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Problem with NAT and RTPProxy
by David Candamil Santos 14 Nov '11

14 Nov '11

Hi, I am using Kamailio 3.2.0 (x86_64/linux) and since some days ago I am trying to call from a network with NAT to outside. This is the configuration: Softphone (192.168.0.5) <--> Kamailio (192.168.0.3) <--> Router (192.168.0.1) <--> Softphone over smartphone All the router ports are opened and redirected to 192.168.0.3. I have installed rtpproxy 1.2.1-1 with the following configuration: --------------------------/etc/defaults/rtpproxy-------------------------------- # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 # Additional options that are passed to the daemon. EXTRA_OPTS="-l candamil.dyndns.org" ---------------------------------------------------------------------------------------- and works properly. This is the log message: ---------------------------------------------------------------------------------------------- Nov 12 20:09:13 condor kamailio[7001]: INFO: rtpproxy [rtpproxy.c:1415]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled ----------------------------------------------------------------------------------------------- Both softphones are "linphone". The configuration of the network softphone is the following: SIP identity: sip:1001@192.168.0.3 SIP proxy: sip:192.168.0.3 Direct connection to the Internet. In the softphone of the smartphone I set as proxy and domain " candamil.dyndns.org", the DNS address of my router IP. In this case, these are the symptoms: Both softphones can autentify correctly. When calling, both softphones ring, but when I answer, the voice signal is not received. This is the log error: ---------------------------------------------------------------------------------------------- Nov 12 20:23:14 condor kamailio[6991]: ERROR: rtpproxy [rtpproxy.c:2260]: incorrect port 0 in reply from rtp proxy ---------------------------------------------------------------------------------------------- The same happens if I set in the network softphone that it's behind NAT, and I set the router IP. The same happens if I set in the network softphone that it's behind NAT and with a STUN server (stunserver.org). In the three cases, in the softphone over smartphone, the caller is 1001(a)192.168.0.3, and in the network softphone, the caller is 1002(a)candamil.dyndns.org. If I set in the network softphone the proxy as sip:candamil.dyndns.org, the same happens. If what I do is setting the SIP identity as sip:1001@candamil.dyndns.org, when calling form inside to ouside the network, the softphone doesn't know that the call was answered, and there is the following error in the log: ---------------------------------------------------------------------------------------------------------- Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2600]: ERROR: parse_via: invalid port number <5060ranch=z9hG4bKc 50f.b4825246.0> Nov 12 20:53:00 condor kamailio[7306]: ERROR: <core> [parser/parse_via.c:2629]: ERROR: parse_via on: <SIP/2.0/UDP 192.168.0.3:5060ranch=z 9hG4bKc50f.b4825246.0;received=87.223.138.84#015#012Via: SIP/2.0/UDP 87.223.138.84:5060;rport=5060;branch=z9hG4bK1021772993#015#012From: <sip:1001@candamil.dyndns.org>;tag=783852345#015#012To: < sip:1002@candamil.dyndns.org>#015#012Call-ID: 1644787160#015#012CSeq: 21 INVITE# 015#012User-Agent: Linphone/3.4.0 (eXosip2/unknown)#015#012Content-Length: 0#015#012#015#012> ------------------------------------------------------------------------------------------------------------ If the call is from outside to inside the network, it happens the same than in the previous cases. This is the relevant kamailio configuration: -----------------------------------------kamailio.cfg---------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_NAT ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!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 ######### /* 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="candamil.dyndns.org" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- 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) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- 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", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } -------------------------------------------------------------------------------------------------------------- I hope you can help me. Thanks for your time.

2 1

Can the SBC (Kamailio 3.1.x and FreeSWITCH 1.0.6+) do Upper Registration?
by edson.gomes.leme 14 Nov '11

14 Nov '11

2 1

SnmpStats problem
by Adam Kuśmirek 14 Nov '11

14 Nov '11

Hi All I've compiled kamailio 3.2.0 with snmpstats module. When trying to start the module I received the following message: 0(28197) DEBUG: <core> [sr_module.c:515]: load_module: trying to load </usr/lib/kamailio/modules_k/ snmpstats.so> 0(28197) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader I run it on CentOS 5.7 After digging Internet, I found the solution for the problem. I had to modify Makefile in snmpstats module's directory and recompile it. I had to change BUILDAGENTLIBS env param before change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp after change: BUILDAGENTLIBS =-L$(LOCALBASE)/lib -lnetsnmpmibs -lnetsnmpagent \ -lnetsnmphelpers -lnetsnmp \ -Wl,-E -Wl,-rpath,/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -L/usr/local/lib /usr/lib/perl5/5.8.8/i386-linux-thread-multi/auto/DynaLoader/DynaLoader.a \ -L/usr/lib/perl5/5.8.8/i386-linux-thread-multi/CORE \ -lperl -lresolv -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc \ -lwrap \ -lsensors \ -lrpmdb -lrpm After that I successfully started snmpstats module. I don't know why I didn't have compilation errors before adding this linking info, but got problem when trying to run compiled module. Can you explain it ? Do you know another way to solve the problem ? Regards Adam

2 1

a2billing
by roman dmytriv 11 Nov '11

11 Nov '11

Hello folks, Any one tried using a2billing with Kamailio and Asterisk instead of Siremis? Can you share your experiens? Regards roman

3 6

tracking ACC for a 486
by Sven Knoblich 10 Nov '11

10 Nov '11

Hi, I have a short question regarding a tm callback. I want to track a local generated ACK during a cfa. scenario: A calls B with a cfa to C B response with a 486 Busy this will be confirmed with an ACK. I saw, that TMCB_ACK_NEG_IN will only be called when a new transaction is created (t_lookup.c:t_newtran). Unfortunately in this case an ACK will be send directly when a reply will be received (t_reply.c:reply_received). Any ideas? thanks in advance, Sven

2 2

segmentation fault in init_mod_child function
by Bruno Bresciani 10 Nov '11

10 Nov '11

Hi All, Kamailio 3.1.2 terminated with signal 11, segmentation fault, in the init_mod_child function at sr_module.c file. This problem occurs only when I try loading the loadbalance.so module, it was written by me. Someone Can help me to understand why this segmentation fault was generated? There are two other module written by me and this problem doesn't happen. Kamailio's trace: 9(27616) DEBUG: db_postgres [km_pg_con.c:80]: PQsetdbLogin(0x8db2878) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): sl 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): auth_db 9(27616) DEBUG: <core> [db.c:294]: connection 0x82f4108 found in pool 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): usrloc 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): mi_fifo 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): registrar 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): nathelper 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): permissions 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): loadbalance 9(27616) DEBUG: loadbalance [loadbalance_mod.c:271]: loadbalance: child started! (5) 9(27616) DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (5): route 0(27582) ALERT: <core> [main.c:741]: child process 27585 exited by a signal 11 0(27582) ALERT: <core> [main.c:744]: core was generated 0(27582) INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD 8(27615) INFO: <core> [main.c:807]: INFO: signal 15 received 8(27615) DEBUG: <core> [main.c:818]: Memory status (pkg): 8(27615) DEBUG: fm_status: fm_status (0x82aefc0): 8(27615) DEBUG: fm_status: heap size= 4194304 8(27615) DEBUG: fm_status: used= 236720, used+overhead=264064, free=3930240 8(27615) DEBUG: fm_status: max used (+overhead)= 299824 8(27615) DEBUG: fm_status: dumping free list: 8(27615) DEBUG: fm_status: hash = 1 fragments no.: 27, unused: 0 bucket size: 8 - 8 (first 8) 8(27615) DEBUG: fm_status: hash = 44 fragments no.: 1, unused: 0 bucket size: 352 - 352 (first 352) 8(27615) DEBUG: fm_status: hash = 74 fragments no.: 1, unused: 0 bucket size: 592 - 592 (first 592) 8(27615) DEBUG: fm_status: hash = 77 fragments no.: 1, unused: 0 bucket size: 616 - 616 (first 616) 8(27615) DEBUG: fm_status: hash = 81 fragments no.: 1, unused: 0 bucket size: 648 - 648 (first 648) 8(27615) DEBUG: fm_status: hash = 84 fragments no.: 1, unused: 0 bucket size: 672 - 672 (first 672) 8(27615) DEBUG: fm_status: hash = 113 fragments no.: 51, unused: 0 bucket size: 904 - 904 (first 904) 8(27615) DEBUG: fm_status: hash = 2056 fragments no.: 1, unused: 0 gdb bt full trace: Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/route.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/route.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/rtpproxy.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/rtpproxy.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/tls.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/tls.so Reading symbols from /media/sda1/local/kamailio/lib/kamailio/modules/perms_db.so...done. Loaded symbols for /home2/local/kamailio/lib/kamailio/modules/perms_db.so Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 827 sr_module.c: Arquivo ou diretório não encontrado. in sr_module.c (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 init_mod_child (m=0xa238, rank=2) at sr_module.c:827 No locals. #1 0x00000000 in ?? () No symbol table info available. (gdb) Best Regards

2 2

How to limit number of calls in Kamailio proxy
by Austin Einter 10 Nov '11

10 Nov '11

This is continuation with limiting number of calls per Kamailio proxy server. I modified kamailio.cfg as below. 1. loadmodule "dialog.so" 2. modparam("dialog", "enable_stats", 1) 3. modparam("dialog", "dlg_flag", 4) Next, as dialog module I can use either active_dialogs() or $DLG_count to find number of calls present at any given point of time. However if I use active_dialogs() I get parse error when I run kamailio proxy. Next I used $DLG_count , but always I am getting value of $DLG_count as 0. In *route* section, I added code as below. if (is_method("INVITE")) { if($DLG_count > 1) { sl_send_reply("503","RESOURCE UNAVAILABLE"); exit; } else { xlog("Dialog count = $DLG_count\n"); } } Everytime a new call starts , it prints $DLG_count as 0. Kindly let me know whats going wrong here. Regards Austin

5 15

Re: [SR-Users] Kamailio Dispatcher fails in front of FreeSwitch, help (again).
by Henrik Aagaard Sørensen 10 Nov '11

10 Nov '11

Hi Daniel. I just wanted to hear if you have had any time to look at the Kamailio Dispatcher <-> FreeSwitch problem? This is NOT to rush you. I've followed the mailing list and can see how busy you are and how much involved you are in all this, so I fully respect all the help you give to this mailing list etc. Best regards, Henrik 2011/10/20 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> > Hi Daniel. > > I've just upgraded to 3.2.0. > > This is my NEW kamailio.cfg: http://pastebin.com/XzXjf1VY > This is my dispatcher.list: http://pastebin.com/9dLS8243 > > I've also attached a ngrep where it errors with "Too many hops" with the > new version 3.2.0. > > Once again, I really appreciate all the help you give! It's remarkable > with your effort in this. > > Best regards, > Henrik > > > 2011/10/17 Daniel-Constantin Mierla <miconda(a)gmail.com> > >> I will try to look over the sip traces you sent, although it is tight >> schedule for the 3.2.0 right now -- so it may take few days. >> >> Cheers, >> Daniel >> >> >> On 10/14/11 10:07 PM, Henrik Aagaard Sørensen wrote: >> >> Yes, pbx.my-domain.com is pointing to the IP of FreeSwitch. >> sip.my-domain.com is the domain name for Kamailio. >> >> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >> >>> >>> >>> On 10/14/11 7:06 PM, Henrik Aagaard Sørensen wrote: >>> >>> My dispatcher.list currently just contains: >>> 1 pbx.my-domain.com:5060 >>> >>> Is pbx.my-domain.com pointing to the IP address of freeswitch? What is >>> the domain name for kamailio ip address? >>> >>> Daniel >>> >>> >>> >>> I will provide a ngrep trace within an hour or two. Thank you so much >>> for helping me. I really hope to get this solved. >>> >>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>> >>>> Hello, >>>> >>>> >>>> On 10/14/11 4:48 PM, Henrik Aagaard Sørensen wrote: >>>> >>>> Hi Daniel (and everyone else) again. >>>> >>>> I've just tried to add: >>>> loadmodule "path.so" >>>> modparam("path", "use_received", 1) >>>> >>>> and: >>>> add_path_received() >>>> just before the "if(!t_relay())..." in the ROUTE part. >>>> >>>> That made both devices get the error "Too many loops". >>>> >>>> what is the content of the dispatcher list file? Maybe is better to >>>> use ds_select_domain(...) and ds_next_domain() -- cannot say it since I >>>> haven't seen full sip trace, but maybe the problem is that freeswitch is >>>> sending back if the r-uri address is kamailio's address. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> >>>> Moving the add_path_received() into the part: >>>> if (nat_uac_test("19")) { if (method=="REGISTER") { >>>> fix_nated_register(); add_path_received(); >>>> made the same issues as before loading the path.so module. >>>> >>>> 2011/10/14 Henrik Aagaard Sørensen <henrikaagaardsorensen(a)gmail.com> >>>> >>>>> Hi Daniel (and everyone else). >>>>> >>>>> This is a tshark trace (is that as good as ngrep?) from Kamailio when >>>>> there are too many loops: http://pastebin.com/yRjduB0s >>>>> In this trace I got the note that 253 packets was dropped (and >>>>> therefor not in the trace). >>>>> >>>>> And this is a trace when it times out: http://pastebin.com/KT0bJzEU >>>>> >>>>> I've actually looked at the PATH module, and tried several things. And >>>>> I know this type of question is rather newbee-like, but can you help me out >>>>> with the PATH module and how to integrate it correctly? >>>>> >>>>> I really do appreciate all the help! >>>>> >>>>> >>>>> 2011/10/14 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>> >>>>>> Hello, >>>>>> >>>>>> when using the load balancer in front of the registrar, you have to >>>>>> use PATH extensions for SIP (see path module in kamailio) in order to be >>>>>> able to cope with nat. >>>>>> >>>>>> Then, what would be helpful, will be a ngrep trace taken on kamailio >>>>>> server for failed calls. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> >>>>>> On 10/14/11 4:13 PM, Henrik Aagaard Sørensen wrote: >>>>>> >>>>>> I've asked a lot of question about Kamailio Dispatcher in front of >>>>>> FreeSwitch in this mailing list. So I'm sorry that I'm still "spamming", >>>>>> but it still fails. >>>>>> >>>>>> This is my status so far. >>>>>> >>>>>> When connecting 2 devices directly to FreeSwitch everything works. >>>>>> When connecting to Kamailio, some calls fails with "Timeout" and some fails >>>>>> with "Too many loops". >>>>>> >>>>>> This is my kamailio.cfg: http://pastebin.com/KEuZmikh >>>>>> >>>>>> When connected directly to FreeSwitch where everything works, this is >>>>>> how my clients are registrered in FreeSwitch: >>>>>> http://pastebin.com/prkjkb7g >>>>>> >>>>>> When connected to Kamailio where nothing works, this is how my >>>>>> clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv >>>>>> >>>>>> 215.197.145.132 (sip.my-domain.com) is my Kamailio. >>>>>> pbx.my-domain.com is my FreeSwitch. >>>>>> 96.224.14.164 is my IP at home, where my 2 devices are from. >>>>>> >>>>>> Can anyone help me out? I'm kinda lost at this point. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla -- http://www.asipto.com >>>>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>>>> >>>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla -- http://www.asipto.com >>>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.com >>> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >

2 2

uacreg issue
by Lucas Alvarez 09 Nov '11

09 Nov '11

Hi, after following the steps of this guide: http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html, poulating the uacreg table, loading the uac module and restarting kamailio I didn't see any REGISTER packet sent out to the sip provider, is something missing? How can I know if the values from the table are being loaded? I would appreciate any kind of help. Regards, Lucas Alvarez

1 0

negative www_authorize result codes
by Juha Heinanen 09 Nov '11

09 Nov '11

auth_db/README tells this about www_authorize() negative result codes: Negative codes may be interpreted as follows: * -1 (generic error) - some generic error occurred and no reply was sent out; * -2 (invalid password) - valid user, but wrong password; * -3 (invalid user) - authentication user does not exist. whereas in auth_radius i return: Negative result codes may be interpreted as follows: * -7 (internal error) - some internal error occurred (see syslog); * -6 (nonce reused) - nonce is used more than once; * -5 (no credentials) - credentials were not found in request; * -4 (stale nonce) - stale nonce; * -3 (authorization failed) - RADIUS responded with Access Reject which may be, for example, due to user not found or wrong password; * -1 (error) - some error occured during authorization (see syslog); since both modules depend on the same auth module, why does www_authorize return the full set of negative result codes or is its README out of date? -- juha

2 2

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