- sr-users - kamailio.org

Kamailio b2bua
by Grant Bagdasarian 20 Jun '14

20 Jun '14

Hello, OpenSIPS has a module for B2BUA. Does Kamailio have a similar one or one that comes close to it? Perhaps by combining modules to make Kamailio act as a B2BUA?

3 4

Initial Install
by Phil Daws 20 Jun '14

20 Jun '14

Hello and good evening all! Am trying to build a Kamailio installation in my lab and have hit the first hurdle; database build. I have a three node MariaDB cluster in place and attempting to get kamdbctl create to set up the tables. What seems apparent is that command requires the database to not already exist. Where can I find the necessary schema to load into MariaDB myself please ? Thank you. Phil

2 2

Setting priority (q) for kamctl ul add
by David Wilson 20 Jun '14

20 Jun '14

Hi All, I'm trying to add a permanent usrloc entry via kamctl ul add. This works, but the created entry has a q value of 1.0 which is higher than I need. Is there a way to either: 1. Specify a q value when using kamctl ul add, or 2. Edit the q value of an existing record by using a kamctl command. Cheers, Dave.

3 2

Starting Kamailio SIP server
by Horo, Priyam 19 Jun '14

19 Jun '14

Hi, I am following the quick-start installation guide at http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x and I am stuck when I start the kamailio, using /etc/init.d/kamailio start it gives me an error of "invalid configuration file". I am using the /usr/local/etc/kamailio/kamailio.cfg file which came along with tar.gz file. I do not want sql and no authentication to begin with. What else needs to be modified or is required in the config file. Thanks!

1 0

Error codes with management commands
by PIERRE Laurent 19 Jun '14

19 Jun '14

Hello, It's unfortunate that error codes are not included with management commands. For example, if you tape " kamctl diaplan rm rule 2 999 " instead of " kamctl diaplan rmrule 2 999 " then all the entries in dialplan table will be deleted. In fat the system perform "kamctl diaplan rm" It's very annoying . -- Laurent PIERRE http://www.linkedin.com/in/lpierre

1 0

Unable To Start MSRP Session
by Megha Patil 19 Jun '14

19 Jun '14

Hi, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error(Reason: SIP ;text="media stream failed to start" ;cause=500). What might be the problem?? Does it mean that kamailio can not handle msrp session? If it can handle, how to enable it? I have attached the captures with this mail. Please help me in this regard.

1 0

Re: [SR-Users] SIP Trunk
by Keith 19 Jun '14

19 Jun '14

Hi, Thanks Daniel, however don't I need to send a register to get the 401 back? If so how do I do this bit? Thanks Keith

2 1

Possible Oversight, but CDRs are not being written
by Joel White 19 Jun '14

19 Jun '14

I made a significant change to the way the system I am building is designed. I removed the Database to a redundant cluster and not on the same server as the kamailio install. I changed all references to the DBURL to reflect the IP of the new cluster. I also changed the SQLOPS statement for CDRs to reflect the new IP modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@10.11.12.5/kamailio") All ACC transactions are being recorded to the new DB and the configuration is running well with the exception of CDRs. What am I missing that would prevent writing of CDRs to the DB? I did check connectivity from the Kamailio server to the DB server as well as access to the cdrs table

2 3

Cisco 79XX Series Phones and TLS integration with kamailio
by Jonathan Hunter 19 Jun '14

19 Jun '14

Hi All, As you guys might remember I have been doing alot of work with these legacy handsets recently, in particular the Cisco IP phones 7945G and 7965G. They work well now with kamailio, using UDP or TCP as the transport protocol. I am now looking to implement SIP over TLS with them, and wondered if anyone had completed the same, as it appears they were designed very much to support SIP over TLS and SRTP but with Cisco call manager and not other SIP devices. I am trying to understand if its possible to integrate with kamailio, as from Cisco documentation it appears a CTL file (client trust list) is required which seems to be generated by cisco software, and the handset needs it before it will try to initiate a SIP connection over TLS. I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so its more a question around the handset and if anyone has achieved this. Thanks Jon

3 8

Internal server error
by Deepak Bhagwat 19 Jun '14

19 Jun '14

Hi all, I am trying to establish a chat session(MSRP) with two sipclients endpoints by registering to kamailio server(4.0.0), but I am getting 500 Internal server error. What might be the problem?? Does it mean that kamailio can not handle msrp? If yes how to enable it in the server? Packets captured are attached. Please help me in this regard. -- Thanks Deepak Bhagwat

1 0

dialog module -- timeout parameter
by Måns Nilsson 19 Jun '14

19 Jun '14

Hi, I need to allow arbitrarily long calls while still using the dialog module. The module documentation does not mention a maxiumum value (2^32 sec?) nor a setting to disable the timeout functionality. I'd guess that setting it to 0 is a common convention in such cases, but I'd rather get it verified. Anything above 10 years is probably a safe bet, so a 32-bit unsigned int is quite fine. :-) Is there a max value? Is there a way to disable the check? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I think I'll KILL myself by leaping out of this 14th STORY WINDOW while reading ERICA JONG'S poetry!!

2 1

One way Audio with kamailio -Rtpproxy
by master voip 18 Jun '14

18 Jun '14

Hi can some one help me . I am using kamailio 4.1.3 and having one way audio issue. Scenario is Pjsip 2.0 based UA (private IP)------->Router(publicIP)--------> Kamailio (with RTP proxy)-----------> Third Party_Sip Server ------>PSTN UA is registering fine with sip server but when i make a call they can hear but i can not hear the other party . What i found is in SDP first c=IN is private ip of UA and second c=IN is public ip of UA i dont know if this is the cause When I trace the call on sip server , sip server is sending RTP to UA private ip so it is not reaching kamailio back to forward to UA . I am attaching the kamailio configuration file as well as SIP/SDP trace user agent private ip : 192.168.1.4 User agent public ip : 61.61.61.61 Kamailio IP : 81.81.81.81 Sip Server IP: 71.71.71.71 (* ip addresses are not actual ips) Please reply with detailed instructions to fix this isue config file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # #!define WITH_NAT # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ' '; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ' '; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=3 log_stderror=no #!endif memdbg=1 memlog=1 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=7878 #!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" #!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" loadmodule "nat_traversal.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 7) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication # Below line commented by test-MATHUR as PC-2-PC calls are not getting through #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()) { xlog("testM : Unable to relay !!!! :{ \n"); 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 $fu (IP:$s i:$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")) { #xlog("testM : Got BYE from some PSTN/UAC.\n"); setflag(FLT_ACC); # do accounting ... #xlog("testM : Did accounting.\n"); setflag(FLT_ACCFAILED); # ... even if the transa ction fails #xlog("testM : Calling route{NATMANAGE} from WI THINDLG.\n"); #t_newtran(); #t_reply("200", "OK"); #xlog("testM : Called route{NATMANAGE} for BYE froom WITHINDLG.\n"); #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 R FC 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 ... i gnore 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))) #{ # rtpproxy_manage(); # return; #} rtpproxy_manage("z90"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { add_rr_param(";nat=yes"); } } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n" ); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bu g in # xmlrpclib: it waits for EOF before interpreting the response). if ($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_i p) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($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 } ------------------------------------------------------------------------------------------------------------------ SIP TRACE REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62147 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 P-hint: outbound SIP/2.0 407 Proxy Authentication Required CSeq: 62147 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK1b54.98ebaeef387ced8eb54624e6c7a90504.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjI5YxCTUEIkkWEg0VswESOCq7lpvwYUkE From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141058 Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301389817101016806045824224" Content-Length: 0 REGISTER sip:71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW To: "57778" <sip:test@71.71.71.71> Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 CSeq: 62148 REGISTER User-Agent: pj_arubaslim-16/r2 Contact: "57778" <sip:test@192.168.1.4:58931;ob> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301389817101016806045824224", uri="sip:71.71.71.71", response="0ca765f266b6bfc8ea0ef00227272393" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 62148 REGISTER Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKfb54.1eaeef967bf46fb647eed0bb87a9406e.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjRPPULjuny1lG1x2Qn0UumpeHPLGxqQHI From: "57778" <sip:test@71.71.71.71>;tag=ODxcd4MPCfZc.o5M6gAZObocSE9ZpbyW Call-ID: XsaPsPHKODzjjDr9TgogZFZs61hetHv7 To: "57778" <sip:test@71.71.71.71>;tag=170604141059 Contact: "57778" <sip:test@192.168.1.4:58931;ob>;expires=600 Expires: 600 Content-Length: 0 SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10987 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Content-Length: 0 P-hint: outbound SIP/2.0 401 Unauthorised CSeq: 10987 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK0f2c.8c652a23fe5f790c4bb9216d48b80199.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPj.syjKlsSGhtf420D3Gs4.gYpLomA2oQq From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141000 Content-Length: 0 WWW-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301390017101016806050024224" Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> SUBSCRIBE sip:test@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC To: "57778" <sip:test@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB CSeq: 10988 SUBSCRIBE Event: message-summary Expires: 3600 Supported: replaces, 100rel, timer, norefersub Accept: application/simple-message-summary Allow-Events: presence, message-summary, refer User-Agent: pj_arubaslim-16/r2 Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301390017101016806050024224", uri="sip:test@71.71.71.71", response="15ee3de757c18e084fca607438017093" Content-Length: 0 P-hint: outbound SIP/2.0 200 OK CSeq: 10988 SUBSCRIBE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKef2c.315393ac3a4f54a9fd7fe71cb140b41b.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjIFtOYolYGyRGCBo77jg6j38Co3oSuB6l From: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=170605141001 Expires: 3600 Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> NOTIFY sip:test@71.71.71.71 SIP/2.0 CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Contact: <sip:71.71.71.71:5060;transport=udp> Subscription-State: active Event: message-summary Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:test@71.71.71.71 SIP/2.0 404 Not here CSeq: 100 NOTIFY Via: SIP/2.0/UDP 71.71.71.71:5060;rport=5060 From: "57778" <sip:test@71.71.71.71>;tag=170605141001 Call-ID: RvFl7GSaJwV6YVHkewSAJ4NjqH239DqB To: "57778" <sip:test@71.71.71.71>;tag=hkjR9-fCrHBhiGYK1xpZGvZmEqjx6bIC Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 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: pj_arubaslim-16/r2 Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 23382 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjcJJWEomCJmbTkvY8iXMdrmQ935aS1h1g From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.mydomain.co", nonce="140301391017101016806051024224" Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> ACK sip:919000000002@71.71.71.71 SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKb76c.aa93d0b16ced02bdc00227054365c825.0 Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23382 ACK Content-Length: 0 INVITE sip:919000000002@71.71.71.71 SIP/2.0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71> Contact: "57778" <sip:test@192.168.1.4:58931;ob;alias=61.61.61.61~58931~1> Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 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: pj_arubaslim-16/r2 Proxy-Authorization: Digest username="test", realm="sip.mydomain.co", nonce="140301391017101016806051024224", uri="sip:919000000002@71.71.71.71", response="4edf5e2efd1e8e03bdbe4a693fdfee4c" Content-Type: application/sdp Content-Length: 356 P-hint: outbound v=0 o=- 3612003234 3612003234 IN IP4 81.81.81.81 s=pjmedia c=IN IP4 192.168.1.4 t=0 0 m=audio 43314 RTP/AVP 3 18 0 8 101 c=IN IP4 81.81.81.81 a=rtcp:43315 a=sendrecv a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 23383 INVITE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.caf535d51d5f83fc6b4139b411ebc727.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;branch=z9hG4bKPjmFPFtuAR124gIuWOHue6pEiRcHdc5wkC From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Type: application/sdp Content-Length: 252 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes> v=0 o=SipSwitch 6248 7248 IN IP4 71.71.71.71 s=VoipSIP i=Audio Session c=IN IP4 71.71.71.71 t=0 0 m=audio 6248 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv ACK sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bKc76c.e624c28002d41f48ae76813b26e516f3.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJ8XsuToNj.92GHJLiuK-JZ5T5PpfTbhr Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23383 ACK Content-Length: 0 BYE sip:71.71.71.71:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU Max-Forwards: 69 From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 CSeq: 23384 BYE User-Agent: pj_arubaslim-16/r2 Content-Length: 0 SIP/2.0 200 OK CSeq: 23384 BYE Via: SIP/2.0/UDP 81.81.81.81:7878 ;branch=z9hG4bK976c.6ca2249268e6f9592a6948becb29c474.0 Via: SIP/2.0/UDP 192.168.1.4:58931 ;received=61.61.61.61;rport=58931;branch=z9hG4bKPjJy-pnGAhtqwqV5uRixN-mvDWOTPLupLU From: "57778" <sip:test@71.71.71.71>;tag=D3R8zNsNKNyY-eyMmHPVV9SOXI200ESJ Call-ID: ROQWQKmcBUN2H8vrrKSAsRe7AUgdHR81 To: <sip:919000000002@71.71.71.71>;tag=1706051410107236868006249 Contact: <sip:71.71.71.71:5060;transport=udp> Content-Length: 0 Record-Route: <sip:81.81.81.81:7878;lr=on;nat=yes>

1 0

Trigger m_dump() with REGISTER
by Marc M. 18 Jun '14

18 Jun '14

Hi, I have a basic MSILO setup up and running. For the moment every time the client sends a REGISTER, m_dump() is called. I would like to achieve the following: The clients are registering with either 200 or 300 sec expiration time. I would like to dump the messages only when the client is registering for 300 secs. Only a REGISTER with an expiration time of 300 should trigger m_dump(). Can you help me with this? Any help or idea is appreciated. thanks Marc

1 0

Re: [SR-Users] SIP Trunk
by Keith 18 Jun '14

18 Jun '14

Hi, I have a carrier who requires SIP authentication on calls, can anyone point me in the right direction? Normally in Asterisk or Freeswitch you can authenticate a peer, but can't see how to do this in Kamailio. Thanks Keith

2 1

Use of asterisk * in Kamailio routing
by Joel White 18 Jun '14

18 Jun '14

I am having an issue using an asterisk * in the kamailio routing. How would I implement this? Thank you in advance

3 6

JSON RPC Usage in Kamailio
by Jaytirth S Khairatkar 18 Jun '14

18 Jun '14

Hello, Can anyone give me a link or document which explains how to use the JSON and JSON-RPC module in Kamailio SER? I'll greatly appreciate it. Thanks and regards Jaytirth *Disclaimer* ------------------------------------------------------------ Notice: This e-mail and / or attachments (communication) may contain information that is confidential and proprietary to The Tata Power Company Limited, Strategic Engineering Division (Tata Power SED). If you are not the intended recipient of this communication, please do not use or disseminate the information contained therein, notify the sender by e-mail or telephone and delete the communication (permanently) from your system. Any information in this communication that does not relate to the official business of Tata Power SED shall be understood as neither given nor endorsed by it. Please note that e-mails are susceptible to change and Tata Power SED shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt. This communication is not guaranteed to be free from computer viruses and it is recommended that you check for all viruses before download. Print this communication only if absolutely essential. Thank you for your co-operation. ------------------------------------------------------------

1 0

s-cscf - limit of entries for 1 contact
by Ciprus, Daniel 17 Jun '14

17 Jun '14

All, is any of you aware about limitation how many reg entries can be stored in s-cscf for 1 contact ? From what I'm observing here : s-cscf can have up to 12 reg entries for 1 contact and then other entries are silently discarded. Use case: - crash of app happens 12 times - app retries to register - s-cscf sends back 200 OK with 13 contacts - another app registers and tries to send SIP MESSAGE to 1st app Result: message never gets delivered and tcpdump shows 12 messages sent out to p-cscf but not 13th one which is actually live session between client and proxy. thanks Dan ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 2

Install SEMS
by Agiftel 17 Jun '14

17 Jun '14

Hi All, i am trying to figure out how to get SEMS and install it. If i go to iptel website, documentations and last versions give always back NOT FOUND. Is this project still maintaned?/avaible? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Install-SEMS-tp128500.html Sent from the Users mailing list archive at Nabble.com.

2 1

Re: [SR-Users] LCR from URI (CLI)
by Keith 17 Jun '14

17 Jun '14

Hi all, I have an issue where I have two carriers, one only accepts +44 then the CLI and the other only accepts the number with the leading zero dropped. I am using LCR to route to carriers, however I can only switch carriers by manually altering the CLI. Is there a way I can manipulate the CLI based on the destination the call is being sent to? Thanks Keith

2 1

Problems to start kamailio
by Carlos Rangel 17 Jun '14

17 Jun '14

Hello I am having problems to start kamailio 4.1.3 on a fresh centos 6.5 install with Asterisk 11.9 following the tutorial. Everything has gone well except that I cannot start kamailio. Basically I have two problems. The message that I got from the console when I try to start is [root@gt-ev-i08 kamailio-4.1.3]# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed An when I look at the error messages in var/log/messages I see the following errors. Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 839, column 18: bad command Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: syntax error Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: missing '(' or ')' ? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 6: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: '('')' expected (function call) Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 16: bad command: missing ';'? Jun 11 18:29:21 gt-ev-i08 kamailio: : <core> [cfg.y:3411]: yyerror_at(): parse error in config file /usr/local/etc/kamailio//kamailio.cfg, line 856, column 18: bad command I did not touch that section of the kamailio.cfg file. I am really lost now. Hopefully someone can help. Thanks Carlos

2 2

Unable to make calls within the extensions
by Chandramouli P 17 Jun '14

17 Jun '14

Hi, I am new to Kamailio. I started RTPProxy using "rtpproxy -A 54.85.12.15 -F -l 10.0.0.122 -s udp:localhost:7722" command and see that my sip phones are registered with Kamailio. I am able to see using "kamctl ul show" command. But, I am unable to establish call between my registered extension through Asterisk using Kamailio. I see that calls are hitting my Asterisk server when I call from extension. I am not getting any audio and the other extension is also not at all ringing. I could not able to figure out where I am doing the mistake? I am not sure whether the mistake is in Kamailio or Asterisk. In my Asterisk server, I am seeing the correct configuration using "odbcinst -q -d" and "isql -v MySQL-asterisk chandra test123456" and "odbc show (At CLI>)". Please find the below environment: Operating System: Ubuntu 14.04 Server (64-Bit) Kamailio: 4.0 Asterisk: 11.10 Database: MySQL (UNIX ODBC) Environment: Amazon EC2 Follwed Links: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://pastebin.com/VPpfErYn Daniel's Patched RTPProxy Installation (Please note that My Asterisk and Kamailio servers are behind NAT in Amazon EC2) Below is my configuration: *http://pastebin.com/fr8m9gr7 <http://pastebin.com/fr8m9gr7>* *Note:* I inserted the records in to the respective tables with "cmp" as context. When I call from 100 to 500, I am not getting any sound and another extension is not ringing and getting the below messages at Asterisk CLI: *http://pastebin.com/y4tXXrnF <http://pastebin.com/y4tXXrnF>* Any update would be appreciated. Thanks in advance. Regards, Chandra.

3 5

Problem with CANCEL handling
by Mordechay Kaganer 17 Jun '14

17 Jun '14

B.H. Hello, all! I'm trying to set up kamailio as a simple load balancer proxy between PSTN gateway and a cluster of asterisk servers - rather common use case IMHO. I'm completely new to kamailio. I have started with the default kamailio.cfg that comes with Ubuntu package and added the dispatcher module as described in the docs ( http://www.kamailio.org/docs/modules/4.0.x/modules/dispatcher.html). At the end, i got this up and running, but with one exception: if incoming call is not answered by the asterisk end (i.e. it is in ringing state), and then the gateway sends CANCEL request, kamailio is unable to recognize the transaction and is not forwarding the request to asterisk. The result is: the call is left "dangling" in the PBX and is not properly hang up. When the gateway sends CANCEL to kamailio, i get the following log messages: 4(25442) DEBUG: tm [t_lookup.c:769]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 31287 4(25442) DEBUG: tm [t_lookup.c:492]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(25442) DEBUG: tm [t_lookup.c:859]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: no CANCEL matching found! 4(25442) DEBUG: tm [t_lookup.c:861]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed 4(25442) DEBUG: tm [tm.c:1039]: w_t_lookup_cancel(): lookup_original: t_lookupOriginalT returned: (nil) 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=485 a=25 n=xdbg 4(25442) DEBUG: <script>: is_method(CANCEL): t_check_trans()==false 4(25442) ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=487 a=2 n=exit The relevant script snippet is standard from the distro package: # CANCEL processing if (is_method("CANCEL")) { xdbg("is_method(CANCEL): entry"); if (t_check_trans()) { xdbg("is_method(CANCEL): t_check_trans()==true"); route(RELAY); } else { xdbg("is_method(CANCEL): t_check_trans()==false"); } exit; } How can i fix this? Thanks! -- משיח NOW! Moshiach is coming very soon, prepare yourself! יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!

1 2

Re: [SR-Users] sip aliases
by Anna Schuster 17 Jun '14

17 Jun '14

Hi Quick question: If I create different sip aliases under my account, is any way I can see under Missed Called section on serweb.iptel.org or any elsewhere on which alias the call came on? Thank you, Anna On Sat, Jun 7, 2014 at 3:11 PM, <registrar(a)iptel.org> wrote: > > Thank you for registering with iptel.org. > > We are reserving the following SIP address for you: sip:mielul@iptel.org > > To finalize your registration please check the following URL within 24 > hours: > http://serweb.iptel.org/user/reg/confirmation.php?nr=0eb4729fad0f99434d205f… > (If you confirm later you will have to re-register.) > > Here are the details of your account: > --------------------------------------------------------------------- > Username: mielul(a)iptel.org > Password: q1q1q1q1 > Email: annasc74(a)gmail.com > Full Name: Anna M > Sip address: sip:mielul@iptel.org > (You can always retrieve these via the "Forgot Password" link on the > signup page) > > > > -- This message is automatically generated by SerWeb. >

2 1

Re: [SR-Users] exec_avp problem
by Ansuman Manna - Asergis Telecom 17 Jun '14

17 Jun '14

Hi, I am using the exec_avp command as followed: exec_avp("/usr/bin/mwi/test.sh $fU"); $fU taking command line arguments. Here is the shell script content: #!/bin/bash -e echo $1 logger done! $1 logger done mkdir /usr/bin/mwi/$1 || { logger 'touch failed' $? $errorMessage ; exit 1; } logger is working fine but.mkdir and other system commands are not executing. I cant figureout my misses Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

2 1

How to use unixodbc/freetds for kamailio backend
by Tim Chubb 17 Jun '14

17 Jun '14

Hi Im just getting started with kamailio with a view of evaluating it for production use. Is it possible to do the following: 1) I would like to use ODBC/FreeTDS to connect to an existing ms sql server cluster and use it for all DB related activity relating to kamailio, is this possible as we would be running kamailio in a HA configuration in production and I really don't want to have to support another DB cluster? 2) Looking at the kamdbctl scripts it seems that currently every popular db but sql server is supported which is a bit of a shame, however am I right in saying that if I was to port the sql files to SQL server, and implemented a script similar to kamdbctl.pgsql to wrap around the freetds TSQL executable that would be all that I would need to do add support for SQL server? 3) Is there any particular reason why there isn't an ODBC option for the kamdbctl script? Cheers.

2 1

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users