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Modules directory missing?
by Skyler 21 May '11

21 May '11

Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

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Re: [SR-Users] ACK to 486 is not being proxied
by Carl Wagner 21 May '11

21 May '11

Daniel, Sorry, I had meant to send it to the list. You are correct about missing messages. The ACK is being sent, but it is sent to it's self (loop condition) over the 'lo' device. So basically its a design flaw on my part of the kamailio.cfg file. My current config works for initial messages from the provider and from subscribers. But for initial messages from the Asterisk boxes, I have Asterisk send to the Kamailio box and then have Kamailio look at the message and then route it with a forward(x.x.x.x). While that works for successful calls it is what is breaking the ACK to the 486, and I think it is also breaking Cancel. Do you know of a good example config file where: 1) SIP message from provider hits Kamailio and is routed to one of several Asterisk boxes (load balancing, using ds_select) 2) Messages from a subscriber will be routed to one of the Asterisk boxes (load balancing, using ds_select) 3) Messages from an Asterisk box will be routed to the provider OR a subscribers SIP device based on a fixed string in the R-URI Or you can laugh at what I have: (constructive criticism is welcome from anyone!) (I had to trim this message as it was exceeding the 40K limit) #!KAMAILIO # ####### Defined Values ######### # - 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 ####### Global Parameters ######### # debug=4 # log_stderror=yes debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* 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:2.2.2.164:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 dns=no rev_dns=no open_files_limit=4096 disable_core_dump=yes disable_tcp=yes check_via=0 reply_to_via=0 # we may be able to use the above with force_rport ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" # loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "dispatcher.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) 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) # ----- 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) # the line below puts extra stuf at the end of the log line modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ # ----- usrloc params ----- /* enable DB persistency for location entries */ # appears to use table 'location' and this table has data in it modparam("usrloc", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # ----- auth_db params ----- # appears to use table 'subscriber' and this table has data in it modparam("auth_db", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", 1) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("avpops","db_url","mysql://kamailio:password@my.db.com/AppServer") modparam("avpops","avp_table","User") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # initial checks (check for bad/malformed/unsupported messages) route(REQINIT); # if it is not a OPTIONS message - log it if (!is_method("OPTIONS")) { xlog("L_INFO", "======== processing new message\n"); xlog("L_INFO", "MAIN: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); } # handle requests within SIP dialogs (active ongoing call - has TO: tag) does not return route(WITHINDLG); ### only initial requests (no To: tag) from here on # CANCEL processing if (is_method("CANCEL")) { # - this needs some work and cancels are not being replied to or forwared # cancle is a hop-by-hop message (send a 200 OK then forward it on) # t_check_trans = if cancel, this is supposed to be true if the Invite transaction exits # see: http://www.kamailio.org/docs/modules/1.1.x/tm.html#AEN460 if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); t_relay(); exit; } xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is FALSE\n"); # forward it on (not sure why we are getting here with valid calls that need to be canceled) # this may be bad as somone might be able to "cancel storm" us?? t_relay(); exit; } # this was added to handle OPTIONS messages route(HANDLEOPTIONS); # I 'think' this absorbs retransmissions of Invites t_check_trans(); if (!is_method("REGISTER")) { if ($rU==$null) { # request with no Username in RURI xlog("L_INFO", "Error: main route: request with no Username in RURI\n"); sl_send_reply("484","Address Incomplete"); exit; } } # set up a trigger for failures of INVITEs if (is_method("INVITE")) { xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); t_on_failure("FAIL_ONE"); } # at this point we need to determine how to route the call # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if the call is from a subscriber, it has to go to an AST box (sub can't dial directly out) # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # log all invites to syslog # if (is_method=="INVITE")) # { # xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); # }; # handle the easy cases first # from PROVIDER if ($src_ip=="3.3.3.40") { route(FROMPROVIDER); exit; # just to be sure } # if from an AST box (will determine wether to send it to PROVIDER or a subscriber) if ($src_ip=~"2\.2\.2\.1") # regexp match { route(FROMAST); exit; # just to be sure } force_rport(); # attemp to make kamailio use the emhemeral port of the sender #=============== Only subscribers left from here down # handle registrations (from a sip client) route(REGISTER); # authentication for everyone route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route(FROMSUBSCRIBER); # route(RELAY); xlog("L_INFO", "ERROR - no route chosen \n"); } ################################################################### # Per SIP request initial checks route[REQINIT] { # xlog("L_INFO", " REQINIT: Initial Checks\n"); if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", " REQINIT: Too Many Hops (Request: [$rm])\n"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_WARN", " REQINIT: Malformed SIP message from $si:$sp (Request: [$rm]) \n"); exit; } # xlog("L_INFO", " leaving [REQINIT] \n"); # unsupported messages here if (is_method("SUBSCRIBE")) { # probably want to comment the next line out after it is working xlog("L_WARN", "REQINIT: Subscribe not supported: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); exit; } # if no errors, just fall through and return to caller } ################################################################### # Handle requests within SIP dialogs (request has a TO: Tag) route[WITHINDLG] { # xlog("L_INFO", " WITHINDLG: starting up (WithInDialog)\n"); if (has_totag()) { xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); xlog("L_INFO", " WITHINDLG: has TO: tag\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is true\n"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } xlog("L_INFO", " WITHINDLG: sending to relay\n"); route(RELAY); } else # not loose_route { xlog("L_INFO", " WITHINDLG: has TO: tag AND louse_route is NOT true\n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK\n"); if ( t_check_trans() ) # see if a message is related to a transaction { # Reference: http://www.kamailio.org/docs/modules/stable/modules/tm.html#t_check_trans xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=true\n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE\n"); xlog("L_INFO", " WITHINDLG: the ACK to a 486 was not being processed so I am adding t_relay here\n"); # $var(a) = t_relay(); # $var(a) = forward(); t_relay(); xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] exiting)\n"); exit; } } # not an ACK - as both branches aove exit xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method not = ACK \n"); sl_send_reply("404","Not here"); } exit; } # xlog("L_INFO", " leaving [WITHINDLG] (WithInDialog)\n"); # if msg has to: tag it should be processed above and then exit, else just return for further processing } ################################################################### # By CEW 2Feb2011 route[REGISTER] { if (is_method("REGISTER")) { # NOTE: need to review: http://www.kamailio.org/docs/scripting.html on how they handle registration # at this point all that should be left are sip clients - they are coming over the open internet so authenticate them! # xlog("L_INFO", " REGISTER: processing register from sip client, fu=$fu \n"); # authenticate the REGISTER requests (uncomment to enable auth) # www_authenticate will send a 401 Unauthorized (and will contain WWW-Authenticate in the reply) # proxy_authenticate will send a 407 Proxy Authentication Required. (and will contain Proxy-Authenticate in the reply) # uses the kamailio->subsriber DB table # if (!proxy_authenticate("2.2.2.164", "subscriber")) if (!www_authenticate("2.2.2.164", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error # -2 = invalid password # -3 = invalid user ( or domain is wrong) if ($var(a) == -2) { xlog ("L_INFO", " REGISTER ERROR: bad password, (ReturnCode = $var(a) exiting)\n"); exit; } if ($var(a) == -3) { xlog ("L_INFO", " REGISTER ERROR: invalid user, (ReturnCode = $var(a) exiting)\n"); exit; } xlog("L_INFO", " REGISTER: issuing a challenge, 401 (and then exit script)\n"); # send a "401 Unauthorized", with realm=2.2.2.164, caller should resend invite with auth info www_challenge("2.2.2.164", "1"); # send a 401 # proxy_challenge("2.2.2.164", "1"); # send a 407 # xlog("L_INFO", " REGISTER: exiting\n"); exit; } xlog("L_INFO", " REGISTER: Match found in database for register\n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $tU = reference to username in URI of 'To' header if ($au!=$tU) { xlog("L_INFO", " REGISTER: sending 403 Forbidden auth ID\n"); sl_send_reply("403","Forbidden auth ID"); exit; } # now store it in the DB and then exit if (!save("location")) { xlog("L_INFO", " REGISTER: Error saving location, sending SIP error to other end\n"); sl_reply_error(); } exit; } } ################################################################### # Authentication route route[AUTH] { xlog("L_INFO", " entering route[AUTH] Authentication route\n"); # authenticate if from local subscriber # I think this should be true for LOCAL sip callers (in production there shouldn't be any of these) if (from_uri==myself) { xlog("L_INFO", " AUTH: from_uri = myself ($fu)\n"); if (!proxy_authenticate("$fd", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error - like no valid auth # -2 = invalid password # -3 = invalid user (or domain is wrong) if ($var(a) <= -2) { xlog ("L_INFO", " AUTH: ERROR: Location lookup, ReturnCode = $var(a) exiting\n"); exit; } # xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (407), from_uri=$fu \n"); xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (401), from_uri=$fu \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); exit; } xlog("L_INFO", " AUTH: proxy_autorization OK! \n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden, sending 403\n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Authorization accepted! Consuming credentials\n"); consume_credentials(); # caller authenticated } else { # !! NOTE: this branch has not been tested as I cant simulate it in my lab!!! # a lot of it was just made up so it does need a lot of testing. # subscriber from the Internet # this is where all our sip clients should hit xlog("L_INFO", " AUTH: EXTERNAL subscriber, need to authenticate from_uri $fu\n"); # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { xlog("L_INFO", " AUTH: NOT RELAYING!, (uri!=myself) sending 403\n"); sl_send_reply("403","Not relaying"); exit; } # not sure if we need to do a proxy_authenticate or www_authenticate here - need to test in production if (!proxy_authenticate("$fd", "subscriber")) { xlog("L_INFO", " AUTH: sending proxy_challenge, from_uri=$fu, exiting \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); xlog("L_INFO", " AUTH: sent proxy_challenge, exiting\n"); exit; } # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden auth ID, from_uri=$fu, exiting \n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Consuming credentials\n"); consume_credentials(); } xlog("L_INFO", " leaving route[AUTH] Authentication route\n"); return; } ################################################################### # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if (uri=="3.3.3.40") route[FROMPROVIDER] { # xlog("L_INFO", " FROMPROVIDER: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMPROVIDER: REGISTER from PROVIDER, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); }; # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }; # if a conference call, send it to ast1 # ds_select_dst(set, alg) the method selects a destination from address set # set - first column from dest file (dispacher.list) # alg 0 = hash over callid if (uri=~"12223334444") { xlog("L_INFO", " FROMPROVIDER: Conference call - sending to ast1 \n"); ds_select_dst("3", "0"); forward(); exit; }; # avp_check(name, op_value) check the value of the avp against an operator and value # op_value = operator '/' value ['/'flags] # $fu = # $ru = SIP Request's URI # if the call came from ast0 (2.2.2.167) via PROVIDER route it to ast1 or we will have a loop condition # $ci = the call ID # AVP = attribute-value-pair if (avp_check("$ci","re/.*(a)2.2.2.167.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast0, sending call to ast1 [$ru]\n"); ds_select_dst("2", "0"); forward(); exit; }; # if the call came from ast1 (2.2.2.168) via PROVIDER route it to ast0 or we will have a loop condition if (avp_check("$ci","re/.*(a)2.2.2.168.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast1, sending call to ast0 [$ru]\n"); ds_select_dst("3", "0"); forward(); exit; }; # Extract the telephone number from the To-URI $avp(i:999)=$tU; # insted of the next line, should we just do a regexp to remove all non-numeric? avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "+" from the variable, if any # xlog("L_INFO", " FROMPROVIDER: about to do \"SELECT AppName FROM Application WHERE TelephoneNumber = $avp(i:999) \" \n"); avp_db_query("SELECT AppName FROM Application WHERE TelephoneNumber = '$avp(i:999)'","$avp(i:888)"); # xlog("L_INFO", " FROMPROVIDER: Query result =[$avp(i:888)] \n"); # see if we got anything back from the DB, if we did it would look like "product" or some future varient if(!is_avp_set("$avp(i:888)")) # if avp 888 is not set, dump the call { xlog ("L_INFO", " FROMPROVIDER: Bad number. Sending 404. method=[$rm] ruri=[$ru] to=[$tu] from=[$fu]\n"); # I don't think the next line is needed as it appears this table is never read (just written to) avp_db_query("INSERT IGNORE INTO InvalidTN VALUES ('$avp(i:999)')"); sl_send_reply("404", "Not Found"); # do we really want to do this? Or shoud we just silently drop it? exit; }; # this is where we do the main load balancing, ast0 or ast1 xlog ("L_INFO", " FROMPROVIDER: Normal Round-Robin call, routing to [$ru], from [$fu]\n"); # ds_select_dst("1", "0"); ds_select_dst("3", "0"); forward(); # Note: forward does return, so we could do post call processing here exit; } ################################################################### # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # if (uri=~"2\.2\.2\.1") from our ast box route[FROMAST] { xlog("L_INFO", " FROMAST: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMAST: REGISTER from AST, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # this block is probalby not needed, except possilby for logging $avp(i:999)=$tU; # Extract the telephone number from the To-URI avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "sip:+" from the variable xlog("L_INFO", " FROMAST: avp(999) = $avp(i:999)\n"); # at this point we need to see if the number contains a "SIP_". If so route it to the subscriber, else route it to L3 xlog("L_INFO", " FROMAST: tU=$tU\n"); if ($tU=~"SIP_") { xlog("L_INFO", " FROMAST: contains \"SIP_\" so routing to subscriber\n"); route(TOSUBSCRIBER); exit; # should be handled in TOSUBSCRIBER, but just to make sure we bail here and don't continue } xlog("L_INFO", " FROMAST: does not contain SIP_ so routing to PROVIDER\n"); # need to rewrite the RURI and To URI $td="3.3.3.40:5060"; $rd="3.3.3.40:5060"; forward("3.3.3.40"); # send it on to PROVIDER } ################################################################### # if the call is from a subscriber, it has to go to an AST box (subscribers can't dial directly out) route[FROMSUBSCRIBER] { xlog("L_INFO", " FROMSUBSCRIBER: starting up \n"); # at this point we should have already been authenticated by route[AUTH] so just route it to an ast box # this is where we do the main load balancing, ast0 or ast1 (round robin) ds_select_dst("3", "0"); forward(); xlog("L_INFO", " leaving route[FROMSUBSCRIBER] \n"); exit; # just in case } ################################################################### # The call needs to be routed to a scubscriber route[TOSUBSCRIBER] { xlog("L_INFO", " TOSUBSCRIBER: starting up \n"); # this is NOT tested yet! # do a DB lookup on kamailio->location to see where to route (should only be for SIP-to-Subscribers) if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); xlog("L_INFO", " TOSUBSCRIBER: Subscriber not Found, sending 404 \n"); t_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", " TOSUBSCRIBER: Method Not allowed, sending 405 \n"); sl_send_reply("405", "Method Not Allowed"); exit; } } xlog("L_INFO", " TOSUBSCRIBER: subscriber is authorized! \n"); # not sure about any of this that follows # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { xlog("L_INFO", " TOSUBSCRIBER: setting flag FLT_ACCMISSED \n"); setflag(FLT_ACCMISSED); } xlog("L_INFO", " TOSUBSCRIBER: attempting to do: t_relay \n"); if (!t_relay()) { xlog("L_INFO", " TOSUBSCRIBER: ERROR in t_relay \n"); sl_reply_error(); }; xlog("L_INFO", " TOSUBSCRIBER: message proxied, now exiting \n"); exit; } ################################################################### # Sample failure route # One source for this block: http://pastebin.com/NMZiBuNm # A source: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#failure_route # 'failure_route' is executed only by TM module after it was armed via t_on_failure(“failure_route_index”) failure_route[FAIL_ONE] { xlog("L_INFO", " FAIL_ONE: starting up Sample failure route (looks for canceled)\n"); if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { xlog("L_INFO","[1] ($ru) \n"); sl_send_reply("ACK", "Busy"); # send an ACK back to sender t_relay(); # forward it on } # this may do what I want ?? # if(is_method("INVITE")) { xlog("L_INFO", " leaving route[FAIL_ONE] Sample failure route \n"); } ################################################################### route[RELAY] { xlog("L_INFO", " RELAY starting up\n"); /* example how to enable some additional event routes */ if (is_method("INVITE")) { xlog("L_INFO", " RELAY: Error: INVITE sending sl_on_failure\n"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { xlog("L_INFO", " RELAY: Error sending sl_reply_error\n"); sl_reply_error(); } xlog("L_INFO", " leaving [RELAY]\n"); exit; } ################################################################### # Presence server route route[PRESENCE] { xlog("L_INFO", " PRESENCE: starting up \n"); if(!is_method("PUBLISH|SUBSCRIBE")) return; # if presence enabled, this part will not be executed # if (is_method("PUBLISH") || $rU==$null) # { xlog("L_INFO", " PRESENCE: PUBLISH and SUBSCRIBE not supported sending 404\n"); sl_send_reply("404", "Not here"); exit; # } # xlog("L_INFO", "leaving route[PRESENCE] Presence server route\n"); # return; } ################################################################### # Caller NAT detection route route[NAT] { xlog("L_INFO", " NAT: NAT detection route (currently does nothing)\n"); xlog("L_INFO", " leaving route[NAT] NAT detection route \n"); return; } ################################################################### # RTPProxy control route[RTPPROXY] { xlog("L_INFO", " RTPPROXY: RTPProxy control(currently does nothing)\n"); xlog("L_INFO", " leaving route[RTPPROXY] RTPProxy control\n"); return; } ################################################################### # Routing to foreign domains route[SIPOUT] { xlog("L_INFO", " SIPOUT: Routing to foreign domains\n"); if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } xlog("L_INFO", " leaving route[SIPOUT] Routing to foreign domains\n"); } ################################################################### # PSTN GW routing route[PSTN] { xlog("L_INFO", " PSTN: PSTN GW routing (currently does nothing) \n"); xlog("L_INFO", " leaving route[PSTN] PSTN GW routing\n"); return; } ################################################################### # PROVIDER or our AST boxes may send an options message, they are our friends so just 200 OK them # By CEW 2Feb2011 route[HANDLEOPTIONS] { if (is_method("OPTIONS")) { # xlog("L_INFO", " HANDLEOPTIONS: found options message \n"); # if from AST if ($src_ip=~"2\.2\.2\.1") { # xlog("L_INFO", " HANDLEOPTIONS from our AST, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if from PROVIDER if ($src_ip=~"2\.2\.2\.40") { xlog("L_INFO", " HANDLEOPTIONS from PROVIDER, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if we made it here, just throw the message away # this assumes that none of our sip clients will send us Options messages! exit; }; }

1 0

Via branch parameter in end2end ACK
by Juha Heinanen 20 May '11

20 May '11

Daniel-Constantin Mierla writes: > if I understood right, it is about 200 ACK which are forwarded > stateless. If yes, then should be only in forward.c, the condition at > line 545, where the assignment can be changed in a memcpy. for example like this? if so, i can go and make the commit. -- juha if (syn_branch ) { memcpy*(*msg->add_to_branch_s, "z9hG4bKcydzigwkX", 16); msg->add_to_branch_len=16; } else {

2 1

IPv6
by Jon Farmer 20 May '11

20 May '11

Hi How do I represent the following if the address was IPv6? if(src_ip ==127.0.0.1) If I specify the IPv6 address with our without [] I get WARNING:core:fix_socket_list: could not rev. resolve <ipv6_addr> Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

xHTTP module.
by David J. 20 May '11

20 May '11

I was wondering if I can make an HTTP request to Kamailio; and then have kamailio do a lookup based on passed parameters to connect callers. I am trying to make a click-to-dial type application; I was looking at the HTTP server inside kamailio; It was interesting to me to try to use this as an alternative to php-asterisk; (I know kamailio is not a full blown application server;) If anyone could help me with this I would appreciate it; Thanks in advance. David.

2 4

Re: [SR-Users] ACK to 486 is not being proxied
by Daniel-Constantin Mierla 20 May '11

20 May '11

Please do not write private emails, keep the mailing list cc-ed. Can you get the output of: ngrep -d any -qt -W byline port 5040 on kamailio server for such a call? Include all messages, since you may miss something in your filtering (like you did below), of course you can mask the ip addresses. Also, be sure you do t_relay() to forward the invite from kamailio on. Cheers, Daniel On 5/20/11 3:47 PM, Carl Wagner wrote: > Daniel, > > With a non-2xx reply being hop-by-hop, shouldn't Kamailio receive the > 486, send the ACK to the sender, and then proxy the 486? > > I am not seeing the 486 message in the Kamailio log so I assume that > it automatically handles it without hitting the kamailio.cfg script? > Or do I need to set up a failure_route to catch the 486 and send the > ACK there? > > Please see the log below. > > Thanks, > Carl > > > > # Asterisk to Kamailio > U 2011/05/19 14:13:01.467157 3.4.5.167:5060 -> 3.4.5.164:5060 > INVITE sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 70. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > # Kamailio to provider > U 2011/05/19 14:13:01.468627 3.4.5.164:5060 -> 2.3.4.40:5060 > INVITE sip:+13031112222@2.3.4.40:5060 SIP/2.0. > Record-Route: <sip:3.4.5.164;lr=on>. > Via: SIP/2.0/UDP 3.4.5.164;branch=0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 69. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2011/05/19 14:13:01.511775 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > > U 2011/05/19 14:13:01.513162 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > # provider says they are busy > U 2011/05/19 14:13:03.239462 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > # > # should Kamailio send the ACK here? > # > > # Kamailio proxies the 486 to Asterisk > U 2011/05/19 14:13:03.240820 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > > # ACK from Asterisk - should be swallowed by Kamailio > U 2011/05/19 14:13:03.243870 3.4.5.167:5060 -> 3.4.5.164:5060 > ACK sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>;tag=VPST506071629352. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 ACK. > User-Agent: Asterisk. > Max-Forwards: 70. > Content-Length: 0. > . > > > ==== a bunch of retransmitted 486's chopped ============= > > > > > > > > On 05/19/2011 02:30 AM, Daniel-Constantin Mierla wrote: >> Hello, >> >> the ACK for non-2xx replies is hop-by-hop, meaning that kamailio will >> generate one when the reply is received, will send back the reply and >> when the ack comes from upstream, it will absorb it. >> >> In this case it seems that ack is not matching any invite transaction >> and thus is not known where to send it. >> >> Can you send the invite, reply and the ack requests taken with ngrep >> for such case? Maybe is something broken in the content of the message. >> >> Cheers, >> Daniel >> >> On 5/18/11 11:23 PM, Carl Wagner wrote: >>> Hi, >>> >>> I have Kamailio set up to act as a proxy and load balancer. Most >>> things are working correctly but for some reason I can't get >>> Kamailio to proxy an ACK to a 486 Busy. >>> I assume that it is something wrong that I have done in the >>> kamailio.cfg file but I am not seeing it. >>> >>> Does the t_check_trans() return false because the call state is Busy? >>> >>> Please let me know if you need other logs or information. >>> >>> Thanks in advance, >>> Carl >>> >>> >>> >>> ================== Call flow: (from ngrep) ====================== >>> >>> Asterisk Kamailio Provider >>> INVITE---------> >>> INVITE---------> >>> <----------100 Trying >>> <---------100 Trying >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> ====== Kamailio does not proxy the ACK here ======= >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> <----------486 Busy >>> <---------486 Busy >>> >>> >>> === kamailio.cfg snippet >>> ... >>> route >>> { >>> route(REQINIT); # remove malformed messages >>> >>> # handle requests within SIP dialogs >>> route(WITHINDLG); >>> ... >>> } >>> >>> ################################################################### >>> # Handle requests within SIP dialogs (request has a TO: Tag) >>> route[WITHINDLG] >>> { >>> if (has_totag()) >>> { >>> xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from >>> [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); >>> if (loose_route()) >>> { >>> ... >>> } >>> else # not loose_route >>> { >>> if ( is_method("ACK") ) >>> { >>> if ( t_check_trans() ) # see if a message is related to >>> a transaction >>> { >>> ... >>> } >>> else >>> { >>> # ACK without matching transaction ... ignore and >>> discard >>> xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true >>> and is_method = ACK and t_check_trans=FALSE\n"); >>> >>> # not forwarded here!!! Tried both t_relay and forward. >>> # $var(a) = t_relay(); >>> $var(a) = forward(); >>> xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] >>> exiting)\n"); >>> >>> exit; >>> } >>> } >>> sl_send_reply("404","Not here"); >>> } >>> exit; >>> } >>> } >>> >>> ============= End of kamailio.cfg ================ >>> >>> >>> ============= /var/log/messages - snippit of the message ========== >>> >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> ======== processing new message >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> MAIN: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6]) CSeq=[102] >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6], CSeq=[102]) >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = >>> ACK and t_check_trans=FALSE >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: the ACK to a 486 was not being processed so I am >>> adding t_relay here >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: (ReturnCode = [1] exiting) >>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

nonce reuse parameter
by Bruno Bresciani 20 May '11

20 May '11

Hi, on kamailio 3.1.2 the one_time_nonce parameter of the auth module is similar to the nonce_reuse parameter present on kamailio 1.5.0? I want to disable the check of the nonce reuse in kamailio 3.1.2... regards

2 2

Reactivating a failed dispatcher destination
by Lee Archer 20 May '11

20 May '11

Hi all I wonder if someone can help by pointing me in the right direction. I have dispatcher set to ping and set any failed gateways to probing but how do you get them out of this state when they start working again? I have run some tests and I must be missing something as active servers seem to stay in probing mode. I am using Kamailio 3.0.4. Thanks Lee

3 10

packaging 3.1.4
by Daniel-Constantin Mierla 20 May '11

20 May '11

Hello, I am thinking of packaging v3.1.4 next week on Thursday -- last minor release was more than 1.5month ago. Any comments against it? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Forward on unanswered call
by Andrew O. Zhukov 20 May '11

20 May '11

Right now I'm try to add the subj feature. Linksys/SPA2102-3.3.6 (just from the box) respond SIP/2.0 480 Temporarily not available Landline over Asterisk with E1: SIP/2.0 603 Limit exceeded Ether I so unlucky, or just forget it and do not provide it to the customers? If so, notes to developers let's take off 408 code processing from sample config. Thanks in advance for your opinion. Andrew.

2 1

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