> problem! >> >> Thanks again!!! >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2014-09-17 9:13 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >> Hello! >>> >>> I am relatively new to Kamailio and I'm trying to create a new >>> enviroment using it in my company. I am thinking about use Amazon to host >>> the servers and use OpsWorks to automatically escalate then if necessary. >>> To accomplish this, my idea is to separate the servers, using one dedicated >>> server to run as WebSocket, one to run as proxy and one as a Registrar. >>> I'll be using just one database to store informations to all of my servers. >>> With this, if I need more resources later, I can just create new servers >>> with the specific roles (WebSockets, Proxy, Registrar). >>> >>> By now, the idea is clear, but the point is that I don't know how to >>> separate the WebSockets server from the proxy server. Actually, I can do >>> this, but when I have one agent using a regular softphone and one agent >>> using WebSockets (with JSSIP) they are not able to establish a session if >>> the softphone user starts it. Now, I want to know if is there a way to use >>> two websockets servers, register users using both of then and start >>> sessions between then, with a separated proxy and registrar. >>> >>> Has anyone done this before? Is possible to use kamailio like this? >>> >>> >>> Thanks. >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >> >> >

3 6

Group call
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Hi guys, is possible to make a group call with kamailio? Example...i dial 1000 and kamailio make a parallel forking to 1001,1002,1003? In case of yes...every group member (1001,1002,1003) will be resolved by "location"? Because some member is natted so RTPPROXY should be engaged Is there any examples? BRMarino

4 3

Re: [SR-Users] Sip Event Filter
by Raúl Alexis Betancor Santana 24 Sep '14

24 Sep '14

Hi, you are facing the problem the wrong way. You could do that easy on the UAC side. For example, with Asterisk as UAC, you only need to pass a 10s timeout as paran to the dial app. A sip-proxy its not for what you are trying to acomplish. What you are trying to do it's a UAC or B2BUA task. Best regards. El 24/09/2014 09:00, Delifisek Tux <delifisek.tux(a)gmail.com> escribió: > > Hello, > > Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) > > I had auto dialer and I have problem with unreachable numbers. > > So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. > > After some googling I found a sip proxy may solve my problem. > > I setup kamailio and manage to work with an example edge proxy setup. > > Then what ? > > My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. > > Which module should I look after. > > My Best Regards. > >

1 0

Sip Event Filter
by Delifisek Tux 24 Sep '14

24 Sep '14

Hello, Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) I had auto dialer and I have problem with unreachable numbers. So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. After some googling I found a sip proxy may solve my problem. I setup kamailio and manage to work with an example edge proxy setup. Then what ? My wish is, after the number dialed, start to filter every sip message, then let say if I do not get 183 session in progress in 10 seconds, or get a 10 trying in 10 seconds hangup line. Which module should I look after. My Best Regards.

1 0

upgrade question to Kamailio 4, how to handle ruid in location/aliases
by Klaus Darilion 24 Sep '14

24 Sep '14

Hi! I have to upgrade a production server from 3.x to 4.x. 4.x added a unique constraint on ruid. I have plenty of entries in aliases and usrloc table. How do I update these entries to get the ruid field populated? Can I do an upgrade reusing the entries stored in the usrloc table or do I have to truncate the table and then wait until every client reREGISTERs? Thanks Klaus

2 1

R: Re: RTPPROXY & BRANCH
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ 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

1 0

RTPPROXY & BRANCH
by marino.mileti＠alice.it 23 Sep '14

23 Sep '14

Hi I've a problem with rtpproxy during a parallel ring scenario. I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I try to call them in parallel mode (ringall) rtpproxy module sends in to the INVITE the same RTP ports. Is it possible to manage rtpproxy in order to generate a couple of new port for each branch? My rtpproxy string in the kamailio conf (in manage branch section): rtpproxy_manage("bFAIE","192.168.10.10"); Brief extract of 2 SIP INVITE message send by kamailio to natted clients: 192.168.10.10 --> 192.168.10.20 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1001@192.168.10.20:6090 SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes 192.168.10.10 --> 192.168.10.50 User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:1002@192.168.10.50:6090 SIP/2.0 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1004 1476 1657 IN IP4 192.168.10.10 Session Name (s): Talk Connection Information (c): IN IP4 192.168.10.10 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 62910 RTP/AVP 111 110 0 8 101 Media Description, name and address (m): video 47160 RTP/AVP 102 Media Attribute (a): nortpproxy:yes

2 1

Wrong from/to header compose way in print_to and print_from
by dongwf 23 Sep '14

23 Sep '14

Hi kamailio dev: I am writing a module to implement some features, my version is 4.1.2, I am not sure if it is a known issue, I found: 1. When I get a INVITE request with from/to like from:"display"<aaa(a)sip.com>, then I call tm's new_dlg_uas with the sip req, now the dlg's rem_uri set as value: "display"<aaa(a)sip.com>; 2. After that, when I need use the dlg to send BYE with tm's req_within with the above saved dlg, the print_to function will check the rem_uri's first value if "<" or not, now my uri has display name, so the outgoing BYE's to header becomes to:<"display"<aaa(a)sip.com>>, is it wrong? Should we check the last char in the rem_uri as ">"? The detail code is : static inline char* print_to(char* w, dlg_t* dialog, struct cell* t) { t->to.s = w; t->to.len = TO_LEN + dialog->rem_uri.len + CRLF_LEN + ((dialog->rem_uri.s[0]!='<')?2:0); memapp(w, TO, TO_LEN); if(dialog->rem_uri.s[0]!='<') memapp(w, "<", 1); memapp(w, dialog->rem_uri.s, dialog->rem_uri.len); if(dialog->rem_uri.s[0]!='<') memapp(w, ">", 1); ....

2 6

Kamailio fronting Freeswitch authentication
by Ashwin Jain 23 Sep '14

23 Sep '14

Hi all, I am using Kamailio to front my freeswitch servers and it is working great. I am getting some problems with authentication, sometimes users which are not registered with kamailio are also able to make calls. I have a feeling that I am doing something really silly with kamailio routing. Following is the AUTH route that I am using, please let me know I am missing something very obvious. route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) 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 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; } route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more details are required. System configuration: Ubuntu: 12.04 LTS Kamailio: 4.1.5 (x86_64/linux) Freeswitch: 1.4.7 stable release -- Thanks and Regards, Ashwin Jain

2 1

Siremis v4.1.0 Questions.
by Mahmoud Ramadan Ali 23 Sep '14

23 Sep '14

Hello everyone, Please i want more clarification about what i can do with the Siremis v4.1.0 Can i configure Kamailio to load balance between multiple Asterisk servers and RTP proxy using the Siresmis web interface or i have to interface with the Kamailio configuration file and do that manually...? Thanks in advance.

4 6

User definable Ring Duration before forwarding
by Tim Chubb 23 Sep '14

23 Sep '14

Hi I was wondering if the following scenario is possible and if it is how I would go about implementing it: I would like to be able read from a database a list of destinations associated with a given dialled number which are tried in order of priority with each destination having a user definable ring duration before the next destination is tried. i.e. A Call is received from the PSTN, the number the caller dialled is looked up, it has 3 destinations attached to it, the users deskphone which is configured to ring for 10 seconds before the call is forwarded to their Mobile phone with a ring duration of 15 seconds, which if unanswered would be redirected to the main switchboard IVR. I think im looking for the equivalent of the Asterisk Dial app, where you can specify a timeout as an argument, is this possible to do with kamailio? All examples I have seen use a global timer value, which is unsuitable for this requirement, as each user could define n failover routes each with a distinct timeout before the next route is tried. Currently I have got as far as querying the database to retrieve the data but im a bit stumped as to how to handle the timeout variable. Any ideas? Tim Chubb

2 1

Kamailio - Separated Servers
by Bruno Emer 22 Sep '14

22 Sep '14

Hello! I am relatively new to Kamailio and I'm trying to create a new enviroment using it in my company. I am thinking about use Amazon to host the servers and use OpsWorks to automatically escalate then if necessary. To accomplish this, my idea is to separate the servers, using one dedicated server to run as WebSocket, one to run as proxy and one as a Registrar. I'll be using just one database to store informations to all of my servers. With this, if I need more resources later, I can just create new servers with the specific roles (WebSockets, Proxy, Registrar). By now, the idea is clear, but the point is that I don't know how to separate the WebSockets server from the proxy server. Actually, I can do this, but when I have one agent using a regular softphone and one agent using WebSockets (with JSSIP) they are not able to establish a session if the softphone user starts it. Now, I want to know if is there a way to use two websockets servers, register users using both of then and start sessions between then, with a separated proxy and registrar. Has anyone done this before? Is possible to use kamailio like this? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

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R: Re: Kamailio multihomed - BYE problem
by marino.mileti＠alice.it 22 Sep '14

22 Sep '14

This is SIP capure. Kamailio listen on 172.20.11.172 & 192.168.10.10Caller is 172.20.11.208Called user that pickup call is 192.168.10.20Other group member is 172.20.11.152 Callee (172.20.11.208) make a call to group number 5000172.20.11.208 --> 172.20.11.172--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:5000@172.20.11.172 SIP/2.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rportFrom: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrContact: <sip:1004@172.20.11.208:6090>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio replies with 100 Trying172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 trying -- your call is important to usVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrServer: kamailio (4.1.5 (i386/linux))Content-Length: 0 Kamailio lookup db and send 1st INVITE to 1st group member (1000(a)172.20.11.152)172.20.11.172 --> 172.20.11.152--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:1000@172.20.11.152:5090 SIP/2.0Record-Route: <sip:172.20.11.172;lr=on;nat=yes>Via: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrMax-Forwards: 69Contact: <sip:1004@172.20.11.208:6090>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio sends 2nd INVITE to 2nd group member (1001(a)192.168.10.20) using another network interface192.168.10.10 --> 192.168.10.20--------------Session Initiation Protocol (INVITE)Request-Line: INVITE sip:1001@192.168.10.20:6090 SIP/2.0Record-Route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-Route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-Route: <sip:172.20.11.172;lr=on;nat=yes>Via: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172CSeq: 20 INVITECall-ID: q0KGluu3SrContact: <sip:1004@172.20.11.172:5060>;+sip.instance="<urn:uuid:a6e51976-8d02-4f51-a065-0f836be30038>" Kamailio receive 100 trying from 172.20.11.152172.20.11.152 --> 172.20.11.172--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 TryingVia: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172Call-ID: q0KGluu3SrCSeq: 20 INVITE Kamailio receive 180 ringing from 172.20.11.152172.20.11.152 --> 172.20.11.172--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.172;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.0Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=puL7wn8Call-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 180 ringing to caller 172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=puL7wn8Call-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio receive 100 trying from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (100)Status-Line: SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: sip:5000@172.20.11.172Call-ID: q0KGluu3SrCSeq: 20 INVITE Kamailio receive 180 ringing from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 180 ringing to caller 172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (180)Status-Line: SIP/2.0 180 RingingVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITERecord-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio receive 200 OK from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Session Initiation Protocol (200)Status-Line: SIP/2.0 200 OkVia: SIP/2.0/UDP 192.168.10.10;branch=z9hG4bK208d.ba27ec12b58ce991e5f589b977bb7075.0Via: SIP/2.0/UDP 172.20.11.172;rport=5060;branch=z9hG4bK208d.8d88bd630566a8d23640dbb14fa4f9ef.1Via: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITEContact: <sip:1001@192.168.10.20:6090>;+sip.instance="<urn:uuid:6382d45f-042d-4be8-b6bd-1621911e88a9>"Record-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio sends 200 OK to caller172.20.11.172 --> 172.20.11.208--------------Session Initiation Protocol (200)Status-Line: SIP/2.0 200 OkVia: SIP/2.0/UDP 172.20.11.208:6090;branch=z9hG4bK.BLejEhqfD;rport=6090From: <sip:1004@172.20.11.172>;tag=TNzDNgaNNTo: <sip:5000@172.20.11.172>;tag=-iPr1gnCall-ID: q0KGluu3SrCSeq: 20 INVITEContact: <sip:1001@192.168.10.20:6090>;+sip.instance="<urn:uuid:6382d45f-042d-4be8-b6bd-1621911e88a9>"Record-route: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Record-route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio CANCEL other leg (messages removed...not so useful)-------------- Kamailio receive BYE from 192.168.10.20192.168.10.20 --> 192.168.10.10--------------Internet Protocol Version 4, Src: 192.168.10.20 (192.168.10.20), Dst: 192.168.10.10 (192.168.10.10)User Datagram Protocol, Src Port: 6090 (6090), Dst Port: sip (5060)Session Initiation Protocol (BYE)Request-Line: BYE sip:1004@172.20.11.172:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.10.20:6090;branch=z9hG4bK.20aa9gXIY;rportFrom: <sip:5000@172.20.11.172>;tag=-iPr1gnTo: <sip:1004@172.20.11.172>;tag=TNzDNgaNNCSeq: 111 BYECall-ID: q0KGluu3SrRoute: <sip:192.168.10.10;r2=on;lr=on;nat=v46>Route: <sip:172.20.11.172;r2=on;lr=on;nat=v46>Route: <sip:172.20.11.172;lr=on;nat=yes> Kamailio replies with 404 Not Here192.168.10.10 --> 192.168.10.20--------------Internet Protocol Version 4, Src: 192.168.10.10 (192.168.10.10), Dst: 192.168.10.20 (192.168.10.20)User Datagram Protocol, Src Port: sip (5060), Dst Port: 6090 (6090)Session Initiation Protocol (404)Status-Line: SIP/2.0 404 Not hereMessage HeaderVia: SIP/2.0/UDP 192.168.10.20:6090;branch=z9hG4bK.20aa9gXIY;rport=6090From: <sip:5000@172.20.11.172>;tag=-iPr1gnTo: <sip:1004@172.20.11.172>;tag=TNzDNgaNNCSeq: 111 BYECall-ID: q0KGluu3SrServer: kamailio (4.1.5 (i386/linux))Content-Length: 0 ----Messaggio originale---- Da: davy.van.de.moere(a)gmail.com Data: 22-set-2014 13.52 A: "marino.mileti(a)alice.it"<marino.mileti(a)alice.it>, "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] Kamailio multihomed - BYE problem Could you show us some SIP headers? 2014-09-22 13:50 GMT+02:00 davy van de moere <davy.van.de.moere(a)gmail.com>: A very long shot, look into handle_ruri_alias(). 2014-09-22 12:58 GMT+02:00 marino.mileti(a)alice.it <marino.mileti(a)alice.it>: Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a Kamailio v4.1.5 with two NIC (172.20.11.0 & 192.168.10.0) and three phones. Two phones (1 for each LAN) are in a group, so using alias_db I can call them in parallel mode. Caller is on 172.20 network...When I answer from phone on 192.168.10.0, RTP is ok ... but on BYE Kamailio replies with 404 Not Here...and the RURI of BYE has the IP of Kamailio instead the IP of the callee.When I answer from phone on 172.20.11.0, RTP is still ok and also BYE is managed correctly from Kamailio and the RURI of BYE has the IP of the callee. How can i solve this issue?Many thanks _______________________________________________ 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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Kamailio multihomed - BYE problem
by marino.mileti＠alice.it 22 Sep '14

22 Sep '14

Hi, I'm having a problem with routing of BYEs in my multi homed Kamailio. My setup is a Kamailio v4.1.5 with two NIC (172.20.11.0 & 192.168.10.0) and three phones. Two phones (1 for each LAN) are in a group, so using alias_db I can call them in parallel mode. Caller is on 172.20 network...When I answer from phone on 192.168.10.0, RTP is ok ... but on BYE Kamailio replies with 404 Not Here...and the RURI of BYE has the IP of Kamailio instead the IP of the callee.When I answer from phone on 172.20.11.0, RTP is still ok and also BYE is managed correctly from Kamailio and the RURI of BYE has the IP of the callee. How can i solve this issue?Many thanks

2 2

Re: [SR-Users] [OT] Accounting Drouting Gateway Selection
by Dan Christian Bogos 22 Sep '14

22 Sep '14

Hi Spencer, We use a similar approach as yours and use 302 Redirects coming from Kamailio instead of proxying the call (in our case we do it via LCR module in Kamailio). In FreeSWITCH you can then manually route these redirects and get your CDR data out of it. Cheers, DanB On 22.09.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > On 18/09/14 17:34, Spencer Thomason wrote: >> >Hello all, >> >I have a need to determine which calls go to which gateway. The challenge is that CDRs are produced by a FreeSWITCH instance upstream of a provider side Kamailio proxy that handles route selection via drouting module. The topology is like this: >> > >> >customer proxy -> FS [1-N] (accounting) -> carrier proxy (route selection) >> > >> >Does anyone have any ideas how to make the FreeSWITCH instances aware of which gateway actually received the call?

1 0

Kamailio fronting Freeswitch authentication
by Ashwin Jain 22 Sep '14

22 Sep '14

Hi all, I am using Kamailio to front my freeswitch servers and it is working great. I am getting some problems with authentication, sometimes users which are not registered with kamailio are also able to make calls. I have a feeling that I am doing something really silly with kamailio routing. Following is the AUTH route that I am using, please let me know I am missing something very obvious. route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) 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 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; } route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more details are required. System configuration: Ubuntu: 12.04 LTS Kamailio: 4.1.5 (x86_64/linux) Freeswitch: 1.4.7 stable release -- Thanks and Regards, Ashwin Jain

1 0

[OT] Accounting Drouting Gateway Selection
by Spencer Thomason 22 Sep '14

22 Sep '14

Hello all, I have a need to determine which calls go to which gateway. The challenge is that CDRs are produced by a FreeSWITCH instance upstream of a provider side Kamailio proxy that handles route selection via drouting module. The topology is like this: customer proxy -> FS [1-N] (accounting) -> carrier proxy (route selection) Does anyone have any ideas how to make the FreeSWITCH instances aware of which gateway actually received the call? Thanks! Spencer

2 1

Fwd: Kamailio 4.0.2 as a new Port for FreeBSD
by Oliver Mahmoudi 22 Sep '14

22 Sep '14

Hello, The port has just been committed to the FreeBSD ports tree and has been mapped to: /usr/ports/net/kamailio It can be compiled and installed with FreeBSD's traditional: # make install clean It compiles on 8.x, 9.x and 10.x for the i386 and x86-64 architectures. Possibly for the others archs too, but they to my knowing haven't been thoroughly tested so far. We'll find out on the way. The way things look, I will be the ports defacto maintainer. After finishing my first draft of the port about a year ago, I ran a few tests between a Kamailio server on FreeBSD and a Windows host using MySQL as a phonebook and Jitsi as a frontend GUI. From the Kamailio point of view, things worked fine, however, the only problem was, that I wasn't really able to test voice functionality, since jitsi had some problems accessing soundcard features on FreeBSD. Maybe things have changed since then and/or this will give rise to new research in this area. In case you are interested and for your cross-reference, you can find the entire discussion that took place concerning the port here: https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=181301 Moreover, to this e-mail attached, you will find a shell archive (kamailio.shar) of the port the way it has been mapped into the ports tree. Enjoy using Kamailio on FreeBSD! Oliver Mahmoudi On Tue, Jul 23, 2013 at 7:51 PM, Daniel-Constantin Mierla <miconda(a)gmail.com > wrote: > Hello, > > > On 7/23/13 1:31 AM, Oliver Mahmoudi wrote: > >> Hello, >> >> I am currently working on porting Kamailio 4.0.2 to the >> FreeBSD ports collection: /usr/ports, and I just have a >> few general questions regarding compilation with regard >> to the extra modules. >> >> There are 172 modules that come with Kamailio, and most, >> however not all of them are compiled by default: >> >> # gmake all >> >> The base server and the base modules compile fine on >> my 9.1 release. >> >> To really compile all of the 172 modules, Kamailio_4_0_2/INSTALL >> says that the following external libraries (LIB_DEPENDS I assume) >> are needed: >> >> /usr/ports/databases/mysql-client-* >> /usr/ports/databases/postgresql-libpqxx >> /usr/ports/textproc/expat2 >> /usr/ports/textproc/libxml2 >> /usr/ports/net/radiusclient >> >> This is put down as information for FreeBSD 5.4. Does that still hold >> current for 9.1? >> >> Is there an exhaustive list of all the modules that the all target >> does/does not compile and - apart from mysql, postgresql - which of the >> extra modules require which libraries? >> >> I would set compile time options accordingly. >> >> This would greatly facilitate my work! >> > one file to look in is Makefile.groups, there you will see lists with > modules organized by common dependencies. In debian, the packages are > generated for each module_group_k... Each such group includes a list of > modules defined in the same file. > > There are some comments giving hints about dependecies, but a better place > is to look in the README of the module, where is a dedicated section for > such details. > > We had some ports done long time ago, you find them in sources > pkg/kamailio/.If you get up to date version, we can replace the old one in > our repository as well. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > >

3 5

Kamailio & dbaliases
by Marino Mileti 19 Sep '14

19 Sep '14

Hi to all, is it possible to define same alias for differents SIP ID? How? Documentations of "Alias_DB Module" says: "If the alias resolves to many SIP IDs, the first is replacing the R-URI, the rest are added as branches" ..but i've tried to add more SIPID with the same alias but system doesn't allow it. A.

2 1

Multi NIC
by marino.mileti＠alice.it 19 Sep '14

19 Sep '14

I've installed Kamailio on multi NIC server (192.168.1.0 & 172.16.1.0), and I've configured rtpproxy in bridged mode in order to manage RTP but I've some trouble during call between endpoint placed on different network. I've find an example regarding Kamailio+RTP in IPv4+IPv6..it's a good point to start but i can't use "af" function... I've tried to replace it with $si but only some call works. How can i replace af correctly? Is there any example conf for my scenario? Many thanks

2 1

Crash Kamailio 4.1.5
by Igor Potjevlesch 19 Sep '14

19 Sep '14

Hello, A crash just occurred. I use the patch for the PAI issue. I had a look to the core dump and it looks to be another issue: (gdb) bt full #0 0x00000030f2230f30 in escape_string_for_mysql () from /usr/lib64/mysql/libmysqlclient.so.16 No symbol table info available. #1 0x00000030f22269c1 in mysql_real_escape_string () from /usr/lib64/mysql/libmysqlclient.so.16 No symbol table info available. #2 0x00007fb984c0b089 in db_mysql_val2str (_c=0x7fb985033d18, _v=0x7fb981921f60, _s=0x1c51bf0 "0669','<phone_number>','A.B.C.D')", _len=0x7fffab695f74) at km_val.c:79 l = 10 tmp = 1 old_s = 0x1c51bef "'0669','<phone_number>','A.B.C.D')" __FUNCTION__ = "db_mysql_val2str" #3 0x00007fb9847e1137 in db_print_values (_c=0x7fb985033d18, _b=0x1c51b0c "'INVITE','as689f6052','a94c095b773be1dd6e8d668a785a9c847afd0320','361aa4536 ba9cd8463b3ec8114d3711a(a)domain.tld','200','OK','2014-09-16 17:05:26','<phone_number>','<phone_number>@domain.tld','trunk."..., _l=65379, _v=0x7fb981921e00, _n=15, val2str=0x7fb984c0ac88 <db_mysql_val2str>) at db_ut.c:318 i = 11 l = 65152 len = 227 __FUNCTION__ = "db_print_values" #4 0x00007fb9847da028 in db_do_insert_cmd (_h=0x7fb985033d18, _k=0x7fb981921b20, _v=0x7fb981921e00, _n=15, val2str=0x7fb984c0ac88 <db_mysql_val2str>, submit_query=0x7fb984c02092 <db_mysql_submit_query>, mode=0) at db_query.c:224 off = 156 ret = 10 __FUNCTION__ = "db_do_insert_cmd" #5 0x00007fb9847da3b9 in db_do_insert (_h=0x7fb985033d18, _k=0x7fb981921b20, _v=0x7fb981921e00, _n=15, val2str=0x7fb984c0ac88 <db_mysql_val2str>, submit_query=0x7fb984c02092 <db_mysql_submit_query>) at db_query.c:249 No locals. #6 0x00007fb984c04419 in db_mysql_insert (_h=0x7fb985033d18, _k=0x7fb981921b20, _v=0x7fb981921e00, _n=15) at km_dbase.c:415 No locals. #7 0x00007fb98170b783 in acc_db_request (rq=0x7fb9797ab0c8) at acc.c:492 m = 15 n = 2038083784 i = 15 t = 0x414cc0 __FUNCTION__ = "acc_db_request" #8 0x00007fb981715bc8 in acc_onreply (t=0x7fb979775070, req=0x7fb9797ab0c8, reply=0x7fb98504fc48, code=200) at acc_logic.c:471 new_uri_bk = {s = 0x7fb97971e1af "sip:<phone_number>@domain.tld SIP/2.0\r\nRecord-Route: <sip:A.B.C.D;lr=on>\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK6a0a.6c2a27c22049161f515e9c9ac1ffe7e1.0\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK4"..., len = 19} br = 0 hdr = 0x7fb98171cc20 __FUNCTION__ = "acc_onreply" #9 0x00007fb98171630a in tmcb_func (t=0x7fb979775070, type=512, ps=0x7fffab6962e0) at acc_logic.c:573 __FUNCTION__ = "tmcb_func" #10 0x00007fb98391046c in run_trans_callbacks_internal (cb_lst=0x7fb9797750e0, type=512, trans=0x7fb979775070, params=0x7fffab6962e0) at t_hooks.c:290 cbp = 0x7fb979633b38 backup_from = 0x934630 backup_to = 0x934638 backup_dom_from = 0x934640 backup_dom_to = 0x934648 backup_uri_from = 0x934620 backup_uri_to = 0x934628 backup_xavps = 0x934760 __FUNCTION__ = "run_trans_callbacks_internal" #11 0x00007fb98391067e in run_trans_callbacks_with_buf (type=512, rbuf=0x7fb979775130, req=0x7fb9797ab0c8, repl=0x7fb98504fc48, flags=200) at t_hooks.c:336 params = {req = 0x7fb9797ab0c8, rpl = 0x7fb98504fc48, param = 0x7fb979633b48, code = 200, flags = 200, branch = 0, t_rbuf = 0x7fb979775130, dst = 0x7fb979775180, send_buf = { s = 0x7fb9796f4070 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK49e57236;rport=5060\r\nCall-ID: 361aa4536ba9cd8463b3ec8114d3711a(a)domain.tld\r\nFrom: \"<phone_number>\" <sip:<phone_number>@domain.tld>;t"..., len = 925}} trans = 0x7fb979775070 #12 0x00007fb983942bfa in relay_reply (t=0x7fb979775070, p_msg=0x7fb98504fc48, branch=0, msg_status=200, cancel_data=0x7fffab696640, do_put_on_wait=1) at t_reply.c:2001 relay = 0 save_clone = 0 buf = 0x7fb98505ddd0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK49e57236;rport=5060\r\nCall-ID: 361aa4536ba9cd8463b3ec8114d3711a(a)domain.tld\r\nFrom: \"<phone_number>\" <sip:<phone_number>@domain.tld>;t"... res_len = 925 relayed_code = 200 relayed_msg = 0x7fb98504fc48 reply_bak = 0x7fffab696490 bm = {to_tag_val = {s = 0x7fb979776608 "", len = 5449371}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7fb979775130 to_tag = 0x0 reason = {s = 0x1ab696440 <Address 0x1ab696440 out of bounds>, len = 1} onsend_params = {req = 0x200924a04, rpl = 0x7fb983962f90, param = 0x414cc0, code = 1, flags = 0, branch = 0, t_rbuf = 0x7fb96f3b4828, dst = 0x7fb98504fe40, send_buf = {s = 0xab696460 <Address 0xab696460 out of bounds>, len = 1024}} __FUNCTION__ = "relay_reply" #13 0x00007fb9839450ab in reply_received (p_msg=0x7fb98504fc48) at t_reply.c:2499 msg_status = 200 last_uac_status = 180 ack = 0x40 <Address 0x40 out of bounds> ack_len = 0 branch = 0 reply_status = -2063230624 onreply_route = 1 cancel_data = {cancel_bitmap = 0, reason = {cause = 200, u = {text = {s = 0x0, len = 9586191}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 9586191}}}} uac = 0x7fb9797751d8 t = 0x7fb979775070 lack_dst = {send_sock = 0x7fb9850202f0, to = {s = {sa_family = 17952, sa_data = "#\000\000\000\000\000`h\377\204\271\177\000"}, sin = {sin_family = 17952, sin_port = 35, sin_addr = {s_addr = 0}, sin_zero = "`h\377\204\271\177\000"}, sin6 = { sin6_family = 17952, sin6_port = 35, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = "`h\377\204\271\177\000\000\220h\377\204\271\177\000", __u6_addr16 = {26720, 34047, 32697, 0, 26768, 34047, 32697, 0}, __u6_addr32 = {2231330912, 32697, 2231330960, 32697}}}, sin6_scope_id = 2231330912}}, id = 32697, proto = 0 '\000', send_flags = {f = 39 '\'', blst_imask = 5 '\005'}} backup_user_from = 0x934630 backup_user_to = 0x934638 backup_domain_from = 0x934640 backup_domain_to = 0x934648 backup_uri_from = 0x934620 backup_uri_to = 0x934628 backup_xavps = 0x934760 replies_locked = 1 branch_ret = 0 prev_branch = -1419155712 blst_503_timeout = 32767 hf = 0x7fb98504fc68 onsend_params = {req = 0x7fffab6966c0, rpl = 0x550bb0, param = 0x2345e0, code = 0, flags = 3, branch = 0, t_rbuf = 0x7fb984e1d380, dst = 0x7fb9850526d0, send_buf = {s = 0x7fffab6966c0 "`G\223", len = 5538065}} ctx = {rec_lev = 0, run_flags = 0, last_retcode = 0, jmp_env = {{__jmpbuf = {140434777374272, -8626733706341394934, 4279488, 140736069201136, 0, 0, -8626733706320423414, 8626919071020996106}, __mask_was_saved = 0, __saved_mask = {__val = {9586419, 901952718779, 124554051613, 9586468, 140434777413984, 9587204, 9586197, 361695345073193192, 9586295, 9586274, 2231736624, 140434777413984, 140434777404704, 140434777374272, 4279488, 140736069201136}}}}} __FUNCTION__ = "reply_received" #14 0x000000000045d853 in do_forward_reply (msg=0x7fb98504fc48, mode=0) at forward.c:777 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} new_len = 32697 r = 1 s = 0x4048504fc50 <Address 0x4048504fc50 out of bounds> len = 0 __FUNCTION__ = "do_forward_reply" #15 0x000000000045e114 in forward_reply (msg=0x7fb98504fc48) at forward.c:860 No locals. #16 0x00000000004a5903 in receive_msg (buf=0x924600 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK6a0a.6c2a27c22049161f515e9c9ac1ffe7e1.0;received=A.B.C .D\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK49e57236;rport=5060\r\nCall-ID: 36"..., len=1028, rcv_info=0x7fffab6969c0) at receive.c:273 msg = 0x7fb98504fc48 ctx = {rec_lev = 8868984, run_flags = 0, last_retcode = 0, jmp_env = {{__jmpbuf = {0, 0, 0, 263853236176, 1, 0, 169733116696, 9586112}, __mask_was_saved = -1419155000, __saved_mask = {__val = {140434775328496, 12884901899, 140434775328496, 4279488, 140736069201136, 140736069200192, 5477982, 0, 140434411832704, 50195, 169369879936, 9586112, 140736069200320, 140736069200240, 5474817, 4279488}}}}} ret = 32697 inb = {s = 0x924600 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK6a0a.6c2a27c22049161f515e9c9ac1ffe7e1.0;received=A.B.C .D\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK49e57236;rport=5060\r\nCall-ID: 36"..., len = 1028} __FUNCTION__ = "receive_msg" #17 0x000000000053c9c4 in udp_rcv_loop () at udp_server.c:536 len = 1028 buf = "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK6a0a.6c2a27c22049161f515e9c9ac1ffe7e1.0;received=A.B.C .D\r\nVia: SIP/2.0/UDP 172.16.3.105:5060;branch=z9hG4bK49e57236;rport=5060\r\nCall-ID: 36"... tmp = 0x9245c0 "10.143.1.10" from = 0x7fb984f8cd70 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {336073913, 140434775328496}, addr32 = {336073913, 0, 2229651184, 32697}, addr16 = {5305, 5128, 0, 0, 50928, 34021, 32697, 0}, addr = "\271\024\b\024\000\000\000\000\360\306å ¹\177\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {67638457, 0}, addr32 = {67638457, 0, 0, 0}, addr16 = {5305, 1032, 0, 0, 0, 0, 0, 0}, addr = "\271\024\b\004", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = { sa_family = 2, sa_data = "\023Ä¹\024\b\024\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 336073913}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 336073913, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7fb984e5c588, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #18 0x000000000046d447 in main_loop () at main.c:1617 i = 12 pid = 0 si = 0x7fb984e5c588 si_desc = "udp receiver child=12 sock=A.B.C.D:5060\000\204\271\177\000\000\b\024î ¹\177\000\000\036\205^\000\000\000\000\000\000w^\000\000\000\000\000@K\237@\ 000\000\000\000\300LA\000\000\000\000\000\360li\253\377\177", '\000' <repeats 18 times>, "0ki\253\377\177\000\000\020\245K\000\000\000\000" nrprocs = 15 __FUNCTION__ = "main_loop" #19 0x000000000047054f in main (argc=7, argv=0x7fffab696cf8) at main.c:2545 cfg_stream = 0x1c42010 c = -1 r = 0 tmp = 0x7fffab697f70 "" tmp_len = 0 port = 0 proto = 0 options = 0x5e0a58 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1073556037 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x3d6f60fb88 p = 0x5caba0 "H\211l$\330L\211d$\340H\215-O\244*" __FUNCTION__ = "main" Regards, Igor.

2 4

Kamailio v4.1.6 Released
by Daniel-Constantin Mierla 18 Sep '14

18 Sep '14

Hello, Kamailio SIP Server v4.1.6 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.6. For more details about version 4.1.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/09/kamailio-v4-1-6-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Registration on SSP
by Alexander Chaliy 18 Sep '14

18 Sep '14

Hello, I' trying to register kamailio on SIP Service Provider. In REGISTER reguest I see that URI is like "sip:DOMAIN:PORT;transport=X", but my SSP wait in another schema "sip:DOMAIN". How can I solve it? Thanks. Regards, Alexander Chaliy mobile: +38 097 102 45 83 mail: a.chaliy84(a)gmail.com skype: achaliy

2 1

Kamailio - Separated Servers
by Bruno Emer 18 Sep '14

18 Sep '14

Hello! I am relatively new to Kamailio and I'm trying to create a new enviroment using it in my company. I am thinking about use Amazon to host the servers and use OpsWorks to automatically escalate then if necessary. To accomplish this, my idea is to separate the servers, using one dedicated server to run as WebSocket, one to run as proxy and one as a Registrar. I'll be using just one database to store informations to all of my servers. With this, if I need more resources later, I can just create new servers with the specific roles (WebSockets, Proxy, Registrar). By now, the idea is clear, but the point is that I don't know how to separate the WebSockets server from the proxy server. Actually, I can do this, but when I have one agent using a regular softphone and one agent using WebSockets (with JSSIP) they are not able to establish a session if the softphone user starts it. Now, I want to know if is there a way to use two websockets servers, register users using both of then and start sessions between then, with a separated proxy and registrar. Has anyone done this before? Is possible to use kamailio like this? Thanks. Bruno Emer

2 1

R: Re: R: Re: Group call
by marino.mileti＠alice.it 18 Sep '14

18 Sep '14

Hi, but with alias_db I can specify max 1 contact for a sip alias, while I need more sip contacts for the same alias. I've tried to add more sip contacts but systems says "Alias already in dbaliases table" :( ----Messaggio originale---- Da: abalashov(a)evaristesys.com Data: 18-set-2014 12.24 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] R: Re: Group call Hi, If you just want to generically map a URI to a bunch of other URIs, this might make the most sense: http://kamailio.org/docs/modules/4.1.x/modules/alias_db.html#idm2736 -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906 _______________________________________________ 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

1 0

No audio/video transmission over different networks
by Abhishek Saini 18 Sep '14

18 Sep '14

Hi, I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video calls seem to work well when both the devices are connected to the same network, however, when one device connects to a different network (the two devices now are on different networks), they are able to register on SIP server, and even call can be triggered and accepted between the two devices but there is no video/audio transmission. I have setup rtpproxy but i don't know whether it's working or not. Any help on this would be highly appreciated. Following is my kamailio.cfg file: #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT #!define WITH_USRLOCDB tcp_async = yes tcp_connect_timeout=5 #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - 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://root:root@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="xyz.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 listen=tcp:172.31.47.138:5060 advertise 54.191.193.***: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/" #!else mpath="/usr/local/lib64/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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:xyz.com") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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 } # 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] { # 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; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # 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 { if(is_first_hop()) set_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("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { 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|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 16

R: Re: Group call
by marino.mileti＠alice.it 18 Sep '14

18 Sep '14

Thanks Alex I've checked your suggestion but it seems i've to register all user with same contact (AOR)...for example all registered with 1000(a)domain.But I can't register all users with same contact because I've to allow "intercom" call inside group. Is there any other way? ----Messaggio originale---- Da: abalashov(a)evaristesys.com Data: 18-set-2014 12.01 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] Group call On 09/18/2014 05:59 AM, marino.mileti(a)alice.it wrote: > Hi guys, > > is possible to make a group call with kamailio? Example...i dial 1000 > and kamailio make a parallel forking to 1001,1002,1003? > > In case of yes...every group member (1001,1002,1003) will be resolved by > "location"? Because some member is natted so RTPPROXY should be engaged Yes; the degree to which this is done automatically be can controlled via the following parameters: http://kamailio.org/docs/modules/4.1.x/modules/registrar.html#idp77968 http://kamailio.org/docs/modules/4.1.x/modules/registrar.html#idp110040 If you have these set to support multiple parallel branches, lookup() + t_relay() should just work "automagically". -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906 _______________________________________________ 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

2 1

howto synchronize ruid across multiple proxy instances
by Andrew Pogrebennyk 18 Sep '14

18 Sep '14

Hi, let's say I'm running two proxies/registrars that need to access Shared location DB in db_mode=1 (all changes to usrloc are immediately reflected in database too). I have observed that if the UAC re-registers before the previous registration's expiry and the new REGISTER reaches the other proxy than the one that processed the registration originally, this new proxy is going to insert the second record for the same username into location table instead of updating the existing registration. I assume the problem is the ruid calculation, because proxy2 generated a different ruid than proxy1 since internal state of the two nodes is not shared. I have two questions: 1) is that assumption correct? 2) is there any way to synchronize the "ruid" of the two nodes in usrloc module, similar to the "secret" parameter in the auth module, or a better solution? I prefer the load-balancer to select random proxy and not do a hash based on From URI. I don't have the trace of the two registrations at the moment, but could get one next week. The issue popped up when running some sipp tests, but it's also possible that some cheap SIP endpoint would register well before the expiration time, creating duplicate registration. So if someone dials its number it's going to receive multiple incoming calls. Here are the module parameters I am using, all quite ordinary: modparam("usrloc", "use_domain", USE_DOMAIN) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_url", CENTRAL_URL) modparam("usrloc", "db_check_update", 0) modparam("usrloc", "nat_bflag", FLB_NATB) modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 43200) modparam("registrar", "method_filtering", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "max_contacts", 5) modparam("registrar", "received_avp", "$avp(s:received)") modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "gruu_enabled", 0) Any ideas? Thanks in advance. Andrew

3 3

Planning Kamailio v4.1.6
by Daniel-Constantin Mierla 18 Sep '14

18 Sep '14

Hello, I am considering releasing minor version 4.1.6 next Wednesday or Thursday, give there is a while from the latest patch release. Also, I will target to release v4.0.7 shortly after. As usual, report any issue not yet announced on sr-dev or bug tracker as well as send a reminder if it is any that didn't get any attentions lately. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 2

Print callee domain in logs
by aawaise 18 Sep '14

18 Sep '14

Hello, I have multidomain scenario with parameters (both domains (servers) using common database): modparam("usrloc", "use_domain", 1); and modparam("domain", "db_mode", 1). I want to know using domain and usrloc what kind of tweaking is possibe ?? Secondly, I wanto to print domain of callee in the logs on arrival of INVITE packet. How can it be done ? Regards, Awaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Print-callee-domain-in-logs-tp13059… Sent from the Users mailing list archive at Nabble.com.

3 5

NO_DTLS_FINGERPRINT when calling from webrtc to webrtc client
by Paweł Sternal 17 Sep '14

17 Sep '14

Hi. Another topic about WebRTC, websockets with kamailio and rtpengine ;-) My problem is how to distinguish a call to WS UAC and how to SIP UAC in scenarios: 1) WS client1 -> WS kamailio -> SIP kamailio -> SIP UAC 2) WS client1 -> WS kamailio -> SIP kamailio -> WS kamailio -> WS client2 WS kamailio is a proxy, SIP kamailio is a registrar When "WS client1" is calling to "123123" WS kamailio doesn't know if "123123" was registered from "WS client2" or from SIP UAC. I have in this case rtpengine_manage("....... RTP/AVP"), but when INVITE is returned to WS kamailio? RTP/SAVPF? Probably it is obvious, however... When WS client2 reply with 200OK, rtpengine_manage("..... ICE=force") to WS client1 SDP is sent without a:fingerprint. sipml5 dumps warning: message: "Could not negotiate answer SDP; cause = NO_DTLS_FINGERPRINT I tried different combinations... and I'm stuck :/ Regards

2 3

Re: [SR-Users] rtpengine upgrade fails
by Juha Heinanen 17 Sep '14

17 Sep '14

On 07/16/14 14:57:59, Richard Fuchs wrote: > Hard to tell without further information. Perhaps dist-upgrade tried to > upgrade the daemon after the kernel module was removed but before the > new module was installed? Did you install the kernel module through > dkms? Do you have the metapackage installed? You can check the dpkg.log > and dkms log files for more info. Richard, Somehow I missed your above reply in July. I have the metapackage installed. Now I made another dist-upgrade upgrading rtpengine from 3.3.0.0+0~mr3.4.1.0 to 3.3.0.0+0~mr3.5.0. Indeed dist-upgrade tries to setup daemon after the kernel module was removed but before new kernel module was installed. Should dependency on ngcp-rtpengine-kernel-dkms be added to ngcp-rtpengine-daemon in control file? -- Juha ------------------------------------------------------- # apt-get dist-upgrade ... Preparing to replace ngcp-rtpengine-daemon 3.3.0.0+0~mr3.4.1.0 (using .../ngcp-rtpengine-daemon_3.3.0.0+0~mr3.5.0.0_amd64.deb) ... Unpacking replacement ngcp-rtpengine-daemon ... Preparing to replace ngcp-rtpengine-iptables 3.3.0.0+0~mr3.4.1.0 (using .../ngcp-rtpengine-iptables_3.3.0.0+0~mr3.5.0.0_amd64.deb) ... Unpacking replacement ngcp-rtpengine-iptables ... Preparing to replace ngcp-rtpengine-kernel-dkms 3.3.0.0+0~mr3.4.1.0 (using .../ngcp-rtpengine-kernel-dkms_3.3.0.0+0~mr3.5.0.0_all.deb) ... Stopping RTP/media proxy: rtpengine-daemon. -------- Uninstall Beginning -------- Module: rtpengine Version: 3.3.0.0+0~mr3.4.1.0 Kernel: 3.2.0-4-amd64 (x86_64) ------------------------------------- Status: Before uninstall, this module version was ACTIVE on this kernel. xt_MEDIAPROXY.ko: - Uninstallation - Deleting from: /lib/modules/3.2.0-4-amd64/updates/dkms/ - Original module - No original module was found for this module on this kernel. - Use the dkms install command to reinstall any previous module version. depmod......... DKMS: uninstall completed. ------------------------------ Deleting module version: 3.3.0.0+0~mr3.4.1.0 completely from the DKMS tree. ------------------------------ Done. Unpacking replacement ngcp-rtpengine-kernel-dkms ... Preparing to replace ngcp-rtpengine 3.3.0.0+0~mr3.4.1.0 (using .../ngcp-rtpengine_3.3.0.0+0~mr3.5.0.0_all.deb) ... Unpacking replacement ngcp-rtpengine ... Preparing to replace ngcp-rtpengine-dev 3.3.0.0+0~mr3.3.1.0 (using .../ngcp-rtpengine-dev_3.3.0.0+0~mr3.5.0.0_all.deb) ... Unpacking replacement ngcp-rtpengine-dev ... Preparing to replace ngcp-rtpengine-kernel-source 3.3.0.0+0~mr3.4.1.0 (using .../ngcp-rtpengine-kernel-source_3.3.0.0+0~mr3.5.0.0_all.deb) ... Unpacking replacement ngcp-rtpengine-kernel-source ... Setting up ngcp-rtpengine-daemon (3.3.0.0+0~mr3.5.0.0) ... Restarting RTP/media proxy: ngcp-rtpengine-daemonFATAL: Module xt_MEDIAPROXY not found. iptables: No chain/target/match by that name. ip6tables: No chain/target/match by that name. FAILED TO CREATE KERNEL TABLE 0, KERNEL FORWARDING DISABLED invoke-rc.d: initscript ngcp-rtpengine-daemon, action "restart" failed. dpkg: error processing ngcp-rtpengine-daemon (--configure): subprocess installed post-installation script returned error exit status 255 Setting up ngcp-rtpengine-iptables (3.3.0.0+0~mr3.5.0.0) ... Setting up ngcp-rtpengine-kernel-dkms (3.3.0.0+0~mr3.5.0.0) ... Creating symlink /var/lib/dkms/rtpengine/3.3.0.0+0~mr3.5.0.0/source -> /usr/src/rtpengine-3.3.0.0+0~mr3.5.0.0 DKMS: add completed. Kernel preparation unnecessary for this kernel. Skipping... Building module: cleaning build area.... make KERNELRELEASE=3.2.0-4-amd64 -C /lib/modules/3.2.0-4-amd64/build M=/var/lib/dkms/rtpengine/3.3.0.0+0~mr3.5.0.0/build MEDIAPROXY_VERSION=3.3.0.0+0~mr3.5.0.0....... cleaning build area.... DKMS: build completed. xt_MEDIAPROXY: Running module version sanity check. - Original module - No original module exists within this kernel - Installation - Installing to /lib/modules/3.2.0-4-amd64/updates/dkms/ depmod.... DKMS: install completed. ...

2 3

kamctl alias add issue
by Igor Potjevlesch 17 Sep '14

17 Sep '14

Hello, Until the update to 4.1.5, when I send a command kamctl alias add like : kamctl alias add test@domain sip:test@domain, I got the following: 400 Too few or too many arguments Any idea of what can be the issue? Regards, Igor.

2 4

Voicemail & EarlyMedia
by marino.mileti＠alice.it 17 Sep '14

17 Sep '14

I would like to send a call to voicemail server after simple timeout. Called client replies with prov response (183 with SDP) but doesn't answer, so no 200OK message. Is it possible to route call to voicemail server? How? Many thanks

2 1

paid support plans
by Mike Hancock 17 Sep '14

17 Sep '14

Hello, We have recently installed the 4.0 Siremis/Kamailio 4.1 on Centos 6.5, to use as our SBC for our VoIP/hosted PBX. We struggled with the installation and aren't sure if it is in fact complete. We weren't able to find documentation that explained the correct setup and implementation of the server. I would like to purchase support to help to get this up and running as well as configured to get us going. We currently use FreePBX VMs for our customer with connections directly from our carriers to the servers. We would like to use this as a session border controller to existing deployments, another instance as the front end to the asterisk servers, and API information to connect to our portal to allow customers to maintain certain areas of their systems. Please contact me so we can get together to get this going as soon as possible. Thank you very much! Mike Hancock Managing Partner Bridge Communications, LLC 678-845-7600 Main 678-845-7616 Fax 678-845-7602 Direct [bridge-logo-email Signature] * VoIP, Hosted PBX, Key, & PBX Telephone Systems * Voice, Data, & Fiber Optic Cable Installation * Wireless Broadband Internet & Networking * Access Control, Audio & Video Systems

2 1

Help to build a Kamailio SBC
by [PRE s.r.l.] - Alex 17 Sep '14

17 Sep '14

Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private network. The only problem I got is that called client get the call as it was originated by one of the media server (the call contact is some like ext(a)media.server.ip), which is wrong in my opinion. Here it is the configuration, can you help me dealing with this? Thank you #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_DISPATCHER # # 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 “database stuff" #!define DBASTURL "database stuff" #!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=yes #!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:172.20.20.29:5060 listen=tcp:172.20.20.29:5060 listen=udp:PUBLIC IP:5060 listen=tcp:PUBLIC IP:5060 mhomed=1 /* 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 # asterisk.bindip = "172.20.20.80" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "172.20.20.29" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/x86_64-linux-gnu/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_DISPATCHER loadmodule "dispatcher.so" loadmodule "ipops.so" loadmodule "sqlops.so" #!endif loadmodule "uac.so" # ----------------- 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", 1) # ----- 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) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", “sip:pinger@PUBLIC IP") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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 #!ifdef WITH_DISPATCHER # ------- Load-balancer params ------ modparam("dispatcher", "db_url”,”MYSQL STUFF") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:kamailio-sbc@172.20.20.29") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") modparam("sqlops","sqlcon",”ca=>OTHER MYSQL STUFF") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); route(RELAY); } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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 ... nore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # 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; } # Authentication route route[AUTH] { #if(route(FROMASTERISK)) # return; # 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; } #!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("$fd", "sipusers", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # 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; } # 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(); route(REGFWD); exit; } } # USER location service route[LOCATION] { 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; } $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($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] { # 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; } # 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; route(RTPPROXY); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # 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; } } # RTPProxy control route[RTPPROXY] { if (is_method("INVITE")){ sql_query("ca", "select destination from dispatcher where destination like '%$dd%'","ra"); if($dbr(ra=>rows)>0){ $avp(duip)=$(du{s.select,-2,:}); if (is_ip_rfc1918("$avp(duip)")) { xlog("L_INFO", "Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("cwei","172.20.20.29"); #rtpproxy_manage("rwei"); } } else if(ds_is_from_list()){ if (is_ip_rfc1918("$si")) { xlog("L_INFO", " Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("cwie”,”PUBLIC IP"); #rtpproxy_manage("rwie"); } }else if(!ds_is_from_list()){ rtpproxy_manage("rwei"); xlog("L_INFO", "NONE! RTP PROXY BOH\n"); } } } # Send to Asterisk route[TOASTERISK] { ds_mark_dst("P"); if(!ds_select_dst("1", "4")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; } xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawie"); route(RELAY); exit; } # Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()){ rtpproxy_manage("cawei"); xlog("L_INFO","[$fU@$si:$sp]{$rm} Call from Media-Server Cluster\n"); return 1; } return -1; } route[REGFWD] { xlog("L_INFO","regfwd "+"sip:" +"\n"); 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(); } -- Alex PRE s.r.l. Backbone Network Mission Control System Engineering&Architecture Task Force alex(a)presrl.net http://www.presrl.it +39 0971 471 430 +39 388 150 6886 Questo messaggio e i suoi allegati sono indirizzati esclusivamente alle persone indicate. La diffusione, copia o qualsiasi altra azione derivante dalla conoscenza di queste informazioni sono rigorosamente vietate. Qualora abbiate ricevuto questo documento per errore siete cortesemente pregati di darne immediata comunicazione al mittente e di provvedere alla sua distruzione, Grazie. This e-mail and any attachments is confidential and may contain privileged information intended for the addressee(s) only. Dissemination, copying, printing or use by anybody else is unauthorised. If you are not the intended recipient, please delete this message and any attachments and advise the sender by return e-mail, Thanks. Rispetta l'ambiente. Non stampare questa mail se non è necessario.

2 3

Are there any good arguments (except political ones) for not using Kamailio within Docker?
by davy van de moere 17 Sep '14

17 Sep '14

Gents, I've been loving OpenVZ for a multitude of deployments in the last 24 months, but recently I stumbled over Docker, and I really like it! It's simplicity is very appealing. I was trying to find good reasons not to use Docker in production environments. So far I couldn't get any further than that it's pretty new, and that it is not having a great reputation in resource allocation. On a first check I think OpenVZ is also much more flexible in it's networking, but that might be my lack of understanding of Docker. Typically I have my setups mixed with Postgres/Mysql and Freeswitch. As such the question, any good reason not to use Docker? thx! Davy Van De Moere

2 1

Kamailio reading an external configuration file from the main config.
by parcerito12＠gmail.com 17 Sep '14

17 Sep '14

Is there any way that i can create a list of ips for kamailio to compare and do something, for example create an authorize list of ips on txt that it can read for each invite that i receive than check a variable against that list. for example do a search $si on the external text file and if match do something if it doesn't then continue the route downwards. I am trying to create a list of banned ips but don't want to do it on the firewall since i want to log all the bad ips and added to the list and do something every time i receive a packed from any of the ips on the list. is that possible with kamailio , is there any module i can use to cook this configuration ? Thank you Julian

2 1

Re: [SR-Users] [sr-dev] Reqd. help on Corex (Obfuscate) - Kamailio 4.2.x
by Rahul MathuR 17 Sep '14

17 Sep '14

Thanks for replying ! But how to check whether a particular message received by Kamailio was sent by UAC or SIP Server ? Also, on the same lines - how to know whether a particular message about to be send from Kamailio is bound to UAC or SIP Server ? On Tue, Sep 16, 2014 at 10:51 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com> wrote: > Hi, > > The network io intercept feature basically allows kamailio script writer > to do whatever s/he may want to do with raw SIP packets (that are just > received by kamailio or about to be sent out by kamailio), e.g. encryption, > compression or any final touches to sip message before it is processed by > kamailio core. That is why it is purposely kept abstract and any particular > use or implementation is left to the script writer. > > In your case the encryption / decryption code is in C/C++, you can try one > of the followings, > > 1. Writeup a C/C++ program that receives outgoing SIP message as text (and > some other parameters, e.g. encryption key) in input arguments and returns > the encrypted message in event_route [ network:msg ] and vice versa (for > incoming messages). You can call this program directly from kamailio.cfg > script. > > 2. Writeup e.g. a PERL wrapper for your encryption / decryption C/C++ code > and call it using kamailio app_perl module within event_route [ network:msg > ] as demonstrated in this example, > > http://kamailio.org/docs/modules/devel/modules/corex.html#idp125704 > > You can also use any other kamailio language bind of you choice as well, > e.g. Python, LUA, JAVA and so on. > > I would recommend the second option, as it has less processing overhead > for kamailio. > > Thank you. > > > > On Tue, Sep 16, 2014 at 6:09 PM, Rahul MathuR <rahul.ultimate(a)gmail.com> > wrote: > >> Hello, >> >> I was going through the new features and stumbled upon this new one - >> developed by Mohd. Shahzad Shafi. >> As already mentioned on the wiki about this module, I intend to use it >> for my custom security layer between UACs and SIP Proxy (Kamailio) but the >> issue is - the custom security layer (encryption/decryption code) is >> written in C and should precisely be applied for the messages between UAC >> and Proxy. >> >> Is there a way I can achieve this using Corex module since it does >> intercept the network I/O messages ? >> >> Any help here would be really appreciated. >> >> -- >> Warm Regds. >> MathuRahul >> >> _______________________________________________ >> sr-dev mailing list >> sr-dev(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >> >> > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev > > -- Warm Regds. MathuRahul

2 2

Fwd: Reqd. help on Corex (Obfuscate) - Kamailio 4.2.x
by Rahul MathuR 16 Sep '14

16 Sep '14

Hello, I was going through the new features and stumbled upon this new one - developed by Mohd. Shahzad Shafi. As already mentioned on the wiki about this module, I intend to use it for my custom security layer between UACs and SIP Proxy (Kamailio) but the issue is - the custom security layer (encryption/decryption code) is written in C and should precisely be applied for the messages between UAC and Proxy. Is there a way I can achieve this using Corex module since it does intercept the network I/O messages ? Any help here would be really appreciated. -- Warm Regds. MathuRahul

2 1

routing SIP REGISTER with app_perl
by Vik Killa 16 Sep '14

16 Sep '14

Hello, I'm learning Kamailio. My ultimate goal is to be able to route SIP messages from User Agents to specific FS boxes. I'm starting with routing REGISTER messages. I used app_perl to randomly lookup one of the FS box's IP Address from postgres database, then I stored the IP in avp and then in hash. It seemed like overkill but it was the only way I could keep the FS IP address in memory between the first REGISTER and REGISTER with Auth. my app_perl script sets the IP as follows: my $sth = $dbh->prepare("SELECT ip_address FROM host_machine WHERE is_online = true ORDER BY random() LIMIT 1"); $sth->execute(); my $host_ip_address = $sth->fetchrow(); Kamailio::AVP::add(10, "sip:$host_ip_address:5060"); And here is my kamailio.cfg for REGISTERs route[REGISTRAR] { if (!is_method("REGISTER|PUBLISH")) return; if (is_present_hf("Authorization")) { add_path_received(); $avp(i:10) = $sht(a=>$ci::dest_server); avp_pushto("$du", "$avp(i:10)"); } else { perl_exec("random_server"); avp_pushto("$du", "$avp(i:10)"); $sht(a=>$ci::dest_server) = $avp(i:10); } t_relay(); exit; } "random_server" is the perl script to get the FS IP address... Seems crude but works... I'm sure there is probably a more elegant way to do what im trying so any advice and criticism is appreciated, thank you! /V

2 3

mimic ds_select_dst with app_perl
by Vik Killa 16 Sep '14

16 Sep '14

I like dispatcher but i need something more custom to my setup so I want to attempt to emulate some basic behavior of dispatcher. I have a simple perl script kamailio calls, it does a database query to get an IP address and port similar to an item listed into the dispatcher list. my question is how do I forward the SIP packet to the destination using the database result query? Here is what i have so far but it doesn't work because it's modifying the request uri, where i just want to forward it to the FS server: my $sth = $dbh->prepare("SELECT ip_address FROM host_machine WHERE is_online = true ORDER BY random() LIMIT 1"); $sth->execute(); my $host_ip_address = $sth->fetchrow(); Kamailio::log(L_ERR, "Random server: " . $host_ip_address); $m->rewrite_ruri('sip:' . $m->pseudoVar('$rU') . '@' . $host_ip_address . ':5060'); $sth->finish; Thanks! /V

2 1

Difference between modules_k and modules_s
by aawaise 16 Sep '14

16 Sep '14

In the source code of kamailio, I found some modules code in modules_k folder. Code of same modules is also given modules_s. Examples of such modules are registrar module and usrloc module. If we want to add somethings like log statements etc. Where the change is supposed to be done ? Thanks. Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Difference-between-modules-k-and-mo… Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] Dispatcher - failure_route to specific node
by Nicholas Gill 16 Sep '14

16 Sep '14

On 14/09/14 20:00, sr-users-request(a)lists.sip-router.org wrote: > In your script snippets it seems > that upon receiving the 302, in the failure route you set the $du and then > you pass the message to a RELAY route, which I suppose sends the 407 reply. This seems to be the case, yes. The 407 comes from kamailio on the already authenticated call. I presume this is done to trigger the caller to send another INVITE. > If you don't need for special reasons that the user performs a proxy > authentication, in the failure route you could just relay the INVITE to the > target FreeSWITCH after setting $du to the value of your custom header. This is exactly what I'm hoping to do, but I'm not sure of the correct configuration to achieve this. It was my intention to relay the INVITE to the target server by setting the $du and then trigger the RELAY route. I have also tried setting $du and then calling t_relay() directly, but I suspect I'm missing something minor but important. Cheers, -nick

2 4

How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?
by Alex Villacís Lasso 15 Sep '14

15 Sep '14

I have a rtpproxy configuration that spawns several rtpproxy instances, using bridge mode. An example is shown below: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 10000 -M 20000 Here, rtpproxy bridges between 192.168.2.18 and 127.0.0.1 . Now I want to migrate to rtpengine with the rtpproxy-ng module in kamailio. However, I do not find an equivalent to bridge mode in the rtpengine command-line parameters. I see the --ip=IP parameter, but the source code expects a single IP address, and cannot be specified more than once. The closest I see is the --advertised-ip=IP parameter, but I am not sure that it will do what I need.

6 11

app_perl persistent database connection
by Vik Killa 15 Sep '14

15 Sep '14

hello, does kamailio start/run the perl script defined by modparam("app_perl", "filename", "/usr/local/test/bin/sips.pl") at startup and constantly run in background? I want a persistent database connection rather than having to constantly re-connect to database on each transaction. Is this possible? I'm connecting to postgres with use DBI; my $dsn = "DBI:Pg:database=$database;host=$host;port=$port"; my $dbh = DBI->connect($dsn, $user, $pass) || die "Could not connect to database: $DBI::errstr";

2 3

Foward REGISTER to all dispatcher members
by Kees Meijs - SIGNET B.V. 15 Sep '14

15 Sep '14

Dear list members, Currently I'm using the dispatcher module to distribute traffic towards media servers (being Asterisk, in my case). I'd like to forward REGISTER requests to all (active) dispatcher members, e.g. using a short and simple while loop. Could someone please give me some pointers towards the right direction? Is it for example possible to use the MI di_list functionality for iterating over all the members? Thanks in advance! Best regards, Kees

2 1

Extract domain of a user from location list table
by aawaise 15 Sep '14

15 Sep '14

I went through usrloc page on kamailio site. Link given below. http://www.kamailio.org/docs/modules/3.1.x/modules_s/usrloc.html It states that the module keeps a user location table and provides access to the table to other modules. The module exports no functions that could be used directly from scripts. How can one extract value of domain from Location List ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Extract-domain-of-a-user-from-locat… Sent from the Users mailing list archive at Nabble.com.

2 1

Dispatcher - failure_route to specific node
by Nicholas Gill 13 Sep '14

13 Sep '14

Hello sr-users, We're using Kamailio to proxy to a number of FreeSWITCH backend servers with the dispatcher module. In some specific scenarios it matters to which node the request is sent. In these cases we've configured FS to respond to the invite with 302 and provide a custom header which contains the uri of the needed backend server (Intending to emulate the behaviour of 305 Use Proxy). To attempt to handle this I've tried the following in kamailio: onreply_route[MANAGE_REPLY] { ... if (t_check_status("302") and is_present_hf("X-imx-redirect")) { $avp(imx-redirect) = $hdr(X-imx-redirect); xlog("SCRIPT: REDIRECT, Store avp for redirect: $avp(imx-redirect)\n"); } } Then in the dispatch failure route: if ($avp(imx-redirect) != $null) { $du = $avp(imx-redirect); xlog("SCRIPT: REDIRECT: going to <$ru> via <$du>\n"); route(RELAY); exit; } Which results in the following sip event sequence: // Normal invite sequence FS1 -> Kamailio SIP/2.0 302 Moved Temporarily (X-imx-redirect = sip:FS2...) Kamailio -> FS1 ACK Kamailio -> Caller SIP/2.0 407 Proxy Authentication Required Then the caller generates a new INVITE however I'm unsure how to route this INVITE to the specific node. The avp appears (intentionally) tied to the transaction so doesn't seem to be able to be used to correlate in this scenario. Advice on how to redirect from a backend server to another specific backend server behind Kamailio would be greatly appreciated. Cheers, -nick

2 1

Need help in how to read INVITE packet SDP c line IP Address
by varun pratapsingh 12 Sep '14

12 Sep '14

Hi All, Can anybody help me by telling how to read INVITE packet SDP c line IP address in Kamailio.cfg. For example a INVITE packet comes and its SDP contains the line v=0 o=- 3619547086 3619547086 IN IP4 1.223.65.215 s=pjmedia c=IN IP4 1.223.65.215 t=0 0 a=X-nat:0 Now I want to read this SDP packet c line IP address and want to put the IP 1.223.65.215 in a variable. Can anybody please help me out in this. Regards Varun

2 1

New..... / kickoff point to learn Kamailio
by Poekel 12 Sep '14

12 Sep '14

Hello, I am totally new/blank at kamailio and would like learn more but do not know where to start........ I would like to build a voicemail platform with kamailio and Asterisk. ( if i CAN use kamailio for this purpose) Someone creates a mailbox in Asterisk realtime for example through a website. I want to manage the voicemail boxes from one place (kamailio?) and route sip traffic to the voicemail boxes? Can i do that with kamailio ? How? can someone give me some advice on how and where to start? Any help appreciated! Poekel

2 1

received and rport via parameters
by Daniel-Constantin Mierla 12 Sep '14

12 Sep '14

Hello, opening this for discussion to get more eyes on the IETF specs to be sure I haven't missed any requirements. We are no adding received parameter to Via header as required by RFC 3261: 18.2.1 Receiving Requests When the server transport receives a request over any transport, it MUST examine the value of the "sent-by" parameter in the top Via header field value. If the host portion of the "sent-by" parameter contains a domain name, or if it contains an IP address that differs from the packet source address, the server MUST add a "received" parameter to that Via header field value. This parameter MUST contain the source address from which the packet was received. This is to assist the server transport layer in sending the response, since it must be sent to the source IP address from which the request came. But then RFC3581 says: 4. Server Behavior The server behavior specified here affects the transport processing defined in Section 18.2 of SIP [1]. When a server compliant to this specification (which can be a proxy or UAS) receives a request, it examines the topmost Via header field value. If this Via header field value contains an "rport" parameter with no value, it MUST set the value of the parameter to the source port of the request. This is analogous to the way in which a server will insert the "received" parameter into the topmost Via header field value. In fact, the server MUST insert a "received" parameter containing the source IP address that the request came from, even if it is identical to the value of the "sent-by" component. Note that this processing takes place independent of the transport protocol. I understand that if rport parameter is present in incoming Via, received has to be added always, even when matching the IP address in via. RFC3261 required to be added only when it source IP was different than the address in Via. On the other hand, the RFC3261 says received is needed for helping the server to route back the reply, so received shouldn't make much sense for upstream. However, apparently, some clients use it (or at least check it and don't work properly if not present after a rport -- one specific example is that some phone is not making calls after sending REGISTER with rport and not receiving received). RFC3581 says that server behavior affect what was specified in 3261, so my plan is to send received always if rport is present, but I wanted to see if anyone sees issues with that or there is another more recent rfc amending further the behaviour related to rport. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

2 2

Dispatcher - failure_route to specific node
by Nicholas Gill 12 Sep '14

12 Sep '14

Hello sr-users, We're evaluating Kamailio to proxy to a number of FreeSWITCH backend servers with the dispatcher module. In some specific scenarios it matters to which node the request is sent. In these cases we've configured FS to respond to the invite with 302 and provide a custom header which contains the uri of the needed backend server (Intending to emulate the behaviour of 305 Use Proxy). To attempt to handle this I've tried the following in kamailio: onreply_route[MANAGE_REPLY] { ... if (t_check_status("302") and is_present_hf("X-imx-redirect")) { $avp(imx-redirect) = $hdr(X-imx-redirect); xlog("SCRIPT: REDIRECT, Store avp for redirect: $avp(imx-redirect)\n"); } } Then in the dispatch failure route: if ($avp(imx-redirect) != $null) { $du = $avp(imx-redirect); xlog("SCRIPT: REDIRECT: going to <$ru> via <$du>\n"); route(RELAY); exit; } Which results in the following sip event sequence: // Normal invite sequence FS1 -> Kamailio SIP/2.0 302 Moved Temporarily (X-imx-redirect = sip:FS2...) Kamailio -> FS1 ACK Kamailio -> Caller SIP/2.0 407 Proxy Authentication Required Then the caller generates a new INVITE however I'm unsure how to route this INVITE to the specific node. The avp appears (intentionally) tied to the transaction so doesn't seem to be able to be used to correlate in this scenario. Advice on how to redirect from a backend server to another specific backend server behind Kamailio would be greatly appreciated. Cheers, -nick

1 0

send_reply in app_java or app_mono?
by Tom Johnson 11 Sep '14

11 Sep '14

Is there a way to call the send_reply method of the route from either Java or csharp? As I have been having issues with the app_java, I have been looking at app_mono and it seems to have better access to the variables, etc. However, the SR.core.ModF only seems to take methods that do not take parameters. Or for that matter, is there a way to return something back to the route in the config file which it can use to call send_reply? I have a strange requirement to return 202 rather than 200 from subscription requests and a 603 if it fails. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

3 2

app_java issue
by Tom Johnson 11 Sep '14

11 Sep '14

Per my previous message, I determined that the module's makefile was not properly finding the jdk/jre directories needed to compile. So, I temporarily hardcoded the paths in the makefile. However, now when I use the module, Kamailio crashes. Below is the lines from /var/log/messages: Sep 9 15:13:10 MAIFREEPBX kamailio: INFO: <core> [tcp_main.c:4836]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) Sep 9 15:13:10 MAIFREEPBX kamailio: WARNING: <core> [daemonize.c:352]: daemonize(): pid file contains old pid, replacing pid Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512 Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:125]: mod_init(): Initializing Java VM with options: -Xdebug -verbose:gc,class,jni -Djava.compiler=NONE -Djava.class.path=/opt/microautomation/i3/I3Notifier.jar:/opt/kamailio/java/kamailio.jar:/opt/microautomation/i3/lib/log4j-1.2.16.jar Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:146]: mod_init(): app_java: Java VM initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: INFO: app_java [java_mod.c:198]: mod_init(): app_java: module initialization OK Sep 9 15:13:10 MAIFREEPBX /usr/local/sbin/kamailio[12268]: : <core> [mem/q_malloc.c:159]: qm_debug_frag(): BUG: qm_*: prev. fragm. tail overwritten(0, abcdefed)[0x7f0941207190:0x7f09412071c0]! Sep 9 15:13:11 MAIFREEPBX abrt[12282]: Saved core dump of pid 12268 (/usr/local/sbin/kamailio) to /var/spool/abrt/ccpp-2014-09-09-15:13:10-12268 (183721984 bytes) The code dump mentioned does not exist. If I do not load the app_java module, Kamailio works. Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

3 7

websocket:ws_current_connections vs. kamctl fifo ws.dump
by Luis F Urrea 11 Sep '14

11 Sep '14

Hello I have a question in regards monitoring websocket connections on a production box. We ususally trust whatever is indicated through the $ kamctl fifo get statistics We've seen that this is what the community contributed munin plugins use. However when trying to monitor web socket connections and using this output we've seen completely different results from the output of $ kamctl fifo get_statistics websocket:ws_current_connections Which seem to indicate current websocket connections and the output of $ kamctl fifo ws.dump Which seems to indicate current OPEN connections. Can anybody shed some light as to what are the differences between both if any, and advice which one is the proper one to monitor for possible file handle exhaustion. TIA

3 3

Auto-response
by Mick Stevens 11 Sep '14

11 Sep '14

1 0

Re: [SR-Users] TLS Handshake failing with WSS
by Manuel Camarg 11 Sep '14

11 Sep '14

According to this article I mentioned in the beginning: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket In http:request they do straight away the set_reply_close(); I don't know exactly if this kamailio.cfg implementation may be useful for working with WSS: event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } HTTP authentication is not being properly set up. Any indications on how to perform this sucessfully? Kind regards Manuel >Hello, > >from the logs you sent now, it appears that you have set_reply_close() >in config, therefore the connection is closed after sending the reply. > >Cheers, >Daniel

4 14

How to increment dialog calee_cseq value when using t_uac_dlg via xmlrpc for generating INFO requests
by Drazen Rep 11 Sep '14

11 Sep '14

Hi, I am using Kamailio 4.1.1 and dlg_manage() function for processing SIP requests with dialog module. After call is established Kamailio is being used for generating in-dialog INFO request toward caller (using t_uac_dlg function via xmlrpc). INFO messages are correctly generated but dialog calee_cseq value is not being incremented with every message. That represents a problem since other SIP requests generated by callee side (like OPTIONS) are not in order, and there is no correct information about calee_cseq value in dialog memory records. The question is how can I increment dialog calee_cseq value when local INFO request toward caller is being generated using t_uac_dlg via xmlrpc? Thx, Drazen

2 4

UAC module
by Yuriy Gorlichenko 11 Sep '14

11 Sep '14

Hello. I tried to use UAC module at my system for registration multiple providers, and call through it. When I implementing this at my cfg file I had some troubles: -- with Authentification at INVITE (using UAC_AUTH). IT does not send second INVITE packet with www_auth header for some trunks (not for all, only for some providers) -- with BYE recieving from PROVIDER ("No BYE from Called party" task at USers Mailing lists) After many days of tests and debugging I ask question from myself - does module correctly works? As I know this module breaks some philosophy about SIP proxy and development of this module is not question of first priority. After that all I need advice - what I must use for registtration my SIP trunks - 3d party registrant or this module, that works fine and I doing something wrong? Thanks for advice.

2 1

Kamailio v4.2 - development frozen
by Daniel-Constantin Mierla 11 Sep '14

11 Sep '14

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 4.2.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 4.2 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to rich a stable state before releasing 4.2.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Help with updating the wiki page for migration from 4.1 to 4.2 as well as what is new in 4.2 is very appreciated. I will post updates about the very soon. Many thanks to everyone involved in development of 4.2 and the early testers that played with master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

LCR base on call rate
by Satish Patel 11 Sep '14

11 Sep '14

I heard somewhere LCR can do routing based on call rate and call price, does it true? I haven't seen any config or doc which does call rate using LCR. It only does routing base on prefix scan. Am I missing something here? Sent from my iPhone

2 3

WebRTC meeting Norfolk, 15 October 2014
by Daniel Pocock 10 Sep '14

10 Sep '14

I'll be in Norfolk, VA for xTupleCon in October On 15 October, there will be two events for WebRTC: 14:15 a talk about the xTuple WebRTC extension at xTupleCon - must register for xTupleCon to attend this 17:30 a technical / developer workshop at xTuple's offices - free, anybody welcome, even if not attending xTupleCon, RSVP through Eventbrite[1] Please see my blog[2] for more comments about all of this and feel free to email me in advance if you have questions about it or if you may like to meet up there. 1. http://www.eventbrite.com/e/browser-based-webrtc-telephony-for-web-apps-wor… 2. http://danielpocock.com/xtuplecon-webrtc-talk-schedule-change

1 0

TCP SYN only
by Laurent Schweizer 10 Sep '14

10 Sep '14

Dear all, Since few weeks I have more and more TCP issue, I can see that sometimes the client send a TCP SYN but never get any response. 106.123315 62.12.179.104 -> X.X.X.X TCP 39174 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942209 TSER=0 WS=1 109.124384 62.12.179.104 -> X.X.X.X TCP 39174 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942510 TSER=0 WS=1 111.202946 62.12.179.104 -> X.X.X.X TCP 4993 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294942717 TSER=0 WS=1 114.204707 62.12.179.104 -> X.X.X.X TCP 4993 > sip [SYN] Seq=0 Win=4380 Len=0 MSS=1380 TSV=4294943018 TSER=0 WS=1 I have found on internet many doc regarding TCP and max connection issue and I have already done change in sysctl (see below) to increase the number of tcp connection but seems still not ok. Any idea ? Best regards Laurent net.ipv4.ip_forward = 0 net.ipv4.conf.default.rp_filter = 1 net.ipv4.conf.default.accept_source_route = 0 kernel.sysrq = 0 kernel.core_uses_pid = 1 net.ipv4.tcp_syncookies = 0 net.bridge.bridge-nf-call-ip6tables = 0 net.bridge.bridge-nf-call-iptables = 0 net.bridge.bridge-nf-call-arptables = 0 kernel.msgmnb = 65536 kernel.msgmax = 65536 kernel.shmmax = 68719476736 kernel.shmall = 4294967296 net.ipv4.tcp_timestamps = 1 net.ipv4.tcp_fin_timeout = 30 net.ipv4.tcp_keepalive_time = 1200 net.ipv4.tcp_max_syn_backlog = 8192 net.ipv4.tcp_max_tw_buckets = 5000 fs.file-max = 2097152 net.ipv4.ip_local_port_range = 2000 65535 net.ipv4.tcp_max_tw_buckets = 1440000 net.ipv4.tcp_tw_recycle = 1 net.ipv4.tcp_tw_reuse = 1 net.core.somaxconn = 65536 net.core.netdev_max_backlog = 65536

3 11

Kamailio Installaton compatibility with ARM9 Architecture
by Nandini madhu 10 Sep '14

10 Sep '14

Dear all, Greetings. We need to install Kamailio server in ARM9 architecture based system (Example: Xilinx XC7Z045-2FFG676I FPGA board). Pleasse suggest me that, is there any possibility of installing Kamailio in ARM9 architecture based systems. Please respond Regards sermj2012

2 1

New developer: Federico Cabiddu
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that Federico Cabiddu has now write access to Kamailio git repository. He is going to push a new module named tsilo, which facilitates adding new branches to active transactions -- the typical case implemented so far is to track invites per callee and add a new branch if the callee registers while invite transaction is active, quite useful for services with mobile clients that are getting push notifications to wake up. Federico is one of the guys using Kamailio for many years, currently working at Orange, France. His commit user id is: fcabiddu My warm welcome and thanks for joining the effort to develop Kamailio! Looking forward to future contribution to the project, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

new module db_kazoo
by Luis Azedo 10 Sep '14

10 Sep '14

Hello Awesome Kamailio Community, We are writing on behalf of 2600hz, where we have been using Kamailio for some time now and are very pleased with our results! So, seems time to commit something back, dontchya think? We would like to present to you a new module, to hopefully be included in master. We call it db_kazoo (although a new name is fine too). db_kazoo is a general purpose AMQP connector (connects to our rabbitmq-server). It exposes publish/consume capabilities into Kamailio. Why is this amazing, you ask? Well even if you didn’t ask, we will explain… >From a high-level, the purpose of the module might be for things like: - Integrate to an AMQP application to make real-time routing decisions (instead of using, say, a SQL database) - Provide a real-time integration into your program, instead of your database, so you can overlay additional logic in your preferred language while also utilizing a message bus - Utilize messaging to have a distributed messaging layer, such that machines processing requests/responses/events can go up/down or share the workload and your Kamailio node will still be happy With this module, someone can: 1 - publish json payloads to rabbitmq 2 - publish json payloads to rabbitmq and wait for correlated response message 3 - subscribe to an exchange with a routing key The module works with a main forked process that does the communication with rabbitmq for issuing publishes, waiting for replies and consuming messages. When it consumes a message it defers the process to a worker process so that it doesn't block this main process. The worker process issues an event-route where we can act on the received payload. The name of the event-route is composed by values extracted from the payload. Consumed messages have the option of being acknowledge in two ways: 1 - immediately when received 2 - after processing by the worker One unique feature of our implementation revolves around failover of the message bus itself. In our design, the module supports multiple RabbitMQ servers and will fallback from one to the next in a list of RabbitMQ servers if the connection fails to the current connected server. When using acknowledge in db_kazoo with clustering in RabbitMQ, we have simulated and experienced full reconnects while in the middle of processing pending messages. In this way, when we get disconnected from one server we proved that, even at high speeds, we are able to connect to the next in the list and continue interacting with our application. We have run a variety of sipp load tests on this module and believe it is ready for prime time. We monitored memory, response accuracy and overall stability and it seemed OK. But we would, of course, love for others to help us find what we have missed, or contribute more features, or overall just use the work we slaved over for so many hours. Or just make comments and suggestions! If nothing else, we hope we’ve provided something useful to the Kamailio community, as you all have provided useful items to us. Thanks to everyone for all the work on Kamailio and related products. Look forward to hearing from you all. This is our first formal / large contribution to Kamailio so if we’ve done something wrong process, code or otherwise please let us know!

3 5

Location of Encryption Algorithm in TLS Module
by aawaise 10 Sep '14

10 Sep '14

Hello, I have downloaded and unzipped kamailio package available online. I want to know about TLS module, My question is that where is the encryption module located in TLS module ?? Is it in some .c file of TLS module or does TLS module use AES encryption available in LINUX or UNIX systems?? This question arises because I want to customize AES algorithm ?? Can it be done in TLS module or not ?? Regards, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Location-of-Encryption-Algorithm-in… Sent from the Users mailing list archive at Nabble.com.

2 1

Freezing development for v4.2 at the end of today
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, the development of new features for next major release, v4.2.0, is going to be frozen by end of today. As we have developers all over the world, we are going to allow everyone to benefit of the full day, so technically is like end of Wednesday, September 10, in Hawaii ( :-) a good fit, because it is the home region for the name of the project. If there is a patch with new features that you want to push, but needs review, we are going to take care of them in the next days, but upload them to the tracker or send them to sr-dev mailing list by today. Then follows about one month of testing and bug fixing for the existing code, with the target to have the 4.2.0 release out around mid of October. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

New developer: Luis Azedo
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that a new developer has now write access to Kamailo git repository: Luis Azedo. His commit username is: lazedo He is going to push soon the 'kazoo' module that was discussed these days on the mailing list - a module connector for the Kazoo - the open source, distributed VoIP platform using Kamailio for core SIP routing. He also provided many patches to other components so far. Luis works for 2600hz, an USA company, and will be accompanied by Darren Schreiber and Karl Andreson in maintenance of the new module. My warm welcome and thanks for joining the effort of developing Kamailio further! Looking forward to future contributions to the project! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

No BYE from Called party
by Yuriy Gorlichenko 09 Sep '14

09 Sep '14

Hello All. I have kamailio with provider connection (trunk) When I call to external number through my provider call extablished Ok. But when i try hangup call from external number no BYE sended to me. When I hangup call from my kamailio (internal num) I send by to exteral number and it respond me Ok so session if fully complete. I guess that BYE from external number not recieves to me because I have wrong routing header fields at my INVITe or ACK messages, but can not find any information what what header must recieve info to external number where send BYE at hangup or thomething like this. This is my little dump for situation wherer I hangup from internal number and BYE finished successfully: IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599 E....< .@.'. ...6........G.RINVITE sip:12345678900@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Max-Forwards: 70 From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068> Contact:<TrunkNum@my.kamailio.com:5068> Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 04 Sep 2014 21:53:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com", nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri=" sip:12345678900@my.provider.com:5060", qop=auth, nc=00000001, cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5 v=0 o=root 1022912010 1022912010 IN IP4 my.kamailio.com s=Asterisk PBX 12.5.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30032 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30033 a=ice-ufrag:3o8JrqkF a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2 a=candidate:TgT1dfTnI3kBgWQ IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882 E..... ./...6... ........b..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE Contact: <sip:12345678900@externail.number.end.ip:5060;transport=udp> User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 746 X-provider agentOutboundGateway: sip:5574012345678900@62.93.147.149 X-provider agentOutboundCarrierID: 23705946361020 X-provider agentCarrierRate: 0.20180 X-provider agentCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip s=FreeSWITCH c=IN IP4 externail.number.end.ip t=0 0 a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP m=audio 23216 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:2990874569 cname:pRs5xP IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596 E..p.@..@.K\ ...6........\`.ACK sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617 E....D..@.KC ...6........q.ZBYE sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: Asterisk PBX 12.5.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568 E..T....-.676... ........@..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Thanks for help.

2 4

Using app_java
by Tom Johnson 09 Sep '14

09 Sep '14

Any tips on compiling the app_java module? I am on CentOS 6. And I have libgcj-4.4.7 installed, but the build process complains about missing jni.h and other files.

2 1

Creating array out of hashtable
by aft 09 Sep '14

09 Sep '14

[1]Is it possible to iterate a htable without using "keys" like an array? [2] Is it possible to search for "keys" using regxp? -- -Cheers -Arif

2 2

Call connection from one server to another
by aawaise 09 Sep '14

09 Sep '14

Hello, I want to deploy two kamailio servers, both using same MYSQL database placed on Server 2008. Both are using different static IP's. Hopefully, I can add two A records in DNS server, both against same domain name, pointing to two different IP's to which two servers are attached. My question is that in load balancing scenario if user 1 is attached to server 1, and user 2 is attached to server 2. How can user1 call user2. Because request will go to server 1 to which user 1 is attached. But it wont be able to locate user 2 because it is not connected to server1 (but connected to server 2). I am using same domain name for two servers. What can be done in kamailio to solve this issue ?? Regards, aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Call-connection-from-one-server-to-… Sent from the Users mailing list archive at Nabble.com.

3 3

TLS Handshake failing with WSS
by Manuel Camarg 08 Sep '14

08 Sep '14

I'm trying to implement WSS with Kamailio Thing is that WS works fine, I've followed: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket modparam("tls", "config", "webrtc/tls.cfg") In a tls.cfg file I have : [server:default] method = SSLv23 verify_certificate = no require_certificate = no private_key = webrtc/private.key certificate = webrtc/ssl.pem ca_list = webrtc/ca_list.pem In the log file: /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=5524 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8e0040, 11, 2, 0x7f7513516958), fd_no=1 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:712]: sr_ssl_ctx_info_callback(): SSL handshake done /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL disable renegotiation /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 123.123.123.123:63300 using TLSv1/SSLv3 AES256-SHA 256 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 124.124.124.124:10443 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=282 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f7513516958, FD 11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1293]: tcp_read_req(): tcp_read_req: EOF /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e0040, 11, -1, 0x10) fd_no=2 called /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f7513516958, state -1, fd=11, id=2 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f7513510a88 /usr/local/sbin/kamailio[4029]: DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f7513516958, -1 from 1 /usr/local/sbin/kamailio[4029]: DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7f7513510a88 In sipml5 the error: *Disconnected: Failed to connect to the server* In the Chrome console: *__tsip_transport_ws_onerror * *__tsip_transport_ws_onclose * SSL certificates seem to be ok: # openssl verify -CAfile ca_list.pem ssl.pem ssl.pem: OK Can't figure out a solution :( Any ideas? *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

2 3

Change caller name in kamailio as the extension remains same.
by Talha Bukhari 08 Sep '14

08 Sep '14

Hi, I am new to kamailio. i configure kmailio server and make call as local extensions like 1001 and etc. Now i want to know how to change the header name appear on softphone using script in kamailo as the extension remain same as 1001. what are the files i used to alter and where are to locate them. Hope you don't mind. Regards, Talha

2 4

out of memory ; needs 455
by Arun Kumar 08 Sep '14

08 Sep '14

Hi Server : kamailio 4.1.1 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Problem : after 1 month kamailio hang up with out of memory error but we have enough free shared memory , it might be memory pool size problem, i dont know how and where to set memory pool size

2 1

Kamailio - voicemail platform - asterisk
by Henny 08 Sep '14

08 Sep '14

Hello, I am new to kamailio...... I have an idea for a community voice-mail platform (with Asterisk)and was wondering if Kamailio could be of any use in this plan. Does somebody have any advice ? Thanx!

2 1

dialog match on loop back
by Kelvin Chua 06 Sep '14

06 Sep '14

hi guys, i'm running out of ideas i have a situation where I need to generate 2 sets of entries INVITE,BYE in acc (acc_db_request won't cut it) it's when a DID should be redirected to a user outside my domain, I need to generate a cdr for DID and another CDR for the outbound call. what i am trying to do is loop it back to itself before routing it out. the idea is kamailio assigns the DID call to a dialog. then the outbound call will be assigned to a separate dialog after it receives the loopback. this is not happening since the from_tag, to_tag and the callid are all the same. kamailio treats this call as just 1, not 2. is there any ideas out there to work around this scenario? Kelvin Chua

2 1

Adding IPs from which 503 messages originates to a database
by aft 06 Sep '14

06 Sep '14

I'm trying to configure HOMER sip capture server to do some accounting of the cases when "503 Service Unavailable" message arrives from capture agents. I've implemented the following to record the source IPs from which 503 message originates. onreply_route { ...................... ...................... ...................... else if(status == "503") { if($sht(a=>alarm::503) == $null) { $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } xlog("Got 503"); xlog("($sht(a=>alarm::$si::503))"); $sht(a=>alarm::$si::503) = $sht(a=>alarm::$si::503) + 1; } ................... ................... ................... The following route supposed run periodically to update the databases. } route[TIMER_STATS] { ..................... ..................... xlog("($sht(a=>alarm::$si::503))"); if($sht(a=>alarm::$si::503) > 0) { sql_query("cb", "INSERT INTO alarm_data (create_date, type, total, description, source_ip) VALUES(NOW(), 'Too Many 503', $sht(a=>alarm::$si::503), 'service unavailable', $si)"); xlog("adding stuff to db"); } $sht(a=>alarm::503) = 0; $sht(a=>alarm::$si::503) = 0; } But the xlog("adding stuff to db") never executed. But in the first block of code, the log message to print the value $sht(a=>alarm::$si::503), which is shown to be non zero. Don't understand why its not working. -- -Cheers -Arif

2 4

Add custom parameter to To: header, and then recover it
by Alex Villacís Lasso 05 Sep '14

05 Sep '14

What functions exist (if any), to add custom parameters to the From: and To: headers?

2 1

Kamailio as a front end?
by Tom Johnson 05 Sep '14

05 Sep '14

I am looking for something that would act as front end to various SIP PBX systems. I am using Asterisk, but compatibility with others might be necessary. What I want to do is pass subscribe requests with certain content types to an external program (written in Java) and the rest onto the SIP PBX. Can Kamailio do this and if so, is there some example config files somewhere that I can look at to get started? Tom Johnson<mailto:tjohnson@microautomation.com> | Senior Software Engineer O: 703-543-2118 | F: 703-543-2099 5870 Trinity Parkway, Suite 600, Centreville, VA 20120 [cid:image001.png@01CE18AF.8F231EB0]<http://www.microautomation.com/>

2 1

mt_match avp behavior
by Juha Heinanen 05 Sep '14

05 Sep '14

looking at the code, i appears that mt_match does not initialize pv_values pv when it stores matched values thus keeping possible existing values from previous calls. is that intended behavior? readme uses word "store". -- juha

2 5

mtree README missing params
by Juha Heinanen 05 Sep '14

05 Sep '14

looks like 4.1 and master mtree README is missing these parameters: {"pv_dstid", STR_PARAM, &dstid_param.s}, {"pv_weight", STR_PARAM, &weight_param.s}, {"pv_count", STR_PARAM, &count_param.s}, i didn'd find in the code any use for the last. -- juha

2 1

has "defined" been documented somewhere?
by Juha Heinanen 05 Sep '14

05 Sep '14

defined($avp(foo)) can be used to test if avp $avp(foo) has been defined. i looked through kamailio wiki, but didn't find defined function documented anywhere. -- juha

2 1

ldap returning attributes list
by Virmantas Variakojis 05 Sep '14

05 Sep '14

Hi, I'm trying to fetch attribute list with kamailio ldap_search: ldap_search("ldap://agents/ou=Xxxxx,dc=xxxx,dc=yyyyyy,dc=zz?memberOf?sub?(sAMAccountName=$fU)"); ldap_result("memberOf/$avp(s:ldapmemberof1)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof1)"); if (ldap_result_next()) { xlog("L_INFO", "if loop ldap_result"); ldap_result("memberOf/$avp(s:ldapmemberof2)"); xlog("L_INFO", "LDAP $fU memberOf = $avp(s:ldapmemberof2)"); } There's must be seven returning attributes list of memberOf, but I can only retrieve only one of them (looks like last of seven). First xlog output log into my log file, but if loop didn't start. What I'm doing wrong?

3 2

create loop for REGISTER SEND in Kamailio.cfg
by ANTHONY HESNAUX 05 Sep '14

05 Sep '14

hello everybody, I would like loop in kamailio.cfg because I want register soft phone on all asterisk system so I have this configuration but also if I want add another asterisk system my Kamailio.cfg #!define WITH_ASTERISK #!define WITH_CLUST_AST #!ifdef WITH_ASTERISK asterisk.bindip = "asterisk.home.fr" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = « kamailio.home.fr" desc "kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif #!ifdef WITH_CLUST_AST #!define HOST 2 asterisk1.bindip = "192.168.0.10" desc « Asterisk - 1 - IP Address" asterisk1.bindport = "5080" desc "Asterisk Port" asterisk2.bindip = "192.168.0.20" desc « Asterisk - 2 - IP Address" asterisk2.bindport = "5080" desc "Asterisk Port" #!endif # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk1.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk1.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(); $var(rip) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk2.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 thank you for your help anthony

2 1

Dispatcher algorithm 9 ignores weight=xx attribute
by Alejandro Recarey 05 Sep '14

05 Sep '14

Hi, I have a kamailio with the dispatcher module setup. I use: ds_select_dst("1", "9"); to load balance between 3 gateways. My database looks like this: id setid destination flags priority attrs 5 1 sip:1.1.1.1 0 0 weight=10 8 1 sip:1.1.1.2 0 0 weight=45 9 1 sip:1.1.1.3 0 0 weight=45 As I understand it, most of the calls should be split evenly between gateways 8 and 9, and some calls (around 10%) going to gateway 5. Well, gateway 8 gets 99% of the calls. I am using kamailio 4.1 and I really cannot understand why this is happening. Running ds_list shows me that all gateays and weights are correctly parsed into kamailio. This is the result of my ds_list: SET:: 1 URI:: sip:1.1.1.1 flags=AP priority=0 attrs=weight=10 URI:: sip:1.1.1.2 flags=AP priority=0 attrs=weight=45 URI:: sip:1.1.1.3 flags=AP priority=0 attrs=weight=45 I have tried changing the priority to 100, or to 1 in every gateway, and the result is still the same. I have tried adding ; after the weight parameter, the colon is gone in the ds_list once kamailio parses it. If, keeping the same database, I use ds_select_dst("1", "0") the calls are split evenly across all 3 gateways. This is great, but not what I want, but it at least proves that something is working. Is there anything else I can try? Thanks a lot for the help! Alex

2 1

OT: RTCP logging
by Daniel Tryba 04 Sep '14

04 Sep '14

Slightly offtopic, but for QA reasons I'm interested in logging RTCP reports from peers that send them. It looks like rtpproxy-ng might be used to do this (I'm already using rtpproxy anyway). Is someone already doing this and willing to share some pointers?

3 4

Looking for some help
by Sharan Harkisoon 04 Sep '14

04 Sep '14

I am in need of a Kamailio expert that has some availability for consulting services (remote is fine). Feel free to contact me for details. Thanks, Sharan Harkisoon

2 1

Redirect server vs CANCEL
by Mihaly Zachar 04 Sep '14

04 Sep '14

Hi Gents, We have created a small redirect server using the Lua API to get the destinations. Lets say that it takes some time while the script generates the answer, during this time we got a CANCEL. How we can handle this properly ? As far as I could see the TM module always sends back the answer (302) and after that I get the "CANCEL" in my script. Here is my config: http://pastebin.com/7PyKBJyD Here is my test Lua script: http://pastebin.com/6VXn5y8m Could you please help me what is the best way to handle the CANCEL ? Is it possible to do it properly ? Thanks, Misi

2 1

nat_traversal vs nathelper for mediaproxy
by Satish Patel 04 Sep '14

04 Sep '14

I am planing to implement mediaproxy so which NAT module will be good for media proxy and why? should i use nat_traversal ro nathelper ? I haven't see any example people using nat_traversa. most of example i found on google are based on nathelper so just want to make sure which one i should use and why?

2 1

Shared memory
by Morten Tryfoss 04 Sep '14

04 Sep '14

Hi, I wonder about the difference between “used” and “real used” shared memory in Kamailio. Best regards, Morten Tryfoss

2 1

How to determine if a 4xx message came from dispatcher or a client?
by Olli Heiskanen 04 Sep '14

04 Sep '14

Hello, A question on Kamailio variables and using dispatcher: When in failure_route I want to know if the request message was going to a dispatcher ip or a sip client ip (as in any other than dispatcher ip), how do I make an if statement for that? If I use ds_is_from_list(), I get wrong results because in both cases the request was sent by Kamailio. Also, I don't seem to be able to get the source ip of the 4xx message (src_ip shows the source ip of the request message). cheers, Olli

2 1

Can log in from SIP
by Alexis Walhovd 04 Sep '14

04 Sep '14

Hi I just created an account with iptel.org. But I can not log in to my account through my SIP software from OSX. Please see below for details: I type “Alexis82” as password. All the best, Alexis

2 1

need help - Insert_hf when Route: missing
by Satish Patel 04 Sep '14

04 Sep '14

I have post many question on topology hiding any not get any reply back from people and developers now I don't have any option except some goofy hack When I use topology hiding it removes Route: and because of that callee doesn't able to send BYE back to opensips. I want use insert_hf to inject Route: so I get clean BYE message. Do you guys have any suggestion? Sent from my iPhone

2 1

Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
by Tim Chubb 04 Sep '14

04 Sep '14

Hi All I'm scratching my head on this one, I have set everything up in the following configuration: Public ip natted to DMZ via Cisco ASA 5510 <= DMZ => Kamailio & RTP Proxy <= Voice DMZ => Asterisk My kamailio configuration is based heavily on the one found here: http://saevolgo.blogspot.co.uk/2013/08/rtpproxy-revisited-kamailio-40.html & http://saevolgo.blogspot.co.uk/2011/11/how-to-increasing-voip-services.html So far everything is working fine, I can register via the public IP address, IM & presence is working, and as does audio, however when I dial an extension the caller id comes up like this 12345@172.16.15.123:5080<mailto:12345@172.16.15.123:5080> which is the ip addresses of the asterisk server that the dispatcher has assigned to the call, I have tried setting the P-Asserted-Identity, P-Preferred-Identity & Remote-Party-ID headers to no effect. What I would like to achieve is that the public IP or domain name comes up when I call an extension. From examining the sip traffic traversing the kamailio box, it seems that the From and Contact headers sent by asterisk are the source of the internal implementation information reaching the end-user. How can I prevent implementation information like this leaking out? I consider this to be a security risk as I don't want anyone to be able to enumerate the number of servers that comprise a given cluster as easily as placing a few calls, ideally I only want data relating to the public side of the network getting out. But I would be happy if it takes digging into the sip traffic to glean any details initially. I'm getting the feeling that this could well be an asterisk configuration issue, so apologies if this is a little off topic, however I have been unable to find a similar configuration to try and achieve my desired results. Tim.

3 3

Support for TLS server_name extension (aka SNI=server name indication)
by Barry Flanagan 04 Sep '14

04 Sep '14

Hi, Back in Kamailio 1.5, the release notes state: "support for TLS server_name extension (aka SNI=server name indication)" However, I cannot find any indication of this in the current TLS docs, and trying to set tls_server_name or server_name in tls.cfg fails with "unsupported option". Is this actually supported? Thanks. -Barry Flanagan

4 6

Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)
by Alex Villacís Lasso 04 Sep '14

04 Sep '14

Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 > 201.234.196.170.5060 INVITE sip:*43@201.234.196.170:5060 SIP/2.0 Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport Max-Forwards: 70 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Contact: <sip:9002@198.58.101.75:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Fri, 29 Aug 2014 18:23:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 299 v=0 o=root 521741684 521741684 IN IP4 198.58.101.75 s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 c=IN IP4 198.58.101.75 t=0 0 m=audio 16426 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060> Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: kamailio (4.1.5 (x86_64/linux)) Content-Length: 0 IP 201.234.196.170.5060 > 198.58.101.75.5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes> From: "9002" <sip:9002@198.58.101.75>;tag=as0bc522a9 To: <sip:*43@201.234.196.170:5060>;tag=as2798a3b9 Call-ID: 2c14c21f5052a74a78ca4ab736657b00@198.58.101.75:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:*43@127.0.0.1:5080;alias=127.0.0.1~5080~1> Content-Type: application/sdp Require: timer Content-Length: 305 v=0 o=root 159029581 159029581 IN IP4 201.234.196.170 s=Asterisk PBX 11.12.0 c=IN IP4 201.234.196.170 t=0 0 m=audio 18446 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between 201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.

3 7

want to know about best book or best way to learn kamailio.
by Talha Bukhari 04 Sep '14

04 Sep '14

please guide me, how i learn kamalio as i am new to kamailio. suggest me some book or resources. waiting.

2 1

Re: [SR-Users] How to add route header (Senthil K)
by Surendra 03 Sep '14

03 Sep '14

Hi Senthil, You can use record_route function to add Record_route header in INVITE message( use rr.so module). This record_route header will become Route header while sending BYE message from client. Suppose if you want add directly Route header, then you can use Insert_hf function. Please go thru the textops modules for the same. Regards surendra

1 0

How to uniquely identify SIP WS / WSS endpoint
by Muhammad Shahzad 03 Sep '14

03 Sep '14

Sorry for putting this question on both dev and user mailing lists, as it is a rather theoretical question and i hope some SIP guru on either mail list will answer. For non-WS endpoints which use TCP or UDP for SIP transport, each upstream request has top most VIA header pointing to the previous hop which forwarded the request to current hop while each downstream request has top most VIA header pointing to next hop to which it will be forwarded from current hop. But for WS endpoints, the top most VIA has dummy static value, so there is no way to identify who sent this request or to whom the reply is going to. Please note that i am not specifically interested in network address of remote endpoint (though VIA header is suppose to provide it), i only need to match requests and responses from / to a specific device using SIP v2.0 standard. Any help is highly appreciated. Thank you.

3 7

Use Kamailio as (SIP-TLS) Gateway to connect to (non TLS) fritz.box for fixed-line calls
by Lukas Wygasch 02 Sep '14

02 Sep '14

Dear Kamailio-Community, i would be very pleased when someone could help me with my private non-profit project. I do not want other people to do my work, but i'm not as good with SIP as i should to solve this problem by myself. So here is my Goal: Use Kamailio as (TLS-SIP-)Gateway to my (non-TLS)-FritzBox to do (local-country) fixed-line Calls from all over the World. In Detail: Connect to (Portforwarded / NAT behind Fritzbox-Router) Kamalio-Server with User from "Subsriber-List" by TLS-SIP Now I want Kamailio to connect to local fritz.box (by SIP) with one fritzbox-user to handle all invites and calls. ------------------------------- Scenario (fictional): Fritzbox (Router and VOIP-Server) Internal-IP: 192.168.178.1 or fritz.box External-IP: dynip.de VOIP-User: 610 VOIP-Passwort: test Kamailio (running on RaspberryPI and Siremis) Internal-IP: 192.168.178.2 Network Gateway: 192.168.178.1 Connection from Outside (NAT): dynip.de:5061 User: euser Pass: epass ------------------------------- Questions: Is there a Module i could use to do this Job? (configurable through Siremis e.g. Dispatcher List?) How can i ensure that my user "euser" from subsriber-list connects with 610:test to fritz.box? Is there a simple configuration to solve this problem? I'm sorry to ask you these my non-skilled questions, since it is not my business to use SIP, but this scenario could be a useful tutorial for many many fritz.box users over the world. Roaming prices for mobile-calls could be safed with sitting into next McDonalds and do a SIP-Call through free Wifi. Thank you very much! I would be very grateful to get some help. Sincerly yours Lukatz

2 1

Regrading Kamailio
by Saurabh Srivastava 02 Sep '14

02 Sep '14

Hello Kamailio Team, I am using kamailio server successfully for some time now. I have a specific set of problem. We have a Kamailio Server setup in Amazon EC2 Cloud, the users are able to register on the server and make use of this. *Now the problem:* If I am trying to register multiple SIP clients from same Public_IP:Private_IP (i.e. alice and bob from same computer) on a kamailio server which is sitting behind Amazon EC2 Cloud Based Server, then only the first user (who sends REGISTER packet first) is able to register successfully on the server and all the other users after the first one are not able to register on the server. Am I missing something in the configuration file such that this is happening. Please help in this regard. Waiting in anticipation Thanks & Best Regards Saurabh Srivastava

2 1

Roadmap to next major release - v4.2.0
by Daniel-Constantin Mierla 02 Sep '14

02 Sep '14

Hello, during the last devel meeting on IRC, done before the summer, we set the 4.2.0 release time frame for this autumn. As planned by that moment, we have to decide the development freezing date and more accurate milestones at the end of summer, therefore it is about the time. My proposal is to freeze development by end of September 10, 2014, have about one month of testing and release sometime around mid of October. If anyone has constraints to meet this date, just send here another proposal and we can decide together which one fits most of us. This one gives about 2 weeks and a half for new features development. Therefore, if you plan to include a new module (many were announced at the IRC devel meeting, hopefully some are ready), hurry up. It is also a good time to start filling the wiki page with what is new in upcoming release: - http://www.kamailio.org/wiki/features/new-in-devel Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 1

How to add route header
by Senthil K 02 Sep '14

02 Sep '14

Hi All, Kindly check my issue and please provide me your suggestions. In my requirement, I want to add a route header to the INVITE message which is sent out from the Kamailio server, Basically I will be making a call and that INVITE will reach the Kamailio, then the Kamailio will send the INVITE to another node, here at this situation when the Kamailio is sending the INVITE to the next node, I want to add a route header in INVITE. Please let me know how can I hard code a route header to the outgoing INVITE from Kamailio server, Kindly help me Kind Regards SenthilK

2 1

PhonerLite cant login
by Donovan 02 Sep '14

02 Sep '14

Hi, I hope you can help me. My login details are correct but it keeps saying timeout. It just suddenly stopped working. I have made no changes? https://url.odesk.com/_01tJ3v5df4REETK4PV-B3hQtMyc7vTyOQO Best, Don --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

2 1

To mark user as unreachable with nathelper.
by Igor Pavlov 02 Sep '14

02 Sep '14

Hi, colleagues. Is it possible to mark user as unreachable (ex. by setting flag in location table) with nathelper-module when it pinging user? I need something like asterisk's qualifying. ----- Best regards, Igor Pavlov

2 1

Adding a media attribute to SDP
by aft 01 Sep '14

01 Sep '14

Hi, I want to add this media attribute to SDP. a=ptime:<packet time> [1] Its not covered in SDPops module [2] its not added when i use rtpproxy_manage("z<packet size>") to repacketize the rtp stream. So how should i do it? -- -Cheers -Arif

2 4

Wrong ACK to Provider
by Yuriy Gorlichenko 01 Sep '14

01 Sep '14

Hello. I try to provide call scheme: internal client -> asterisk -> Kamailio -> provider -> external endpoint call when I make call I see this: asterisk kamailio provider invite --> invite --> <-- 407 ACK --> invite w/Auth --> <-- 100 <-- 100 <-- 180 <-- 180 <-- 183 <-- 183 <-- 200 <-- 200 ACK --> ACK --> My problem with last ACK, that I send to provider. Provider ignores it, and sends me some OK packets. As resultI can notend session ( answer to BYE 481 - transaction does not exists). I think it is wrong ACK but can not undrtand where I do mistake. Please help me to find it: My invite (with Auth creditans): IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 1606 E...]. .@..R ...6........N0TINVITE sip:12345678900@my.provider.ip:5060 SIP/2.0 Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Wed, 27 Aug 2014 22:02:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="provider_username", realm="my.provider.ip", nonce="U/5Wv1P+VZNjFBLf6fwPizgd6iLto5St", uri="sip:12345678900@my.provider.ip:5060", qop=auth, nc=00000001, cnonce="2888860875", response="9f23110471fe9ff751cd55466e70ded2", algorithm=MD5 v=0 o=root 1370647246 1370647246 IN IP4 12.34.56.78 s=Asterisk PBX 12.5.0 c=IN IP4 12.34.56.78 t=0 0 a=ice-lite m=audio 30296 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30297 a=ice-ufrag:p5k92ynl a=ice-pwd:FIOYKt96NlBfEqKsQipUuadUev1g a=candidate:vV3V06Tv Provider trying IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 500 E.........PX6... ..........ySIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1;rport=5068;received=12.34.56.78 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 provider ringing IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1098 E..f......M.6... ........RV.SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no provider seesion in progress IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 1887 E..... ...,.6... ........g.DSIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 742 Remote-Party-ID: "12345678900" <sip:12345678900@my.provider.ip >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=ssrc:326362635 cnam provider OK IP my.provider.ip.5060 > 10.0.1.18.5068: UDP, length 2026 E..... ...,.6... ...........SIP/2.0 200 OK Via: SIP/2.0/UDP my.external.ip:5068;rport=5068;received=12.34.56.78;branch=z9hG4bK48ba.74ed5eb56b172cd802c50dcc201cce56.1 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK258b5220;rport=50600 Record-Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Record-Route: <sip:my.external.ip:5068;nat=yes;ftag=as7d06fc50;lr=on> Fл2rom: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 INVITE Contact: <sip:12345678900@67.192.253.160:5060;transport=udp> User-Agent: Plivo Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE SupлЛ o=FreeSWITCH 1409149800 1409149801 IN IP4 67.192.253.160 s=FreeSWITCH c=л2IN IP4 67.192.253.160 t=0 0 a=msid-semantic: WMS uIWGGSqM8mUp5NEgQ9CU0svyzqjzisqD m=audio 27180 RTP/AVP 0 my ACK IP 10.0.1.18.5068 > my.provider.ip.5060: UDP, length 614 E...]...@... ...6........n.hACK sip:12345678900@my.provider.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.external.ip:5068;branch=z9hG4bK48ba.4250e4d315c4aa6697b6d7f70e861b62.0 Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK4d28fc11;rport=50600 Route: <sip:my.provider.ip;lr=on;ftag=as7d06fc50;did=5bc.33f1> Max-Forwards: 70 From: "John" <sip:provider_username@my.provider.ip>;tag=as7d06fc50 To: <sip:12345678900@my.provider.ip:5068>;tag=v9g4HD4vrNFUH Contact:<provider_username@my.external.ip:5068> Call-ID: 2122fc6a3cbe2e64253289cf23c3dd2a@10.0.1.6:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 So after this ACK provider still sends me 200 OK and my server still sends ACK.... tags and call-id always one. Thanks

3 11

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, if you have a patch and want to get it in the main repository, send it and if all looks ok, we can commit it -- that will ensure the feature is there with new releases of kamailio. Cheers, Daniel On 01/09/14 10:28, Surendra wrote: > Hi Veerabhadra, > > Kamailio doesn't support this. We had done this by changing the code > logic. > > Regards > surendra > > > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

[OT] vim auto-complete plugin for kamailio.cfg
by Daniel-Constantin Mierla 01 Sep '14

01 Sep '14

Hello, I published a vim plugin offering basic autocomplete for kamailio configuration file. It is available via githup at: * https://github.com/miconda/vim-extensions More details about the plugin (including how to install and use) can be read at: * https://github.com/miconda/vim-extensions/blob/master/plugin/kamailio/READM… It was written more as a proof of concept so far, hopefully interested people will join development to bring it to a 'smarter auto-complete' plugin. Right now it relies on custom autocomplete feature from vim -- once activated, as you type a word, by pressing CTRL+x CTRL+u a pop up box presents possible completion by matching the typed part of the word. The dictionaries include module functions and parameters, script variables, core functions and parameters (they were generated with some scripts grepping various files, thus they might not be complete). It is not (yet) aware of context, so it displays all matches (e.g., even when in a route block, it displays core parameters), so again, everyone is welcome to contribute/fork/pull requests to develop the plugin further. The readme lists some other goals as well. Hopefully it is going to be useful at least for some people in the community! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Re: [SR-Users] Presence - How to force Kamailio to use one subscription
by Surendra 01 Sep '14

01 Sep '14

Hi Veerabhadra, Kamailio doesn't support this. We had done this by changing the code logic. Regards surendra

1 0

Presence - How to force Kamailio to use one subscription per user?
by Veerabhara Gundu 30 Aug '14

30 Aug '14

Hello Friends, I have a scenario, where clients are going away and re-registering with new IP and Port numbers. Presence server keep sending NOTIFY messages for both old Subscription and new Subscription for same clients till the old subscription expires. It's unnecessary traffic and processing on both client and server. I am looking for a way to force Kamailio to use one subscription per user similar to mode:04 of location table. If it's not possible, looking for a way to remove subscription from config file when I detect this error.We have an option to remove automatically Subscription when NOTIFY failed, but this is not working for TCP based clients. In TCP mode, when Kamailio not event trying to send NOTIFY if Kamailio already knows that TCP connection is broken and replying with 500 error. Any alternative suggestion also welcome. Thanks, Veera Thanks, VG

1 3

What does "::" operator do in statements related to hashtable?
by aft 30 Aug '14

30 Aug '14

Hi, >From kamailio documentation, the usage of hashtable is given as : modparam("htable", "htable", "a=>size=4;") ... $sht(a=>test) = 1; $sht(a=>$ci::srcip) = $si; I get it that in the first statement, a is the hashtable, a new key-value pair is added to one of its empty bucket (test,1). What is happening in the second statement? -- -Cheers -Arif

3 3

Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS
by Alex Villacís Lasso 29 Aug '14

29 Aug '14

As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration. In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost): REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0 Asterisk answers this through UDP, and Kamailio forwards it through the websocket: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0 Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket: NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0) Since I have not implemented handling of voicemail indications, the browser answers this: SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0 After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws . I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?

2 5

How to use the ip parameter in rtpproxy_manage()
by Sebastian Damm 29 Aug '14

29 Aug '14

Hi, I have the following setup: UAC ---> LB ---> Proxy ---> GW In NAT scenarios the loadbalancer detects it, but the proxy communicates with the RTP proxy. I want to send the original caller IP as detected by the loadbalancer (and transported to the proxy) to the RTP proxy. As far as I understand the documentation of the rtpproxy module, I could call rtpproxy_manage with a second parameter, indicating which IP address should be sent to the RTP proxy. I tried sending a pseudovariable, both within quotes or not, or even a static string. But when I look at those messages sent to the RTP proxy, there's always the IP from where the SIP packet was received (the loadbalancer IP) in the request. Previously we used the "r" parameter, sending the original IP from the SDP to the proxy. But due to strange behavior of some UACs, we want to get rid of that. Do I misunderstand the second parameter? This is what the documentation says: * ip_address* - new SDP IP address. Unfortunately, I couldn't find any example using this. Best Regards, Sebastian

2 3

Help analysing segmentation fault
by Charles Chance 29 Aug '14

29 Aug '14

Hi All, I wonder if some one could help me to diagnose a recurring issue? It happens at random times/intervals and under varying load. But always, just before the time of crash, I see the same critical error in log: CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in state 1 for dlg 0xb0632134 [1367:5814] with clid ' 0695dd7a346188dd24e7520e6c01092c(a)sip.sipcentric.com' and tags 'as77c89620' '' Analysing the core dump reveals: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 128 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 0x081a737c in parse_uri (buf=0x3a70006e <Address 0x3a70006e out of bounds>, len=275, uri=0xbfa2fd2c) at parser/parse_uri.c:389 389 scheme=buf[0]+(buf[1]<<8)+(buf[2]<<16)+(buf[3]<<24); (gdb) frame 1 #1 0x008fe5dd in dialog_publish (state=0x903f37 "Trying", ruri=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:275 275 if (parse_uri(ruri->s, ruri->len, &ruri_uri) < 0) { (gdb) p *ruri $1 = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275} (gdb) up #2 0x008ff277 in dialog_publish_multi (state=0x903f37 "Trying", ruris=0xb0b5fd00, entity=0xb0632188, peer=0xb0632190, callid=0xb0632180, initiator=1, lifetime=7200, localtag=0x0, remotetag=0x0, localtarget=0x0, remotetarget=0x0, do_pubruri_localcheck=1) at dialog_publish.c:387 387 dialog_publish(state,&(ruris->s),entity,peer,callid,initiator,lifetime,localtag,remotetag,localtarget,remotetarget,do_pubruri_localcheck); (gdb) p *ruris $2 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} (gdb) up #3 0x0090187a in __dialog_created (dlg=0xb0632134, type=2, _params=0x6db064) at pua_dialoginfo.c:470 470 dialog_publish_multi("Trying", dlginfo->pubruris_caller, &(dlg->from_uri), (include_req_uri)?&(dlg->req_uri):&(dlg->to_uri), &(dlg->callid), 1, dlginfo->lifetime, 0, 0, 0, 0, send_publish_flag==-1?1:0); (gdb) p *dlginfo->pubruris_caller $3 = {s = {s = 0x31590014 <Address 0x31590014 out of bounds>, len = 275}, next = 0xb0b5fd00} (gdb) p *dlginfo->pubruris_caller->next $4 = {s = {s = 0x3a70006e <Address 0x3a70006e out of bounds>, len = 275}, next = 0x0} In config, for pua_dialoginfo we are enabling the option "use_pubruri_avps" and setting "pubruri_caller_avp" and "pubruri_callee_avp" accordingly. Therefore, in pua_dialoginfo.c it is using get_str_list() function to set dlginfo->pubruris_caller from the avp. Could this be some race condition or something completely different? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

3 7

Add websocket header to SIP
by Kamrul Khan 28 Aug '14

28 Aug '14

Hi, I want to append the origin header of my incoming websockets to the SIP INVITES. I tried something like the below but didnt work: if (ws_handle_handshake()) { if(method=="INVITE"){ insert_hf("Origin: $hdr(Origin)"); } exit; } Is there any other way to do it ?

2 3

[Kamailio-4.1.5] [/etc/init.d/kamailio/start] Error
by Djamel Bahamid 28 Aug '14

28 Aug '14

Hello, When I try to execute the /etc/init.d/kamailio start, I've this error : [root@gtk:~][Mon 18/08/2014 16:21:58] # /etc/init.d/kamailio start Not starting Kamailio SIP server: invalid configuration file! failed failed ERROR: loading config file(/usr/local/etc/kamaiilio/kamailio.cfg): No such file or directory failed failed Thank you for your help, Best Regards, Djamel BAHAMID.

3 5

rfc: distributing dialog profiles
by Daniel-Constantin Mierla 28 Aug '14

28 Aug '14

Hello, if you are on sr-dev list, you should have seen the last commits to dialog module, which add support for remote profiles. As most of you know (or at least the users of dialog module), profiles are used to group active calls, their main purposes being counting dialogs belonging to a profile for implementing active call limits (e.g., all calls in the system, call from/to a specif user/address). It was working on a single Kamailio instance, but one could use other meanings to interact with other kamailio instance to get the number of calls across multiple instances. The purpose of remote profiles is to make the whole process easier and more transparent for config file, simplifying the maintenance. A remote profile is an item stored in local profile list not bound to a local dialog. At this moment, dialog module in master branch exports a function dlg_remote_profile() that allows to add/remove a remote profile. The next step is to build the notification system between kamailio instances. There are couple of options, as well as questions, that I want to discuss before going to implement the mechanism. There are two major aspects to care of: A) Distribution, I thought of two options for it: 1) using dmq module to publish the operations done with local profiles (add/remove) -- this seems to be the natural choice right now, Charles Chance is also willing to put effort in this direction This will be coded inside the module, so config won't be affected much (eventually some extra parameters of dialog flags). 2) add event routes for adding/removing a local profile. The attributes of the profile can be made available via some PV, then in config use what so ever option (e.g., http query, json rpc, etc) for distributing the information. This will require changes in config file and might not be trivial for new users B) Persistence, with the options: 1) store remote profiles in local database. It is good for restart, but if there is big downtime, then some notifications can be lost and the list of remote profiles might be inaccurate 2) keep in memory only and at startup send a 'request-for-update' to the other instances -- these instances will send the list of their local profiles. It can take a bit between request-for-update and receiving data from the other servers -- during this interval, the checks of overall call limits might not give the desired answer Of course, there can be enhancements/combinations, but it has to be analysed if the benefit worth the complexity. For 1), store also the IP of each remote profile and then interrogate for each profile to see if its still active. For 2), add a mechanism to reject new calls until the request-for-update is completed. Anyone having other ideas or comments to the above? Note that each profile got a unique id, so if you upgrade your dialog module, profiles for existing active calls will not be loaded. Also, the code was barely tested, report here any issue you have with dialog from master branch. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

5 9

Calls to devices behind NAT not working
by Adnan 28 Aug '14

28 Aug '14

Hi all, I'm trying to configure a Kamailio server, and am having trouble getting devices behind NAT receive calls. Looking at a SIP trace, nothing is being received by the end point, even though Kamailio seems to be sending out a response. I've opened up my firewall rules, so that shouldn't be an issue. If I call another endpoint which is behind the same NAT, it works (although no audio), and looking at the trace, the SIP messaging is coming from the Kamailio server (ie. not the other endpoint). I've even tried using the default kamailio-advanced.cfg & kamailio-basic.cfg both of which didn't work either. I've attached the current config I'm working on, if someone could please have a look through it, and see where I'm going wrong I'd highly appreciate it. Thanks Adnan

1 0

Minimal configuration without database support
by Eugene Prokopiev 27 Aug '14

27 Aug '14

Hi, Is it possible to run kamailio without database support (even dbtext)? Where can I find minimal configuration example with minimal modules set only for forwarding any requests/responses from one ip address to another one? -- WBR, Eugene Prokopiev

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