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37555 discussions

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.

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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

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