- sr-users - kamailio.org

Presence Module Installation without reinstalling Kamailiio
by aawaise 13 Oct '14

13 Oct '14

I am using kamailio server. I want to add xcap server for presence module usage without reinstalling kamailio server. How can it be done. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Presence-Module-Installation-withou… Sent from the Users mailing list archive at Nabble.com.

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Presence Module Installation Error
by aawaise 13 Oct '14

13 Oct '14

Hello, I am using kamailio-3.1.1 server. I want to add presence module service to it. For that I tried to reinstall server by downloading kamailio-3.1.1 package from official web link. When I include modules "db_mysql mysql tls presence presence_xml xhttp xcap_server" in modules.lst. It gives error while running "make all" command. But when kamailio is installed without addition of presence module with "include modules = tls mysql db_mysql", it installs fine. I tried using kamailio-4.1.0, it … [View More]

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how to always use rtpproxy-ng and rtpengine to relay rtp packets
by andrew 13 Oct '14

13 Oct '14

hi, kamailio 4.1.5 is used with rtpengine, whose older counterpart is rtpproxy-ng. I made some changes based on the default configuration files(i.e. kamailio.cfg), so that rtpproxy-ng module is enabled. Kamailio.cfg has been uploaded. In kamailio.cfg, there is one route block related to rtpproxy, i.e. route[NATMANAGE], where rtpproxy_manage("co") is called. Sometimes, rtpproxy-ng can insert new ip:port candidates in SDP. But in some case, rtpproxy-ng doesn't rewrite SDP, so rtpengine doesn't … [View More]

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Roadmap to release v4.2.0
by Daniel-Constantin Mierla 13 Oct '14

13 Oct '14

Hello, after last days of testing at SIPit 31, I think the master branch is in quite good shape. I hope to get the time for a more detailed report, but we had only one major issue when running some stress tests, affecting websocket connections when using a special crafted websocket packet, fixed already. Otherwise, while the tests here are mainly on SIP interoperability, so not much on particular use cases of kamailio, with the help of Olle we have been in pretty much all multi-party … [View More]

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Alias database
by Alberto Llamas 13 Oct '14

13 Oct '14

Hello kamailio gurus, Thanks if somebody can help me answering nex question: I have a kamailio server as registrar server with asterisk behind. For register porpouse the UA are using subdomain like: 100(a)test.mydomain.com 200(a)test2.mydomain.com I have to add each alias in the kamailio.cfg file an restart the sevice. Alias="test.mydomain.com" Alias="test2.mydomain.com" Is there some option to add it in the database? Then i don't have to restart the server each time i have to add/edit/… [View More]

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483 too many hops
by Anil Kumar 13 Oct '14

13 Oct '14

Hi All, I am trying to register an ims-client/user to kamailio proxy, but I am seeing 483 too many hops response. I know one of the reason for this, that is MAX-FORWARD limit has reached. But in my case I feel I have given a proper value (70) , but still some looping is happening in the middle and limit is reached. So can anyone please help me, whats going wrong here. Any configuration changes required ? Thanks Anil

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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:… [View More]

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

10 Oct '14

Good evening! I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to compile and install everything, if I run : service kamailio start it starts kamailio and if I run service kamailio status I tells me that is running on PID... which is fine, however, if I tweak the /usr/local/etc/kamailio/kamailio.conf according to the instructions on http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb , when I run service kamailio start It says "ok" , but … [View More]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> | [View Less]

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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=[… [View More]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 >> >> > [View Less]

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","… [View More]

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 … [View More]

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 --… [View More]

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 … [View More]

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 … [View More]

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 … [View More]

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 … [View More]

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 … [View More]

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",… [View More]

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) … [View More]

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.… [View More]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. > > [View Less]

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 … [View More]

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. … [View More]

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 … [View More]

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? >… [View More]

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: … [View More]

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 … [View More]

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:… [View More]

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 … [View More]

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:… [View More]

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 … [View More]

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 … [View More]

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 … [View More]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 } [View Less]

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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 … [View More]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... [View Less]

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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

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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 … [View More]

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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: \… [View More]

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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 … [View More]

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[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 … [View More]

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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 … [View More]

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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 … [View More]

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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, … [View More]

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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 … [View More]

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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

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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: … [View More]

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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 … [View More]

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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 … [View More]

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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:… [View More]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 [View Less]

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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/… [View More]

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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 … [View More]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! [View Less]

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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:… [View More]

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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 … [View More]

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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 … [View More]

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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 … [View More]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. [View Less]

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 … [View More]

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 … [View More]

2 1

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 … [View More]

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) - … [View More]

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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 >>> … [View More]

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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; … [View More]

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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, … [View More]

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,… [View More]

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.… [View More]

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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-… [View More]

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(… [View More]

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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 … [View More]

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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, … [View More]

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 … [View More]

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; … [View More]

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 … [View More]

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 "&… [View More]

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; #!… [View More]

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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 … [View More]

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 … [View More]

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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:… [View More]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 [View Less]

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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 … [View More]

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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 … [View More]

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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; #!… [View More]

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[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

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