- sr-users - kamailio.org

bflag mystery
by Juha Heinanen 23 Apr '13

23 Apr '13

i have in config: setbflag(8); xlog("L_INFO", "bflags are <$bf/$bF>\n"); save("location", "0x02"); and i correctly get to syslog: Apr 10 19:51:48 wheezy1 /usr/sbin/sip-proxy[6722]: INFO: bflags are <256/00000100> however, when i look what gets stores in location table, cflags column of the contact has value 384. it is also shown by ul_dump: Cflag:: 384 384 = 256 + 128. i must be missing something trivial. how is it possible that also flag 7 (which is my nat_bflag) gets set in usrloc? -- juha

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Kamailio 3.3.1 and Asterisk 11.3 with realtime integration
by Henning Bragge 23 Apr '13

23 Apr '13

Hi, I'm totally new to Kamailio and I attempt to add Kamailio functionality to an already working Asterisk implementation. I have been following this link: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb Everything seems to work with an exception of encountering NAT issues. The location table in Kamailio reports the local NAT address, instead of the public address, where then Asterisk reports UNREACHABLE status. I have installed and started RTPproxy as adviced in the above link - no changes. Best regards, Henning Bragge

2 1

Kamailio 4.0 IMS
by Pavel Segeč 23 Apr '13

23 Apr '13

Hi all Kamailio IMS community, after a time, we are starting to use Kamailio 4.0 based IMS platform during our university lessons (before we used Kamailio 3.2 based). However I'm missing some usefull links, docs, tutorials regarding of Kamailio 4.0 IMS procesess. The situation was quite poor alerady before for 3.2 but some was available on ng-voice web page, at least, now nothing (especially about new ng-voice HSS). Does anybody know and help us with some usefull information about the Kamailio IMS 4.0 and HSS? I promise to prepare and make publicly available guides on our site nil.uniza.sk (Kamailio award for 2010) Thanks again palo73

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BUG? - Dialog module timeout BYE is not ok.
by Uri Shacked 23 Apr '13

23 Apr '13

Hi, I think there is a bug with the BYE that is sent to the caller and callee when dialog timeout happends. The BYE headers are sent with no "<" or ">". So, some sip singaling points decline the BYE. BR, Uri

2 3

FreeBSD build broken (with patch)
by Victor V. Kustov 22 Apr '13

22 Apr '13

Hi all! # gmake group_include="standard mysql debug radius" CC (gcc) [kamailio] proxy.o CC (gcc) [kamailio] dns_func.o In file included from dns_func.c:31: /usr/include/resolv.h:157: error: array type has incomplete element type /usr/include/resolv.h:171: error: field 'addr' has incomplete type /usr/include/resolv.h:195: error: field 'sin' has incomplete type gmake: *** [dns_func.o] Error 1 It's because incomplete headers for FreeBSD. diff --git a/dns_func.c b/dns_func.c index 6eedf0d..5a9bf7e 100644 --- a/dns_func.c +++ b/dns_func.c @@ -28,6 +28,7 @@ #include "dns_func.h" +#include <netinet/in.h> #include <resolv.h> #include <sys/types.h> #include <netdb.h> I write patch in message body 'cause think need patch to config scripts, not dns_func.c. #include <netinet/in.h> - need before #include <resolv.h> -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 1

Load balancer using dispatcher / BYE relaying
by Ariel Wainer 22 Apr '13

22 Apr '13

StartFragmentHi, I'm new to Kamailio and I'm trying to setup a load balancer using Kamailio 4.0 and rtpproxy, to balance between 2 asterisk servers that provide an IVR service. I've followed several guides, and so far it seems to work along with rtpproxy, except it is not sending the BYE message back to asterisk, and the call never finishes from the asterisk POV. Let me describe the test scenario: I have the following: IP phone: 192.168.200.183 PBX: 192.168.200.3 Kamailio: 192.168.200.132 IVR1: 192.168.200.160 IVR2: 192.168.200.161 >From the IP phone which is registered on the PBX, I initiate a call to exten 800, PBX forwards it to 1234@192.168.200.132:5060 and Kamailio dispatches it to one of the two IVR in round-robin. I'm testing this way and not calling directly to Kamailio because in the production enviroment, calls will be coming from outside. Attached you will find the Kamailio configuration. Packet capture made at the Kamailio host, where you can see the BYE package from the PBX (first one on line 24) but no BYE package sent to the IVRs is here: http://cringer.3kh.net/web/misc/sip.cap Please let me know if I ommited any important detail of my setup and thanks in advance. Best regards. -- Ariel Wainer

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[SER-Users] debug: again.
by Victor V. Kustov 22 Apr '13

22 Apr '13

Hi all! Still same problem: i don't see debug messages. OS: FreeBSD 9.1 stable, fresh git Kamailio version. In present moment just debugger.so with config trace work as expected. Option "debug=X" seems has no effect at all. I saw debug logs in maillist. Maybe I make something wrong? I never see simulary strings in logs: debug: DEBUG: <core> [forward.c:511]: orig. len=460, new_len=483, proto=1 debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list 0xb323c178 debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [usr_avp.c:639]: DEBUG:destroy_avp_list: destroying list (nil) debug: DEBUG: <core> [receive.c:280]: receive_msg: cleaning up -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

1 0

When to use t_newtran/t_release?
by Yufei Tao 22 Apr '13

22 Apr '13

Hi, I've been using the default cfg file as a starting point and added my stuff on top. But I don't think I understand the use of t_newtran/t_release and when they should be used. Read the documentation but still not clear. For example, for presence: t_newtran is called before handle_publish/subscribe and t_release called afterwards. And in route[LOCATION], t_newtran is called before sending 4xx responses. Could someone help explain please what they do and when they should be used please? Thank you! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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maintenance of www.kamailio.org
by Henning Westerholt 22 Apr '13

22 Apr '13

Hi, I'll do today this noon a regular maintenance of www.kamailio.org. I don't expect any big outtages, the webserver could be maybe for a few seconds not available due to some restarts. I'll announce when the maintenance is finished. Best regards, Henning

1 0

Possible memory leak in parse_pai.c
by Tais Plougmann Hansen 22 Apr '13

22 Apr '13

Hi, I'm seeing a possible leak in parsing the P-Asserted-Identity, and possibly P-Preferred-Identity headers in kamailio-3.3.4. I have a server running in which one thread has currently most of the 8 MB private memory allocated. At restart qm_sums reports parse_pai.c as sitting on most of the memory: Apr 16 16:38:58 gws1 /usr/sbin/kamailio[9398]: INFO: qm_status: heap size= 8388608 Apr 16 16:38:58 gws1 /usr/sbin/kamailio[9398]: INFO: qm_status: used= 5369240, used+overhead=6417216, free=1971392 [...] Apr 16 16:39:03 gws1 /usr/sbin/kamailio[9398]: INFO: qm_sums: count= 6198 size= 4721776 bytes from <core>: parse_pai.c: parse_pai_header(58) I am currently using the header through $ai for xlog statements, the accounting module and a validity check (if (!($(ai{uri.user}) =~ "^\+[1-9]+")) {...}). >From digging through the logs it could seem like this happens when using $ai in the accounting module. I'll try removing it from accounting (log_extra) and see if it stops leaking. -- Tais Plougmann Hansen OSD Consulting ApS osd.dk Tel: +45 78101078 CVR: DK31332737

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Kamailio World Conference Presentations
by Daniel-Constantin Mierla 22 Apr '13

22 Apr '13

Hello, The other files with slides presented last week at Kamailio World Conference are made available on the web at: - http://www.kamailio.org/events/2013-KamailioWorld/ There are two presentations missing, hopefully they will be received soon and uploaded. Many thanks again to those coming to speak or attend the event. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Rtpproxy - IPv4/v6 Translation and earlier removing of session
by Jakub Hrabovský 22 Apr '13

22 Apr '13

Hi, I started working on IPv4/IPv6 Translation issue in Kamailio by using rtpproxy as Media relay last year in my Bachelor-project. I solved the trouble with Translation between IPv4 and IPv6 clients, where IPv4 client used public IPv4 address. Now I am working on the issue, where IPv4 client uses private IPv4 (is behind NAT in local network), so I have to provide IPv4/v6 and NAT for all messages in one call. My topology consists of: Kamailio 3.3.4 - SIPproxy (supports IPv4/IPv6, public IPv4 and IPv6 addresses) - makes NAT translation, works as SIPoutbound proxy and communicates with rtpproxy (v1.2.1), Kamailio 3.3.4 - SIPprotocol GW (== SIP bridge) - translates IPv4 messages to IPv6 (and IPv6 to IPv4), uses other rtpproxy, has public IPv4 and IPv6 addresses + only IPv4 and only IPv6 clients (Linphone v3.5.2) I assume, that both (IPv4 and IPv6) clients are in the same SIP domain. I solve IPv4-IPv6 translation so, that each message from IPv6 client goes to SIP proxy and (if needed) is forwarded to SIP Protocol GW. Translated medssage is then forwarded back to SIP proxy and from that to callee. Here i have some problems: -1.) IPv6 client doesnt respond to invites from IPv4 client. The messages are ignored at IPv6 client. -2.) Rtpproxy on the machine with SIP protocol gateway (which provides IPv4/v6 translation) removes session earlier than I need (usually after 30 seconds of call) and the result is call break. this issue doesnt appear for rtpproxy on SIP proxy. I dont know what causes this behavior, if it is only in some parameters, witch i have to set, or it is bug in rtpproxy or .... Here I get config files of topology entities: ----------------------------------------------------------------------------------------------------------------------------- config file for SIP proxy: ----------------------------------------------------------------------------------------------------------------------------- #!KAMAILIO # Konfiguracny subor pre SIP Proxy Server IPv4/IPv6 pre konkretnu domenu ip02.sip.uniza.sk # #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USERLOCDB #!define WITH_ALIASDB #!define WITH_NAT #>> #!define WITH_IPV6 #!define WITH_V4V6 #!ifdef WITH_V4V6 #!define BRIDGE_IP4 "158.193.139.207" #!define BRIDGE_IP6 "[2001:4118:300:122:a00:27ff:fe60:83c]" #!endif #<< #!define ADDR_IP4 158.193.139.46 #!ifdef WITH_IPV6 #!define ADDR_IP6 [2001:4118:300:122:444a:3fff:fecc:61a0] #!endif # *** 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 1 #!define FLB_NATSIPPING 2 #>> #!define FLB_IPV6 6 #!define FLB_V4V6 7 #<< ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="ip02.sip.uniza.sk" alias="[2001:4118:300:122:444a:3fff:fecc:61a0]" alias="158.193.139.46" #!ifdef WITH_IPV6 listen=udp:158.193.139.46:5060 listen=udp:[2001:4118:300:122:444a:3fff:fecc:61a0]:5060 listen=tcp:158.193.139.46:5060 listen=tcp:[2001:4118:300:122:444a:3fff:fecc:61a0]:5060 #!else /* 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 #!endif /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "" voicemail.srv_port = "" desc "" #!endif #>> #!ifdef WITH_V4V6 bridge.srv_ip4 = "" desc "SIPv64 PGW IPv4 Address" bridge.srv_ip6 = "" desc "SIPv64 PGW IPv6 Address" #!endif #<< ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #>> #!ifdef WITH_V4V6 loadmodule "ipops.so" loadmodule "exec.so" #!endif #<< #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #>> # ----- xlog params ----- modparam("xlog", "prefix", "ProxyLog: ") #<< # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "db_update_period", 20) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) #!endif #!ifdef WITH_XCAPSRV # ----- xcap_server params ----- modparam("xcap_server", "db_url", DBURL) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@ip02.sip.uniza.sk") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { 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[INDOMAIN]{ #!ifdef WITH_V4V6 ## if calle in local domain and V4/V6 translation needed -> ## -> rewrite IP in SIP URI back to other version of SIP domain IP if(!has_totag() && isbflagset(FLB_V4V6) && lookup('location')){ xlog("L_INFO","URI domain before change: $rd\n"); if(af == INET){ $ru="sip:" + $rU + "@" + "[2001:4118:300:122:444a:3fff:fecc:61a0]"+":"+$rp; }else{ $ru="sip:" + $rU + "@" + "158.193.139.46" + ":"+$rp; } xlog("L_INFO","Request URI after rewrite: $ru\n"); } #!endif return; } #<< #>> route[FROMBRIDGE]{ if($si==BRIDGE_IP6 || $si==BRIDGE_IP4){ return 1; } return -1; } route[TOBRIDGE]{ #!ifdef WITH_V4V6 if(isbflagset(FLB_V4V6)){ if(af == INET6){ if(!t_relay(BRIDGE_IP6,"5060")) sl_reply_error(); exit; }else{ if(!t_relay(BRIDGE_IP4,"5060")) sl_reply_error(); exit; } } #!endif return; } route[IPV4V6] { #!ifdef WITH_V4V6 if (!has_totag()){ if (to_uri!=myself){ if(from_uri==myself){ # SIP message goes from local domain out if(is_ip($rd)){ # target domain defined with IP address if((af==INET && is_ipv6_reference($rd)) || (af==INET6 && is_ipv4($rd))) { setbflag(FLB_V4V6); } } else { # target domain in R-URI defined with domain name - DNS used # launch script, which returns list of IP addresses of all SIP servers, which provide that service in R-URI # IP addresses saved in $avp(i:1), $avp(i:2), ... exec_avp("/etc/kamailio/test.sh _sip._'$rP'.'$rd'."); if($avp(i:1) != "Unknown"){ $var(i) = 1; $var(bridge) = "yes"; while ($avp($var(i)) != 0){ if((af==INET && is_ipv4($avp($var(i)))) || (af==INET6 && is_ipv6($avp($var(i))))) { # if exists IP with the same AF than AF of sender --> bridge isnt needed $var(bridge) = "no"; break; } $var(i) = $var(i)+1; } xlog("L_INFO","BRIDGE > $var(bridge)"); if($var(bridge) == "yes") { setbflag(FLB_V4V6); } } } } else { # message is only going throught SIP server return; } } else { if(af==INET6 && !isbflagset(FLB_IPV6)){ setbflag(FLB_V4V6); } else { if(af==INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } } } } #!endif return; } #<< 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"); } route(IPV4V6); route(INDOMAIN); route(TOBRIDGE); if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server 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(af==INET6) setbflag(FLB_IPV6); 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) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_V4V6 # messages from SIP Bridge without authentication if(route(FROMBRIDGE)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests 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] { # if from IPv6, NAT not needed if(af==INET6) return; #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if(af==INET6) return; 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("froc"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_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; } $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:" + "vmbox-" + $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(IPV4V6); 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 } ------------------------------------------------------------------------------------------------------------ config file for SIP protocol GW == SIP bridge ------------------------------------------------------------------------------------------------------------ #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### #!define WITH_IPV6 #!define WITH_NAT #!define WITH_V4V6 #!define WITH_BRIDGE #!define ADDR_IP4 158.193.139.207 #!ifdef WITH_IPV6 #!define ADDR_IP6 [2001:4118:300:122:a00:27ff:fe60:83c] #!endif # *** 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/kamailio" #!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 1 #!define FLB_NATSIPPING 2 #!define FLB_V4V6 6 ####### Global Parameters ######### # DNS parametre use_dns_cache=on #dns_try_ipv6=yes #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="bridge.local" alias="[2001:4118:300:122:a00:27ff:fe60:83c]" alias="158.193.139.207" #!ifdef WITH_IPV6 listen=ADDR_IP4 listen=ADDR_IP6 #!else /* 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 #!endif /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_V4V6 loadmodule "ipops.so" loadmodule "exec.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 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) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") #modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(BRIDGE); 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[BRIDGE]{ #!ifdef WITH_BRIDGE if(to_uri!=myself && from_uri!=myself){ # message goes only throught this server - BRIDGE # => translation provided setbflag(FLB_V4V6); } #!endif return; } 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] { # if from IPv6, NAT not needed if(af==INET6) return; #!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); }else { if(check_route_param("nat=v46")) { setbflag(FLB_V4V6); } } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)|| isbflagset(FLB_V4V6))) return; #rtpproxy_manage(); if(isbflagset(FLB_V4V6)) { if(af==INET6) { rtpproxy_manage("FAEI"); } else { rtpproxy_manage("FAIE"); } } else { if(af==INET6) { # zroj aj ciel su IPV6 rtpproxy_manage("FAEE"); } else { # zdroj aj ciel su IPV4 rtpproxy_manage("FAII"); } } if (is_request()) { if (!has_totag()) { if(isbflagset(FLB_V4V6)) { add_rr_param(";nat=v46"); } else { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(af==INET) { 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 } ----------------------------------------------------------------------------------------------Here some information from logging:from rtpproxy, wich translates SIP messages between IPv4 and IPv6-----------------------------------------------------------------------------------------------Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: new session 4817, tag 12674;1 requested, type strongApr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: new session on a port 36792 created, tag 12674;1Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: pre-filling caller's address with 2001:4118:300:122:c41b:584d:c288:90f6:7078Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: adding strong flag to existing session, new=1/0/0Apr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/0, session timer restartedApr 11 10:12:51 bridge rtpproxy[1103]: INFO:handle_command: adding strong flag to existing session, new=1/0/0Apr 11 10:12:51 bri dge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/0, session timer restartedApr 11 10:12:53 bridge rtpproxy[1103]: INFO:handle_command: lookup on ports 36792/52592, session timer restartedApr 11 10:12:53 bridge rtpproxy[1103]: INFO:handle_command: pre-filling callee's address with 158.193.139.46:36258>>>Apr 11 10:13:21 bridge rtpproxy[1103]: INFO:handle_delete: forcefully deleting session 1 on ports 36792/52592Apr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: RTP stats: 215 in from callee, 214 in from caller, 429 relayed, 0 droppedApr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: RTCP stats: 0 in from callee, 1 in from caller, 1 relayed, 0 droppedApr 11 10:13:21 bridge rtpproxy[1103]: INFO:remove_session: session on ports 36792/52592 is cleaned upApr 11 10:13:21 bridge rtpproxy[1103]: INFO:handle_command: delete request failed: session 4817, tags 12674/NONE not found----------------------------------------------------------------------------- ------------------- My problem is related to the line with >>> If somebody has experience with this kind of problem, i will be very thankful for help. Jakub Hrabovsky (sorry for my english grammar)

2 1

UAC example
by Moacir Ferreira 22 Apr '13

22 Apr '13

Hi community, I will start testing the UAC module. Does anyone have a simple UAC configuration that could share with me? Also, for small (less than 500 users) deployments, would you recommend Kamilio UAC as the solution for giving PSTN access via VoIP providers or would you recommend Asterisk, SEMS, etc.? My intention is to host the solution on a single server. Thanks, Moacir

2 1

Problem accessing git repository
by Mikko Lehto 22 Apr '13

22 Apr '13

Hi Yesterday I got "fatal: read error: Connection reset by peer" to my "git pull" command. I thought it was only temporary network error, but same problem still exist today. I can however access other remote git repositories succesfully from the same host. --- #sip-router 10.58 < mslehto> hmm... I get "Connection reset by peer" to git fetch command 10.59 < mslehto> propably local issue if you just updated master branch succesfully 11.10 < iZverg> git pull 11.11 < iZverg> # git pull 11.11 < iZverg> Already up-to-date. 11.11 < iZverg> i think all ok with server 13.07 < mslehto> I don't get it .pcap dumps shows some git traffic to git.sip-router.org, but finally I get TCP RST 13.07 < mslehto> I have remote.origin.url=git://git.sip-router.org/sip-router 13.08 < mslehto> seems to be same as in http://kamailio.org/wiki/install/devel/git 13.13 < iZverg> url = git://git.sip-router.org/sip-router 13.13 < iZverg> Already up-to-date. --- I think this is not local issue, any pointers what I should troubleshoot next? UPDATE: I am able to get source code via "git clone" to fresh directory with no errors, confusing. -- Mikko

4 9

Basic question
by אורן אברהם 21 Apr '13

21 Apr '13

Dear List. I'm trying to change the contact header of a message. ( by calliing add_contact_alias - that shuld add the ;alias= parameter to the contact header) but, when i print the contact header (with xlog), it shows no change. I know the function works, and that it added the parameter as expected by using debug mode. (it sais that it was added) Can anyone explain me why i don't see that change in the xlog ? (i've printer both $ct and $hdr(Contact)) thank's in advance...

3 4

NAT Traversal Problem
by Ishan Sawhney 19 Apr '13

19 Apr '13

Hi, We have a solution which requires only outbound NAT on SIP calls. Do you have a product which would solve the NAT Traversal problem on the outbound SIP calls? If yes, how much would be the cost of this product? Will it support 2000 simultaneous calls? BR// Ishan

3 2

Transformations
by Victor V. Kustov 19 Apr '13

19 Apr '13

Hi! According "Transformations cookbook" exist string, uri, name transformations. I think need ip transformations, simulary atoi()/aton() and split for stings with divider. By the way: avp_subst("$avp(s:sipa)/$avp(s:abc)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\1/"); avp_subst("$avp(s:sipa)/$avp(s:def)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\2/"); avp_subst("$avp(s:sipa)/$avp(s:ghk)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\3/"); avp_subst("$avp(s:sipa)/$avp(s:xyz)/g", "/(.*)\.(.*)\.(.*)\.(.*)/\4/"); $avp(s:numipa)=$avp(s:abc{s.int})<<24||$avp(s:defi{s.int})<<16||$avp(s:ghki{s.int})<<8||$avp(s:xyz{s.int}); where is error? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

Kamailio World Quick Follow Up
by Daniel-Constantin Mierla 18 Apr '13

18 Apr '13

Hello, first I want to thank to all participants to Kamailio World Conference, everyone having a relevant contribution to it, turning out the two days in a very successful event. Special thanks to our sponsors - Sipwise, Fraunhofer Fokus, Asipto, Sipgate, NG-Voice, IT Center, Amooma and Zoiper - without them it would have been pretty impossible to bring a great group of speakers, have a nice location, excellent logistics and services on site - practically all that had matter to get such great event. As organizers, we learned a lot, significantly helping to get even better for next editions. Please do send feedback to us, it is very important to improve for the next years, even if you didn't participate, you can tell us what you would like to have at such event. For everyone in the community I want to say that the slides should be published in the near future. We also recorded the sessions, I haven't checked the movies yet, if the quality is fair enough, then they will be made available (probably will take a bit longer than the slides). Several sessions were recorded separately also by Randy from VUC, being already available on youtube: - https://www.youtube.com/user/voipusers/videos?sort=dd I will send soon more details, as new resources are made available. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Implementing the Presence module(Kamailio) to work with websockets (sipML5)
by אורן אברהם 17 Apr '13

17 Apr '13

Dear List,

1 0

Ldap Authenticate issues
by Sebastian Bergerot 17 Apr '13

17 Apr '13

Hi I´m trying to authenticate on active directory but i have issues with the password. Here is my CFG Kamailio.cfg #!define WITH_DEBUG #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users at lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT # *** 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 #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://root:bernardo@128.2.101.19/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="coto.com.ar" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:128.2.101.19:5060 #listen=tcp:192.168.14.25:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp") # ----- 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) # suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967) # ----- 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) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- 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 #chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # 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:192.168.14.25:22222") # ----- 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 at teopad-toip.corp") # 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", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT"); # NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT"); # handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG"); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); xlog("L_INFO","apres t_check_trans"); # authentication route(AUTH); xlog("L_INFO","apres 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(); xlog("L_INFO","apres RECORD ROUTE"); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE"); # dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT"); ### requests for my local domains # handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE"); # handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR"); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN"); # user location service route(LOCATION); xlog("L_INFO","apres LOCATION"); route(RELAY); xlog("L_INFO","apres RELAY"); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { xlog("L_INFO","reply error"); 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] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else 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 xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); 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(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar"); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif 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; # } { #edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif { xlog ("L_INFO", "ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=$fU))"); # ldap search #if (!ldap_search("ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?teopad-Sip-Username,teopadSipPassword?sub?(teopad-Sip-Username=$fU)")) if (!ldap_search("ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (sAMAccountName=$fU)"); exit; case -2: # internal error sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("sAMAccountName/$avp(s:username)"); #ldap_result("userPassword/$avp(s:password)"); #ldap_result("teopad-Sip-Username/$avp(s:username)"); #ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); xlog('Username=$avp(s:username)'); #xlog('Password=$avp(s:password)'); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ; if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); 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 { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } } #consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167"); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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 = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); 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 } #edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif And the Log Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x825fde0, 19, -1, 0x0) fd_no=14 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:4299]: tcp: DBG: sending to child, events 1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3970]: selected tcp worker 2 7(3063) for activity on [tcp: 128.2.101.19:5060], 0xb34ed968 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1507]: received n=4 con=0xb34ed968, fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1898036878>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=333 global id=332 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=61476, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=333 global id=333 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=861 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3063 (7)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=429 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:50210;rport=49733;branch=z9hG4bK1898036878#015#012From: <sip:sbergerot@128.2.101.19>;tag=1684345209#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.5fba#015#012Call-ID: 1455060142#015#012CSeq: 5 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=862 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x8282020, 8, 2, 0xb34ed968), fd_no=1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:300]: EOF on 0xb34ed968, FD 8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1287]: tcp_read_req: EOF Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x8282020, 8, -1, 0x10) fd_no=2 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1434]: releasing con 0xb34ed968, state -1, fd=8, id=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3063]: DEBUG: <core> [tcp_read.c:1435]: extra_data (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3382]: handle_tcp_child: reader response= b34ed968, -1 from 2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 172.30.203.182 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:1089]: tcpconn_new: on port 49734, type 2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:1403]: tcpconn_add: hashes: 128:420:583, 6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x825fde0, 19, 2, 0xb34ed968), fd_no=13 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x825fde0, 19, -1, 0x0) fd_no=14 called Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:4299]: tcp: DBG: sending to child, events 1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3066]: DEBUG: <core> [tcp_main.c:3970]: selected tcp worker 3 8(3064) for activity on [tcp: 128.2.101.19:5060], 0xb34ed968 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1507]: received n=4 con=0xb34ed968, fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK815445088>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=327 global id=326 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=27623, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=327 global id=327 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=861 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3064 (8)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=428 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:25204;rport=49734;branch=z9hG4bK815445088#015#012From: <sip:sbergerot@128.2.101.19>;tag=1120136754#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.de09#015#012Call-ID: 1585139518#015#012CSeq: 1 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=862 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x8282020, 8, 2, 0xb34ed968), fd_no=1 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1107819308>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <2> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: siputils [checks.c:103]: no totag Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=496 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=501 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=508 a=24 n=t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=328 global id=327 T start=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=27620, isACK=0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=328 global id=328 T end=(nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=509 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres t_check_trans Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=512 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=928 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=795 a=25 n=is_method Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=863 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=817 a=25 n=is_present_hf Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=821 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1120136754 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap://sipaccounts/OU=_Usuarios,DC=redcoto,DC=com,DC=ar?sAMAccountName,userPassword?sub?(sAMAccountName=sbergerot)) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=844 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=824 a=25 n=ldap_search Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:273]: LDAP URL parsed into session_name [sipaccounts], base [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:433]: [sipaccounts]: performing LDAP search: dn [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)], client_timeout [5000000] usecs Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:479]: [sipaccounts]: ldap_search_ext_st failed: Can't contact LDAP server Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_connect.c:231]: [sipaccounts]: LDAP bind successful (ldap_host [ldap://172.16.3.51]) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: ldap [ldap_connect.c:278]: [sipaccounts]: LDAP reconnect successful Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:433]: [sipaccounts]: performing LDAP search: dn [OU=_Usuarios,DC=redcoto,DC=com,DC=ar], scope [2], filter [(sAMAccountName=sbergerot)], client_timeout [5000000] usecs Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: ldap [ldap_api_fn.c:240]: [sipaccounts]: [1] LDAP entries found Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=844 a=25 n=ldap_result Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=848 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap_search: found [1] entries for (uid=sbergerot) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=849 a=25 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: <script>: Username=sbergerot Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=856 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=851 a=27 n=pv_www_authenticate Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: <core> [sr_module.c:1657]: Could not convert PV to str Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: auth [auth_mod.c:636]: failed to get passwd value Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=852 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: ldap pv_authenticate failed Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=853 a=26 n=www_challenge Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:127]: build_challenge_hf: realm='128.2.101.19' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:138]: build_challenge_hf: qop='auth' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: auth [challenge.c:269]: auth: 'WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012' Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [msg_translator.c:206]: check_via_address(172.30.203.182, 172.30.203.182, 0) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2317]: tcp_send: send from reader (3064 (8)), reusing fd Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2553]: tcp_send: sending... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: after real write: c= 0xb34ed968 n=429 fd=8 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_main.c:2588]: tcp_send: buf=#012SIP/2.0 401 Unauthorized#015#012Via: SIP/2.0/TCP 172.30.203.182:25204;rport=49734;branch=z9hG4bK1107819308#015#012From: <sip:sbergerot@128.2.101.19>;tag=1120136754#015#012To: < sip:sbergerot@128.2.101.19>;tag=1e63bc776842177dff382010e03d32cf.8290#015#012Call-ID: 1585139518#015#012CSeq: 2 REGISTER#015#012WWW-Authenticate: Digest realm="128.2.101.19", nonce="UW6DllFugmoxy+OjSDvSKiZswnyYapMe", qop="auth"#015#012Server: kamailio (4.0.0 (i386/linux))#015#012Content-Length: 0#015#012#015#012 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=854 a=2 n=exit Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list 0xb34ed928 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [tcp_read.c:1311]: tcp_read_req: content-length= 0 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:128.2.101.19> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 235, <rport> = <n/a>; state=6 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK294428223>; state=16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=487 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=620 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=610 a=25 n=mf_process_maxfwd_header Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [30]; uri=[ sip:sbergerot@128.2.101.19] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body [<sip:sbergerot@128.2.101.19 >#015#012] Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <3> <REGISTER> Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=488 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres REQINIT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=491 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=946 a=2 n=return Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=492 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: apres NATDETECT Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=495 a=5 n=route Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=624 a=26 n=xlog Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: INFO: <script>: Dans WITHINDLG Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=671 a=16 n=if Apr 17 08:07:22 serverconf /usr/sbin/kamailio[3064]: ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=625 a=24 n=has_totag

1 0

Typecast
by Victor V. Kustov 17 Apr '13

17 Apr '13

Hi all. I've got following construction: modparam("acc_radius", "radius_extra", " \ Calling-Station-Id=$tU; \ User-Name=$rU; \ Called-Station-Id=$fU; \ Framed-IP-Address=$si; \ h323-call-origin=$var(ho); \ h323-disconnect-cause=$var(hdc); \ h323-connect-time=$var(ct); \ h323-disconnect-time=$var(dt); \ h323-setup-time=$var(st); \ Cisco-AVPair=$var(cavp); \ Cisco-Call-Type=$var(cct) \ ") Framed-IP-Address in dictionary described as "ipaddr". If i send $si (as writen above), then radius server get incorrect ip like 46.48.44.49. If I try to assign $si any attribute has string type - all ok. How can I typecast here? Simulary problem for attributes "h323-disconnect-time", "h323-setup-time" - need timestamp in format "0000-00-00 00:00:00" -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 3

Buffer size with xmlrpc module
by Jon Bergli Heier 17 Apr '13

17 Apr '13

Hello, we use the xmlrpc module to fetch the list of users with the ul_dump MI command. When the number of users increased, this started to fail as the output size grew. Apparently this is because of too small write queue sizes (tcp_conn_wq_max). I discussed increasing the tcp_conn_wq_max setting with a coworker, who was reluctant to change it as it would affect SIP over TCP as well. We would probably also have to increase it if we hit the limit again at a later time. What we would like to know is if there's a way around this without having to fiddle with buffer sizes (or if this behavior is considered a bug). We originally used the mi_xmlrpc module, but changed to the xmlrpc module as the former made kamailio spawn zombie processes on xmlrpc requests (which is considered a bug I guess). FWIW, here's the log from a failed request, running kamailio 3.3.4: Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: [REMOVED] Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:1089]: tcpconn_new: on port 40261, type 2 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:1403]: tcpconn_add: hashes: 40:3802:373, 326 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [io_wait.h:390]: DBG: io_watch_add(0x828660, 94, 2, 0x7f2f87f31ca8), fd_no=85 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x828660, 94, -1, 0x0) fd_no=86 called Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:4298]: tcp: DBG: sending to child, events 1 Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3969]: selected tcp worker 5 58(4960) for activity on [tcp:0.0.0.0:8080], 0x7f2f87f31ca8 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1358]: received n=8 con=0x7f2f87f31ca8, fd=10 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1168]: tcp_read_req: content-length= 153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:626]: method: <POST> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:628]: uri: </RPC2> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:630]: version: <HTTP/1.0> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [receive.c:149]: After parse_msg... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: xmlrpc [xmlrpc.c:2273]: new fake xml msg created (355 bytes):#012<POST sip:127.0.0.1:9 HTTP/1.0#015#012Via: SIP/2.0/TCP [REMOVED]:40261#015#012Host: [REMOVED]:8080#015#012User-Agent: xmlrpclib.py/1.0.1 (by www.pythonware.com)#015#012Content-Type: text/xml#015#012Content-Length: 153#015#012#015#012<?xml version='1.0'?>#012<methodCall>#012<methodName>mi</methodName>#012<params>#012<param>#012<value><string>ul_dump</string></value>#012</param>#012</params>#012</methodCall>#012> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:626]: method: <POST> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:127.0.0.1:9> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:630]: version: <HTTP/1.0> Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=153 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [msg_translator.c:206]: check_via_address([REMOVED], [REMOVED], 0) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: sl [sl_funcs.c:499]: execute callback for event type 1 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: siptrace [siptrace.c:1343]: trace off... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2316]: tcp_send: send from reader (4960 (58)), reusing fd Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2552]: tcp_send: sending... Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2586]: tcp_send: after real write: c= 0x7f2f87f31ca8 n=15928 fd=10 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_main.c:2587]: tcp_send: buf=[REMOVED] Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: <core> [tcp_main.c:708]: ERROR: wbufq_add(64634 bytes): write queue full or timeout (0, total 0, last write 6648472 s ago) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: sl [../../forward.h:171]: msg_send: ERROR: tcp_send failed Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: ERROR: xmlrpc [xmlrpc.c:804]: Error while sending reply Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3620]: handle_ser_child: read response= 7f2f87f31ca8, -2, fd -1 from 58 (4960) Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: ERROR: <core> [tcp_main.c:3634]: handle_ser_child: ERROR: received CON_ERROR for 0x7f2f87f31ca8 (id 7494), refcnt 3, flags 0x4018 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: xmlrpc [xmlrpc.c:2194]: xmlrpc: error while trying script Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1285]: releasing con 0x7f2f87f31ca8, state -2, fd=10, id=7494 Apr 1 23:38:24 server /usr/sbin/kamailio[4960]: DEBUG: <core> [tcp_read.c:1286]: extra_data (nil) Apr 1 23:38:24 server /usr/sbin/kamailio[4982]: DEBUG: <core> [tcp_main.c:3381]: handle_tcp_child: reader response= 7f2f87f31ca8, -2 from 5 Thanks, Jon

2 1

Max TCP connections
by Krishna Kurapati 16 Apr '13

16 Apr '13

Hi, Has anyone tested max TCP connections that a Kamailio node can handle? I have an i7, 8GB server. I am trying to set max to 64K now. I want to set it to 256K connections. Would that work? Thnx Krish Kura

2 1

Kamailio 4.0.0 not listening on TCP
by David Wilson 16 Apr '13

16 Apr '13

Hi Jeremy, Thanks for the suggestion, I'm hoping I haven't overlooked something quite that obvious! The last debug output I see on startup is: > Listening on > udp: 192.168.44.66:55060 > tcp: 192.168.44.66:55060 > Aliases: > <snip> > > WARNING: no fork mode > 0(20372) INFO: <core> [tcp_main.c:4833]: init_tcp: using epoll_lt as the io watch method (auto detected) > 0(20372) INFO: usrloc [hslot.c:53]: locks array size 512 > 0(20372) INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376 > 0(20372) INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 netstat shows listening for UDP on 55060, but nothing on TCP: $ netstat -l | grep 5060 udp 0 0 192.168.44.66:55060 *:* ps shows four kamailio processes running, all of which end when I Ctrl-C. Cheers, Dave. > Date: Fri, 12 Apr 2013 14:22:45 +0800 > From: Jeremy Ardley <jeremy.ardley(a)gmail.com> > Subject: Re: [SR-Users] Kamailio 4.0.0 not listening on TCP > > > Have you checked to make sure it's still running? > > > On 12/04/13 14:19, David Wilson wrote: >> Hello All, >> >> New to Kamailio, I'm running 4.0 on Ubuntu 12.04 (precise). >> >> Running from the command line with a minimal config, it tells me that it is listening on port 5060 for both TCP and UDP. I see the same behaviour with the default cfg. >> >> However, netstat does not list anything listening on TCP port 5060 and an attempted connection is rejected consistent with there being no open socket. >> >> Any suggestions on where I need to be looking to fix this? >> > > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 95, Issue 47 > ****************************************

5 8

xcap server crash
by Juha Heinanen 13 Apr '13

13 Apr '13

i tested kamailio 4.0 xcap server with sipclients and got crash. should i add a tracker entry? -- juha (gdb) where #0 0x00007fa35ab47764 in ?? () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007fa3547ff9e2 in check_match_header (body=..., etag=0x7fff50162170) at xcap_server.c:1625 #2 0x00007fa3547ff88a in check_preconditions (msg=0x7fff501631c0, etag_hdr=...) at xcap_server.c:1583 #3 0x00007fa3547f85ec in w_xcaps_put (msg=0x7fff501631c0, puri=0x7fa35a091470 "\300-\tZ\243\177", ppath=0x7fa35a091510 "\360:\tZ\243\177", pbody=0x7fa35a0915b0 " ;\tZ\243\177") at xcap_server.c:594 #4 0x000000000041bbe0 in do_action (h=0x7fff501630e0, a=0x7fa35a092f50, msg=0x7fff501631c0) at action.c:1092 #5 0x00000000004240a7 in run_actions (h=0x7fff501630e0, a=0x7fa35a092f50, msg=0x7fff501631c0) at action.c:1573 #6 0x0000000000422852 in do_action (h=0x7fff501630e0, a=0x7fa35a0aad20, msg=0x7fff501631c0) at action.c:1321 #7 0x00000000004240a7 in run_actions (h=0x7fff501630e0, a=0x7fa35a0726c0, msg=0x7fff501631c0) at action.c:1573 #8 0x00007fa3545e9355 in xhttp_process_request (orig_msg=0x7fa35a10d0d0, new_buf=0x7fa35a10e390 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nVia: SIP/2.0/TCP 192.98.102.10:45208\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple"..., new_len=1290) at xhttp_mod.c:271 #9 0x00007fa3545e9b8a in xhttp_handler (msg=0x7fa35a10d0d0) at xhttp_mod.c:346 #10 0x0000000000482492 in nonsip_msg_run_hooks (msg=0x7fa35a10d0d0) at nonsip_hooks.c:125 #11 0x00000000004982c7 in receive_msg ( buf=0x7fa351da7e30 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple 0.34.0\r\nConnection: close\r\nIf-None-Ma"..., len=1252, rcv_info=0x7fa351da7b68) at receive.c:158 #12 0x000000000051f294 in receive_tcp_msg ( tcpbuf=0x7fa351da7e30 "PUT /xcap-root/resource-lists/users/sip:jh@test.fi/index HTTP/1.1\r\nAccept-Encoding: identity\r\nContent-Length: 996\r\nHost: 192.98.102.10:8080\r\nUser-Agent: sipsimple 0.34.0\r\nConnection: close\r\nIf-None-Ma"..., len=1252, rcv_info=0x7fa351da7b68, con=0x7fa351da7b50) at tcp_read.c:1179 #13 0x0000000000520271 in tcp_read_req (con=0x7fa351da7b50, bytes_read=0x7fff50163c20, read_flags=0x7fff50163c18) at tcp_read.c:1384 #14 0x0000000000521aa4 in handle_io (fm=0x7fa35a0fa980, events=1, idx=-1) at tcp_read.c:1614 #15 0x000000000051b0ae in io_wait_loop_epoll (h=0x8c27e0, t=2, repeat=0) at io_wait.h:1092 #16 0x0000000000522341 in tcp_receive_loop (unix_sock=20) at tcp_read.c:1725 #17 0x0000000000515e3d in tcp_init_children () at tcp_main.c:4956 #18 0x00000000004658f9 in main_loop () at main.c:1723 #19 0x0000000000468594 in main (argc=18, argv=0x7fff501640b8) at main.c:2566

2 5

Authentcation with RADIUS on kamailio
by Juraj Dančík 12 Apr '13

12 Apr '13

Hi, I have troble with authentication on Kamailio using RADIUS authentication. For configure freeradius and radiusclient I used this tutorial: http://www.kamailio.org/docs/openser-radius-1.0.x.html In config file of freeradius I have enabled digest in modules. In sites-available/default is digest enabled too. In config file of radiusclient I use for authentication server named localhost and in file servers I have password for localhost. I think, in freeradius and radiusclient config file, there is nothing wrong. When I test configuration of freeradius and radiusclient with radclient, there is no problem. Access is accepted. But when I want to authenticate with SIP Client (I use Jitsi), all registration are accepted. It doesn't matter what username and password I write. Problem is probably in config file of Kamailio. I don't know for what are dictionary files exactly used. I include dictionary.kamailio in freeradius's dictionary and radiusclient's dicitionary too. I attached The content of this dictionary file. When I start Kamailio in debug mode I can see any record about authentication or radius. Can anyone help me? Thanks for reply. [1622]: DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: [1622]: DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> [1622]: DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:192.168.0.112> [1622]: DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> [1622]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <1> <REGISTER> [1622]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 [1622]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [31]; uri=[sip:fas@192.168.0.11 [1622]: DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["fas" <sip:fas@192.168.0.112>#015#012] [1622]: DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-383438-2e2d7047b [1622]: DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 [1622]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 [1622]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via [1622]: DEBUG: <core> [receive.c:149]: After parse_msg... [1622]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... [1622]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 [1622]: DEBUG: maxfwd [maxfwd.c:161]: value 70 decreased to 16 [1622]: DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 [1622]: DEBUG: <core> [parser/msg_parser.c:106]: found end of header [1622]: DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=2148579d [1622]: DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 [1622]: DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 [1622]: DEBUG: siputils [checks.c:103]: no totag [1622]: DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff [1622]: DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=18808, isACK=0 [1622]: DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 1622]: DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found [1622]: DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) [1622]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 13==9 && [192.168.0.112] [1622]: DEBUG: <core> [socket_info.c:593]: grep_sock_info - checking if port 5060 (advertise 0) matches por [1622]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 13==13 && [192.168.0.112] [1622]: DEBUG: <core> [socket_info.c:593]: grep_sock_info - checking if port 5060 (advertise 0) matches por [1622]: DEBUG: <core> [sruid.c:176]: new sruid is [uloc-51686c8e-656-1] (1 / 19) [1622]: DEBUG: registrar [reply.c:368]: created Contact HF: Contact: <sip:fas@192.168.0.100:5060;transport= [1622]: DEBUG: sl [sl.c:289]: reply in stateless mode (sl) [1622]: DEBUG: <core> [msg_translator.c:206]: check_via_address(192.168.0.100, 192.168.0.100, 0) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) [1622]: DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) [1622]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up #### Attributes ### ATTRIBUTE Sip-Method 101 integer # Schulzrinne, acc ATTRIBUTE Sip-Response-Code 102 integer # Schulzrinne, acc ATTRIBUTE Sip-Cseq 103 string # Schulzrinne, acc ATTRIBUTE Sip-To-Tag 104 string # Schulzrinne, acc ATTRIBUTE Sip-From-Tag 105 string # Schulzrinne, acc ATTRIBUTE Sip-Translated-Request-URI 107 string # Proprietary, acc ATTRIBUTE Sip-Src-IP 108 string # Proprietary, acc ATTRIBUTE Sip-Src-Port 109 string # Proprietary, acc ATTRIBUTE Digest-Response 206 string # Sterman, auth_radius ATTRIBUTE Sip-Uri-User 208 string # Proprietary, auth_radius ATTRIBUTE Sip-Group 211 string # Proprietary, group_radius ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius ATTRIBUTE SIP-AVP 225 string # Proprietary, avp_radius ATTRIBUTE Digest-Realm 1063 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce 1064 string # Sterman, auth_radius ATTRIBUTE Digest-Method 1065 string # Sterman, auth_radius ATTRIBUTE Digest-URI 1066 string # Sterman, auth_radius ATTRIBUTE Digest-QOP 1067 string # Sterman, auth_radius ATTRIBUTE Digest-Algorithm 1068 string # Sterman, auth_radius ATTRIBUTE Digest-Body-Digest 1069 string # Sterman, auth_radius ATTRIBUTE Digest-CNonce 1070 string # Sterman, auth_radius ATTRIBUTE Digest-Nonce-Count 1071 string # Sterman, auth_radius ATTRIBUTE Digest-User-Name 1072 string # Sterman, auth_radius ### Acct-Status-Type Values ### VALUE Acct-Status-Type Failed 15 # RFC2866, acc ### Service-Type Values ### VALUE Service-Type Call-Check 10 # RFC2865, uri_radius VALUE Service-Type Group-Check 12 # Proprietary, group_radius VALUE Service-Type Sip-Session 15 # Schulzrinne, acc, auth_radius VALUE Service-Type SIP-Caller-AVPs 30 # Proprietary, avp_radius VALUE Service-Type SIP-Callee-AVPs 31 # Proprietary, avp_radius ### Sip-Method Values ### VALUE Sip-Method Undefined 0 VALUE Sip-Method Invite 1 VALUE Sip-Method Cancel 2 VALUE Sip-Method Ack 4 VALUE Sip-Method Bye 8 VALUE Sip-Method Info 16 VALUE Sip-Method Options 32 VALUE Sip-Method Update 64 VALUE Sip-Method Register 128 VALUE Sip-Method Message 256 VALUE Sip-Method Subscribe 512 VALUE Sip-Method Notify 1024 VALUE Sip-Method Prack 2048 VALUE Sip-Method Refer 4096 VALUE Sip-Method Other 8192 VALUE Sip-Method INVITE 1 # Proprietary, acc VALUE Sip-Method CANCEL 2 # Proprietary, acc VALUE Sip-Method ACK 4 # Proprietary, acc VALUE Sip-Method BYE 8 # Proprietary, acc ######Kamailio config file############## loadmodule "auth.so" loadmodule "auth_radius" loadmodule "acc_radius" loadmodule "misc_radius" # -- auth_radius params -- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) # 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")) { # authenticate requests if (!radius_www_authorize("$td") { www_challenge("$td", "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; }

1 0

Kamailio 4.0.0 not listening on TCP
by David Wilson 12 Apr '13

12 Apr '13

Hello All, New to Kamailio, I'm running 4.0 on Ubuntu 12.04 (precise). Running from the command line with a minimal config, it tells me that it is listening on port 5060 for both TCP and UDP. I see the same behaviour with the default cfg. However, netstat does not list anything listening on TCP port 5060 and an attempted connection is rejected consistent with there being no open socket. Any suggestions on where I need to be looking to fix this? Regards, Dave.

2 1

Kamailio + websockets to jssip
by Glyn Lewis 11 Apr '13

11 Apr '13

I am looking for Kamailio consultants in Vancouver BC area to assist with install and setup of system on debian. Many thanks for any pointers or recommendations. Glyn

2 2

Radius - again.
by Victor V. Kustov 11 Apr '13

11 Apr '13

Good day. Need tips for radius auth/accounting. At fist, if I understand correctly, some changes here: modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") must be replaced to modparam("acc_radius", "service_type", 15) modparam("acc_radius", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") But... *Any* radius_extra parameters raise parsing error when K. start. And second problem - setflag(FLT_ACC) doesn't call query to radius server, acc_rad_request("Some comment"); - does. How it work in real life? -- ST, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 7

Radius: extra fields
by Victor V. Kustov 11 Apr '13

11 Apr '13

Good day! I had in config: modparam("acc_radius", "radius_extra", "User-Name=XXX") and get same error when start kamailio: ERROR: acc [acc_extra.c:192]: parse failed in <User-Name=XXX> around position 10 Any attributes from dictionary anyway got same message. -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

Kamalio report 404 not found upon INVITE ?
by Khoa Pham 11 Apr '13

11 Apr '13

Hi, I have my SIP client registered to Kamailio successfully. I've check the location table in Kamailio and assert this successful registration. But when another SIP client tries to make call to me, it receive 404 not found from Kamailio. What cause this problem ? Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 1

path uri problem
by Juha Heinanen 10 Apr '13

10 Apr '13

it is not my lucky day. anything new that i try, does not work. in proxy 1, i call add_path_received() on register request, which results into this kind of path header: Path: <sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58022;transport=tcp'>. then i forward the request to proxy 2 and call save() on it. it results in error: Apr 7 14:01:44 wheezy1 /usr/sbin/sip-proxy[8709]: ERROR: registrar [save.c:887]: Failed to parse Path: URI looks like kamailio is not able to parse path uri that it added itself. perhaps the quotes should be double ones? -- juha

10 42

mediaproxy-ng Tutorial
by Aft nix 10 Apr '13

10 Apr '13

Hi, I stumbled upon this git://github.com/sipwise/mediaproxy-ng.git which looked very neat to me. Its said that it can be used with kamailio. It seems like its backed sipwise inc. But no documentation is given there. Anyone know of a tutorial/documentation for how to use it? -- -aft

11 29

Diversion header authentication
by phillman25 10 Apr '13

10 Apr '13

Thank you Daniel and Morten for your assistance and prompt reply. To use the tobody transformation, i see that i would need to upgrade to 4.1 right? Im currently on 3.3. I tried the below code: $var(i)=0; while($(hdr(Diversion)[$var(i)]) != $null ) { $avp(s:divhdr) = $(hdr(Diversion)[$var(i)]); avp_subst("$avp(s:divhdr)", "/.*sip:(.*)(@.*)/\1/"); xlog("L_WARN", "$avp(s:divhdr)"); if (!is_user_in("$avp(s:divhdr)", "7")) { sl_send_reply("403", "NOT ALLOWED"); exit; }; $var(i) = $var(i) +1; } However, it seems like group module cant parse the output 313 as seen below: Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: WARNING: <script>: 313 Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: ERROR: group [group.c:114]: failed to parse URI <313> Apr 10 13:16:43 SipProxy-Test /usr/local/sbin/kamailio[7471]: ERROR: group [group.c:158]: failed to get username@domain A bigger issue is that a certain client is sending more than one diversion header with different format as seen below: Diversion: <tel:22030009>;reason=no-answer;screen=no;privacy=off Diversion: "Solonas A" <sip:16@10.10.10.22>;reason=unconditional So in this case i cant really know how to extract the diversion number using a static substitution. Is there a way to adapt to different formats to extract the diversion number? Thanking you in advance Phillip Date: Tue, 09 Apr 2013 16:43:56 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Diversion header authentication To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5164292C.4080408(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, just adding that the tobody transformation could be handy to extract the user or uri part of a Diversion header, not to fight with subst expressions: - http://www.kamailio.org/wiki/cookbooks/devel/transformations#to-body_transf… Cheers, Daniel On 4/9/13 3:16 PM, Morten Isaksen wrote: > Hi, > > I have not tested this, but try: > > $avp(s:divhdr) = $(hdr(Diversion)[$var(i)]); > avp_subst("$avp(s:divhdr)", "/.*sip:\+45(.*)(@.*)/\1/"); # Extract > number between +45 and @ > > if (is_user_in("$avp(s:divhdr)", "1") { ... } > > Please note that there can be more than one Diverseion header. In that > case you can use: > > $var(i)=0; > while($(hdr(Diversion)[$var(i)]) != $null ) { > $avp(s:divhdr) = > $(hdr(Diversion)[$var(i)]); > xlog("L_WARN", "$avp(s:divhdr)"); > > > $var(i) = $var(i) +1; > } > > > > /Morten

2 1

too many hope SIP issue with "Route" header in proxy server
by Youngjin Park 10 Apr '13

10 Apr '13

Hi, I have a problem with "Route" header in my proxy server. Incoming invite has "Route" header with the IP of the proxy. So, the proxied invite from proxy server is looping back to the proxy server. I believe that "Route" cannot be removed or deleted because it is one of system header. Is there any way to avoid this loop? Thanks Young

2 1

PLEASE HELP
by Bogdan Denysiuk 10 Apr '13

10 Apr '13

Hi Guys! i need configure kamailio 4.0 with Radius Could you please help me how yo do this? Send me example of config routes. thanks

3 2

Operator !~
by Jon Bonilla 09 Apr '13

09 Apr '13

Hi all I was checking for an operator not matching regular expression. Acording to the doc we have =~ and !~ for matching and not matching. http://www.kamailio.org/wiki/cookbooks/3.3.x/core#script_statements I tried this one: if(!is_domain_local("$rd") && $rd !~ "^.+\.local$") Resulting on these errors: root@sp1:~# /etc/init.d/kamailio-proxy restart Not executing command for kamailio-proxy: invalid configuration file! -e 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 36: syntax error 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 52: bad expression 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1067, column 52: bad command 0(1813) : <core> [cfg.y:3549]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1068, column 2: bad command 0(1813) : <core> [cfg.y:3546]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1083, column 2-21: syntax error 0(1813) : <core> [cfg.y:3546]: parse error in config file /etc/kamailio/proxy/proxy.cfg, line 1083, column 2-21: ERROR: bad config file (6 errors) while this one: if(!(is_domain_local("$rd") || $rd =~ "^.+\.local$")) Works perfectly fine. Am I wrong with the usage of !~ operator? cheers, Jon

2 2

Contact header from linphone too long to delete
by Ross Kusler 09 Apr '13

09 Apr '13

Hello, We noticed an issue with a bunch of stale entries in the mysql location table recently - all for the same user. It turns out that if you install linphone 2.0.2 on an iOS device (other versions/platforms might also be affected) with the transport mode set to TCP and the 'Push Notification' feature enabled, the client generates very long contact (258+ character) headers. The contact column in the location table is varchar(255), anything longer than this automatically gets truncated upon insert. When the entry later expires and usrloc tries to remove it, it generates a delete query that matches on the entire contact value and not the truncated version. This causes the delete query to fail and the usrloc record gets left behind in the location table. The (truncated) contact header value from the database: sip:22448462@98.248.37.244:52008;transport=tcp;app-id=org.linphone.phone.pro d;pn-type=apple;pn-tok=AA31DC9642E76E7D7DA8A0E419B210E1A00E8DF43A7A77C8CD295 5291E1BAC85;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg -snd=msg.caf;line=932257155 The failed delete query:. delete from location where username='XXXXXXXX' AND contact='sip:XXXXXXXX@11.22.333. 44:52000;transport=tcp;app-id=org.linphone.phone.prod;pn-type=apple;pn-tok=A A31DC9642E76E7D7DA8A0E419B210E1A00E8DF43A7A77C8CD2955291E1BAC85;pn-msg-str=I M_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=932257 155f64767' AND callid='1962857815'; What do you all think the best solution here is? My intuition is to simply increase the size of this column. If this is the right approach then it probably also makes sense to update the .sql file for new installs. Thoughts? Thank you, Ross

2 1

Diversion header authentication
by phillman25 09 Apr '13

09 Apr '13

Dear List I am currently using the group module to authenticate inbound calls using the From header using the below code: if (!is_user_in("From", "1")) { sl_send_reply("403", "NOT ALLOWED"); exit; }; }; I want to now authenticate the Diversion header, when the call is diverted, the same way as above using the group module how could i proceed with this? thanking you in advance Phillip

3 2

Re: [SR-Users] sr-users Digest, Vol 95, Issue 33
by julian arsanches 09 Apr '13

09 Apr '13

thank you int transformation worked for the dispatcher module now i can define gw dynamically. much appretiate your sugestion. On Tue, Apr 9, 2013 at 5:59 AM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Help with dispatcher module (julian arsanches) > 2. how to configure presence module with websocket (???? ?????) > 3. Re: Help with dispatcher module (Javi Gallart) > 4. Re: path uri problem (Andrew Pogrebennyk) > 5. Diversion header authentication (phillman25) > 6. Re: path uri problem (Juha Heinanen) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 8 Apr 2013 18:17:36 -0400 > From: julian arsanches <guandulito.com(a)gmail.com> > Subject: [SR-Users] Help with dispatcher module > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAG0Kfxv6-TV4Ghep= > Wj0BNVYXH0TLR6FW281fQA9WsOKGsK1Kw(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > thanks for all support from the forum before hand. > i am having an issue with my setup, i am trying to make the dispatcher > dynamic by asigning a value from a variable, that variable is define on my > dispatcher.lst but i am getting an error, if anyone can help thanks. > > my config on the relay route > first i strip the prefix from my dial string, and set variable y with that > code, then i strip it and dial it. but it does not work. > again thanks if anyone can helpme. > > > > xlog(" anrtes de strip >> el r-uri($ru)>($ru{s.substr,0,3})\n"); > > > $var(x) = $ru; #>> set varx as an string > > $var(y)=$(var(x){s.substr,4,3}); #>>>>strip the first 3 digits to find my > route > > > > xlog("dsroute $var(y) my test ($ru)\n"); #check for variables loaded > > strip(3); # strip the prefix to route to destinatio > > xlog("despues de strip time [$Tf] method ($rm) r-uri ($ru) 2nd via el code > ($var(y)\n"); #check for loaded variables after strip > > > ds_select_dst("$var(y)", "4");#>>carrer dynamic don't work variable is > load propperly but i get the following error > * > * > *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <script>: > despues de strip time [Mon Apr 8 17:59:32 2013] method (ACK) r-uri > (sip:54.236.192.19) 2nd via el code (54). >>> i see a the right code in the > variable* > *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <core> > [sr_module.c:1726]: Could not convert PV to int * > > > #ds_select_dst("097", "4");#statically define works > > xlog("print el destination y el $var(y) ($du)>>($td)>>> ($ru)\n"); > > if (!t_relay_to_udp()) { > > sl_reply_error(); > }; > > > exit; > } > > > > dispatcher.lst > > #lineformat > #setit(integer)destination(sipuri)flags(integer,optional) > > > #Asterisks > > 007 sip:1.1.1.1:5060 ##route 1 > 024 sip:2.2.2.2:5060 ##route 2 > 100 sip:1.1.1.100:5060 ##route 100 > 103 sip:1.1.1.103:5060 ###route 103 >

1 0

Kamailio World Updates - April 9
by Daniel-Constantin Mierla 09 Apr '13

09 Apr '13

Kamailio World Conference is one week away, only few bits to adjust, everything being pretty much in place. The agenda with presentations is available at: - http://conference.kamailio.com/k01/schedule/ Details about the speakers are available at: - http://conference.kamailio.com/k01/speakers/ Besides the two days filled with presentations and demos, the participants have the chance of winning Raspberry Pi devices or Zoiper Biz licenses (SIP softphone - http://www.zoiper.com). A grateful and big thank you for event sponsors and exhibitors that made possible to organize this event: Sipwise, Asipto, Fraunhofer Fokus, Sipgate, IT Center Portugal, NG Voice, Zoiper and Amooma: - http://conference.kamailio.com/k01/sponsors/ - http://conference.kamailio.com/k01/exhibitors/ The venue page gives now few more details about getting to conference area: - http://conference.kamailio.com/k01/venue/ Looking forward to meeting next week in Berlin! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Help with dispatcher module
by julian arsanches 09 Apr '13

09 Apr '13

thanks for all support from the forum before hand. i am having an issue with my setup, i am trying to make the dispatcher dynamic by asigning a value from a variable, that variable is define on my dispatcher.lst but i am getting an error, if anyone can help thanks. my config on the relay route first i strip the prefix from my dial string, and set variable y with that code, then i strip it and dial it. but it does not work. again thanks if anyone can helpme. xlog(" anrtes de strip >> el r-uri($ru)>($ru{s.substr,0,3})\n"); $var(x) = $ru; #>> set varx as an string $var(y)=$(var(x){s.substr,4,3}); #>>>>strip the first 3 digits to find my route xlog("dsroute $var(y) my test ($ru)\n"); #check for variables loaded strip(3); # strip the prefix to route to destinatio xlog("despues de strip time [$Tf] method ($rm) r-uri ($ru) 2nd via el code ($var(y)\n"); #check for loaded variables after strip ds_select_dst("$var(y)", "4");#>>carrer dynamic don't work variable is load propperly but i get the following error * * *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <script>: despues de strip time [Mon Apr 8 17:59:32 2013] method (ACK) r-uri (sip:54.236.192.19) 2nd via el code (54). >>> i see a the right code in the variable* *##Apr 8 17:59:32 ip-10-0-1-206 /usr/sbin/kamailio[8975]: ERROR: <core> [sr_module.c:1726]: Could not convert PV to int * #ds_select_dst("097", "4");#statically define works xlog("print el destination y el $var(y) ($du)>>($td)>>> ($ru)\n"); if (!t_relay_to_udp()) { sl_reply_error(); }; exit; } dispatcher.lst #lineformat #setit(integer)destination(sipuri)flags(integer,optional) #Asterisks 007 sip:1.1.1.1:5060 ##route 1 024 sip:2.2.2.2:5060 ##route 2 100 sip:1.1.1.100:5060 ##route 100 103 sip:1.1.1.103:5060 ###route 103

2 1

how to configure presence module with websocket
by אורן אברהם 09 Apr '13

09 Apr '13

hello. I am tring to SUBSCRIBE for presence status of a user on my kamailio server using the sipMl5 (over websockets) but kamailio failes. (more exactly the handle_subscribe failes.) I was told previusly by peter dunkly that it is not a bug, just a configuration problem. I am quoting his answer: "This is a configuration problem not a bug in the software. SUBSCRIBEs are dialog forming requests and NOTIFYs (in this scenario) are in-dialog requests. As such you need to use force_rport(), add_contact_alias(), and handle_ruri_alias() with presence dialogs in the same way you do with media session dialogs." How can i acheve this ? thanks in advance...

1 0

loose_route received parameter question
by Juha Heinanen 08 Apr '13

08 Apr '13

in my test, proxy p2 receives initial request from proxy p1, that has this kind of R-URI and Route header: INVITE sip:47089231@192.98.102.10:41073;transport=tcp SIP/2.0. Route: <sip:192.98.102.20:5070;transport=tcp;lr;received='sip:192.98.102.10:49127;transport=tcp'> route header is result of calling lookup in p1 on record: AOR:: jh(a)test.fi Contact:: sip:47089231@192.98.102.10:41073;transport=tcp Q=0.5 Expires:: 114 Callid:: 0ed240f3-dc79-4d65-93ef-3e92d178a892 Cseq:: 12 User-agent:: sipsimple 0.34.0 Received:: sip:192.98.102.10:49127;transport=tcp Path:: <sip:192.98.102.20:5070;transport=tcp;lr;received='sip:192.98.102.10:49127;transport=tcp'> i would expect that loose_route() would set $du to received parameter value, but it does not, because i get to syslog: Apr 8 12:44:54 wheezy1 /usr/sbin/sip-proxy[5723]: INFO: Trying to relay to du <<null>> or ru <sip:47089231@192.98.102.10:41073;transport=tcp> how can i tell p2 to set $du to received uri? do i need to add another route header in p1 that contains received uri? -- juha

2 2

pkg mem problems. Leaking?
by Jon Bonilla 08 Apr '13

08 Apr '13

Hi all I have two different kamailio servers running in a single machine. The first one is a sl proxy and the second one acts as stateful proxy/registrar. From time to time the stateless one goes out of pkg mem. It has 8MB (also tried with 16 in another setup). The sl proxy listens the public interface for external connections and the loopback interface for conecting to the stateful proxy. As I have seen since I installed a mem debug kamailio 3.3.2 version there 6 days ago, the internal socket children have grown on mem usage while the external ones remain more or less stable. As example one uf the external udp receivers uses less than 1MB while one of the internal ones uses almost 6MB. I'm attaching the mem dump of one of those internal udp receiver processes via "ngcp-sercmd lb cfg.set_now_int core mem_dump_pkg 25437" command. I see several tens of thousands of lines like these ones: N address=0x7fd1207383c0 frag=0x7fd120738390 size=56 used=1 alloc'd from <core>: parser/parse_param.c: parse_params(539) start check=f0f0f0f0, end check= c0c0c0c0, abcdefed I have no idea about how to continue debugging this. I'm pasting some more info regarding the process list, mem status and kamailio version/flags. Thanks sp1:~# ngcp-sercmd lb ps 25416 attendant 25429 udp receiver child=0 sock=77.77.77.77:5060 25430 udp receiver child=1 sock=77.77.77.77:5060 25431 udp receiver child=2 sock=77.77.77.77:5060 25432 udp receiver child=3 sock=77.77.77.77:5060 25433 udp receiver child=4 sock=77.77.77.77:5060 25434 udp receiver child=5 sock=77.77.77.77:5060 25435 udp receiver child=6 sock=77.77.77.77:5060 25436 udp receiver child=7 sock=77.77.77.77:5060 25437 udp receiver child=0 sock=127.0.0.1:5060 25438 udp receiver child=1 sock=127.0.0.1:5060 25439 udp receiver child=2 sock=127.0.0.1:5060 25440 udp receiver child=3 sock=127.0.0.1:5060 25441 udp receiver child=4 sock=127.0.0.1:5060 25442 udp receiver child=5 sock=127.0.0.1:5060 25443 udp receiver child=6 sock=127.0.0.1:5060 25444 udp receiver child=7 sock=127.0.0.1:5060 25445 slow timer 25446 timer 25447 ctl handler 25448 MI FIFO 25449 tcp receiver (generic) child=0 25450 tcp receiver (generic) child=1 25451 tcp receiver (generic) child=2 25452 tcp receiver (generic) child=3 25453 tcp receiver (generic) child=4 25454 tcp receiver (generic) child=5 25455 tcp receiver (generic) child=6 25456 tcp receiver (generic) child=7 25457 tcp receiver (tcp:127.0.0.1:5060) child=8 25458 tcp receiver (tcp:127.0.0.1:5060) child=9 25459 tcp main process sp1:~# ngcp-sercmd lb pkg.stats { entry: 0 pid: 25416 rank: 0 used: 901136 free: 15417544 real_used: 1359672 } { entry: 1 pid: 25429 rank: 1 used: 909360 free: 15408648 real_used: 1368568 } { entry: 2 pid: 25430 rank: 2 used: 909400 free: 15408608 real_used: 1368608 } { entry: 3 pid: 25431 rank: 3 used: 909352 free: 15408656 real_used: 1368560 } { entry: 4 pid: 25432 rank: 4 used: 909344 free: 15408664 real_used: 1368552 } { entry: 5 pid: 25433 rank: 5 used: 909392 free: 15408712 real_used: 1368504 } { entry: 6 pid: 25434 rank: 6 used: 909344 free: 15408664 real_used: 1368552 } { entry: 7 pid: 25435 rank: 7 used: 909344 free: 15408664 real_used: 1368552 } { entry: 8 pid: 25436 rank: 8 used: 909352 free: 15408656 real_used: 1368560 } { entry: 9 pid: 25437 rank: 9 used: 5884872 free: 2480784 real_used: 14296432 } { entry: 10 pid: 25438 rank: 10 used: 5935272 free: 2359728 real_used: 14417488 } { entry: 11 pid: 25439 rank: 11 used: 5971616 free: 2305912 real_used: 14471304 } { entry: 12 pid: 25440 rank: 12 used: 5821672 free: 2522672 real_used: 14254544 } { entry: 13 pid: 25441 rank: 13 used: 5945584 free: 2310056 real_used: 14467160 } { entry: 14 pid: 25442 rank: 14 used: 5868760 free: 2495264 real_used: 14281952 } { entry: 15 pid: 25443 rank: 15 used: 5903672 free: 2457952 real_used: 14319264 } { entry: 16 pid: 25444 rank: 16 used: 5899208 free: 2453872 real_used: 14323344 } { entry: 17 pid: 25445 rank: -1 used: 983944 free: 15409512 real_used: 2267696 } { entry: 18 pid: 25446 rank: -1 used: 909168 free: 15409512 real_used: 1367704 } { entry: 19 pid: 25447 rank: -2 used: 983664 free: 15407912 real_used: 2267032 } { entry: 20 pid: 0 rank: 0 used: 918424 free: 15400064 real_used: 1377152 } { entry: 21 pid: 25449 rank: 17 used: 989192 free: 15329200 real_used: 1448016 } { entry: 22 pid: 25450 rank: 18 used: 989192 free: 15329200 real_used: 1448016 } { entry: 23 pid: 25451 rank: 19 used: 989192 free: 15329200 real_used: 1448016 } { entry: 24 pid: 25452 rank: 20 used: 989192 free: 15329200 real_used: 1448016 } { entry: 25 pid: 25453 rank: 21 used: 989192 free: 15329200 real_used: 1448016 } { entry: 26 pid: 25454 rank: 22 used: 989192 free: 15329200 real_used: 1448016 } { entry: 27 pid: 25455 rank: 23 used: 989192 free: 15329200 real_used: 1448016 } { entry: 28 pid: 25456 rank: 24 used: 989192 free: 15329200 real_used: 1448016 } { entry: 29 pid: 25457 rank: 25 used: 989192 free: 15329200 real_used: 1448016 } { entry: 30 pid: 25458 rank: 26 used: 989192 free: 15329200 real_used: 1448016 } { entry: 31 pid: 25459 rank: -4 used: 983240 free: 15334096 real_used: 2266992 } sp1:~# kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 13:08:06 Oct 31 2012 with gcc 4.4.5

3 3

usrloc + multimaster replication
by Alex Balashov 08 Apr '13

08 Apr '13

I know I'm revisiting a problem that has been discussed in multiple threads from various angles, so I might be rightly accused of laziness in neglecting to research them all. All the same: I have proxy1 and proxy2 writing to database A and database B, respectively. Database A and database B are active-active masters, synchronised via some replication system attached to the underlying DB technology. The 'location' table is also replicated this way. We know that 'usrloc' doesn't work so well with this: one instance of Kamailio will periodically delete the other's contacts, even if they have a nonlocal SIP domain. Is there any db_mode that can be used (other than 0/purely in-memory) to make this work right? Or is that the essential problem that p_usrloc is written to solve? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

4 12

problem with core forward function
by Juha Heinanen 08 Apr '13

08 Apr '13

now that the send function issue got resolved, i tried core forward function. core cookbook tells: forward Forward the SIP request to the given destination in stateless mode. This has the format of [proto:]host[:port]. Host can be an IP or hostname; supported protocols are UDP, TCP and TLS. (For TLS, you need to compile the TLS support into core). If proto or port are not specified, NAPTR and SRV lookups will be used to determine them (if possible). If destination parameter is missing, the forward will be done based on RURI. Example of usage: forward("10.0.0.10:5060"); #or forward(); i then went i tried forward("tcp:192.98.102.20:5070"); but got errors from that line: Apr 7 13:52:10 siika /usr/sbin/sip-proxy[22923]: : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "tcp:192.98.102.20:5070" Apr 7 13:52:10 siika /usr/sbin/sip-proxy[22923]: ERROR: <core> [route.c:1214]: fixing failed (code=-478) at cfg:/etc/sip-proxy/sip-proxy.cfg:1584 i also tried with forward_tcp("192.98.102.20:5070"); but it didn't help either: Apr 7 13:54:24 siika /usr/sbin/sip-proxy[23056]: : <core> [proxy.c:278]: ERROR: mk_proxy: could not resolve hostname: "192.98.102.20:5070" Apr 7 13:54:24 siika /usr/sbin/sip-proxy[23056]: ERROR: <core> [route.c:1214]: fixing failed (code=-478) at cfg:/etc/sip-proxy/sip-proxy.cfg:1584 how should core cookbook be fixed this time? -- juha

2 2

expired record not removed from location table
by Juha Heinanen 08 Apr '13

08 Apr '13

i have seen emails about expired records not removed from database, but i could not find the thread. i'm not sure if it is related, but today i noticed too that there is an old record in my location table which has expired couple of hours ago. the record does not show up with ul_dump command, so it has been removed from memory. usrloc db_mode=1. what could cause the record not be removed? is it a bug in cleaner process? i have usrloc timer_procs=2. -- juha

2 3

textops warning
by Juha Heinanen 07 Apr '13

07 Apr '13

i don't remember seeing this before: Apr 7 19:09:57 siika /usr/sbin/sip-proxy[8734]: WARNING: textops [textops.c:1816]: free function has not been defined for spve what should i do about it? -- juha

1 1

looks like send core function does not exist
by Juha Heinanen 07 Apr '13

07 Apr '13

i removed corex from loaded modules and added string argument to send: send("192.98.102.20:5070"); that gave error: 0(15695) ERROR: <core> [cfg.y:3415]: cfg. parser: failed to find command send 0(15695) : <core> [cfg.y:3554]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 1578, column 34: unknown command, missing loadmodule? according to core cookbook: send Send the original SIP message to a specific destination in stateless mode. No changes are applied to received message, no Via header is added. Host can be an IP or hostname. Used protocol: UDP Parameter is mandatory and has string format. Example of usage: send("10.10.10.10:5070"); does anyone know what is going on? -- juha

2 2

Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 06 Apr '13

06 Apr '13

Hi, I have tried by setting flag only but still no luck,Here is the configuration i am using. # sip trace modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "db_url", "") modparam("siptrace", "duplicate_uri", "sip:homer.com:9060") modparam("siptrace", "duplicate_with_hep", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) setflag(22); if (!t_relay()) { xlog("L_ERR", "CALL $ci - Relay error\n"); sl_reply_error(); exit; } Thanks Ram

2 1

forward to pseudo var destination
by Juha Heinanen 06 Apr '13

06 Apr '13

looks like forward core function does not accept pseudo var argument. then i found this: I just committed a patch that allow using send() without any parameter: * http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=8e4db80… if that commit is still valid, it is not reflected by core cookbook: send Send the original SIP message to a specific destination in stateless mode. No changes are applied to received message, no Via header is added. Host can be an IP or hostname. Used protocol: UDP Parameter is mandatory and has string format. Example of usage: send("10.10.10.10:5070"); so which way this is? -- juha

2 2

Changing Domain in To URI
by qmail＠polarismail.com 06 Apr '13

06 Apr '13

Sorry if I repeat myself - I just subscribed properly to the list: > if you don't need to change it back for replies and next requests, try: > > $td = "xyz.com"; > > IIRC, that is available on 3.2.x. If not, either you upgrade or use > replace functions from textops. > > Cheers, > Daniel Thank you for replying. We tried the direct assignment of $td="xyz.com" but the variable doesn't change. We are doing an xlog print right after and an ngrep and we confirmed that the domain part of the URI doesn't change. I also tried a subst() and it's the same problem. I read on the online documentation that the variable is read-only and I figured it can't be changed through these methods. That's why I tried uac_replace_to(). To sum it up, is that variable read-only or not ?

3 5

Re: [SR-Users] sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 06 Apr '13

06 Apr '13

hi, i have set the following flag it is still not capturing outbound SIP messages. { setflag(22) sip_trace(); } Daniel-Constantin Mierla<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=from:%…> Fri, 05 Apr 2013 00:45:16 -0700<http://www.mail-archive.com/search?l=sr-users@lists.sip-router.org&q=date:2…> Hello, On 3/31/13 4:27 PM, Ramaseshi reddy kolli wrote: Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. iirc, you have to set a flag to trace outgoing messages. the sip_trace()function just takes the snapshot of processed messages. Cheers, Daniel

2 1

sip_capture columns
by Grant Bagdasarian 06 Apr '13

06 Apr '13

Hello, I couldn't find any documentation on what the table columns of the sip_capture module are used for. Most of them are straightforward, but I still don't know what the following columns mean and what values to expect: - PID_user - Proto - Family - Rtp_stat - Type - Node Type has for example values 1 and 2. Im guessing 1 is for requests and 2 is for responses. I hope someone could clear this out for me. Thanks, Grant

2 2

No RTP between rtpproxy and smartphone application
by aaaa aaaa 05 Apr '13

05 Apr '13

Hello, I have the following topology. Kamailio as SIP Proxy and Asterisk as B2BUA. In Kamailio I use rtpproxy for NAT users. The problem that I have is the following: -When the CSipSimple is registered the first call is always unsuccesfull. The following calls are always successful. If the application remains idle for a while the next call is unsuccessful again. From traces I saw that everything is OK in signaling part. -When I try with Zoiper and 3CX I have 1/20 successful calls. In any case from Asterisk side these calls can be seen as active but no media. Thanx, Theo

3 2

Changing Domain in To URI
by qmail＠polarismail.com 05 Apr '13

05 Apr '13

Hello, As it happens we run Kamailio 3.2.0 on Debian Lenny which for some reason does not support uac_replace_to()! Anyhoo, we are trying to change the domain part in the To URI. What is the best way to do it ? Thank you!

2 1

sip_trace unable to capture after next_branches()
by Ramaseshi reddy kolli 05 Apr '13

05 Apr '13

Hi, I am using sip_trace function after next_branches(), it is not capturing outbound SIP INVITE request can anyone please help me. Thanks Ram

2 1

Kamailio 4.0 configuration
by Martin Krška 04 Apr '13

04 Apr '13

Hi everyone, I'm trying to configure some basic functions in Kamailio 4.0. Now let's say I have two users, for example Alice and Bob. When Alice adds Bob to her contacts, I need from her to ask Bob for permission to watch his presence. It's probably a trivial question but could anyone tell me what do I need for this to work, like what modules are responsible for this etc. Best Regards, Martin

2 1

db_mode 3 in usrloc module
by Morten Isaksen 04 Apr '13

04 Apr '13

Hi, I have an older installation of Kamailio (3.1.1). It is configured with modparam("usrloc", "db_mode", 2) I tried to change this to modparam("usrloc", "db_mode", 3) to not have to restart kamailio when I change the alias table directly from SQL. But then kamailio shutsdown after about one second with this error: Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [db.c:408]: invalid version 0 for table trusted found, expected 5 (check table structure and table "version") Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: permissions [trusted.c:250]: error during table version check. Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module permissions (/usr/local/lib/kamailio/modules_k/permissions.so) Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [pt.c:481]: ERROR: fork_tcp_process(): init_child failed for process 28, pid 11510, "tcp receiver child=6" Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11510]: ERROR: <core> [tcp_main.c:4811]: ERROR: tcp_main: fork failed: Success Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11512]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 44 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: ALERT: <core> [main.c:738]: child process 11510 exited normally, status=255 Mar 26 20:38:55 sip-3-1 /usr/local/sbin/kamailio[11482]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD I have not made any changes to the permissions/trusted module, so I suspect it is some shared database conection problem. Is it a known bug in this version? I would prefer not to upgrade the installation and hopes one of you can point me to a patch I can backport. -- Morten Isaksen

2 7

Problem with expired SIP Etag
by David 04 Apr '13

04 Apr '13

Hello, I have two Kamailio servers. Running 3.3.2. One uses PUA to send PUBLISH to the second which has presence module. When the presence server replies to the PUBLISH, it includes a SIP-ETag and an Expires header. So, in the case I caught on my servers yesterday, the reply had an Expires header of 34. Trouble is, 60 seconds later, when PUA sent the PUBLISH, it had the SIP-ETag in the PUBLISH header even though the entity tag was expired. But RFC3903 says that when the publish expires, the entity tag expires as well. The publish was for a dialog-info and both PUBLISHes were generated using the pua_mi module command pua_publish. I don't set the etag in the pua_publish command, it's kamailio that remembers it. Here is grep pua kamailio.cfg on the first server : loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_dialoginfo.so" modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "include_callid", 0) modparam("pua", "hash_size", 15) modparam("pua", "db_mode", 0) modparam("pua_dialoginfo", "send_publish_flag", 13) modparam("pua", "update_period", 6 ) # I added this to see if the problem would be fixed modparam("pua_dialoginfo", "caller_confirmed", 1) I didn't capture the mi udp packets, but I did get the SIP, see below I think the problem has to do with expires_offset. I set this to reduce SUBSCRIBE expiration by 30 seconds, but it is also reducing PUBLISH expiration by 30 seconds. Do you think this might be the source of my problem? What else might be causing this ? Is there anyway to set the expires_offset for SUBSCRIBE but not for the PUBLISH? Any other ideas what might be going on here? Thanks, David Here is the SIP I captured on the pua server : U 2013/03/29 14:01:41.005025 PUBLISH sip:user161@omnity.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@omnity.net. From: sip:user161@omnity.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Content-Length: 0. Max-Forwards: 70. Event: dialog. Expires: 66. SIP-If-Match: a.1363343972.1019.567098.10. U 2013/03/29 14:01:41.298041 SIP/2.0 200 OK. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK267e.e988a546.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-0dd9. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-c8fe. CSeq: 10 PUBLISH. Call-ID: 79a5f23d217e3597-9109@proxyip. Expires: 36. SIP-ETag: a.1363343972.1009.565433.11. Content-Length: 0. U 2013/03/29 14:03:14.317797 PUBLISH sip:user161@mydomain.net SIP/2.0. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Content-Length: 271. Max-Forwards: 70. Event: dialog. Expires: 91. SIP-If-Match: a.1363343972.1009.565433.11. Content-Type: application/dialog-info+xml. . <?xml version='1.0'?> <dialog-info xmlns='urn:ietf:params:xml:ns:dialog-info' version='0' state='full' entity='user161(a)omnity.net'> <dialog id='x_dialogid(a)192.168.2.5' direction='recipient'> <state>early</state> </dialog> </dialog-info> U 2013/03/29 14:03:14.319691 SIP/2.0 412 Conditional request failed. Via: SIP/2.0/UDP proxyip;branch=z9hG4bK543f.efdb2d8.0. To: sip:user161@mydomain.net;tag=f0253efb8005057b3cc0ab8fa35ff1d9-9add. From: sip:user161@mydomain.net;tag=047a42f14f0e800c0a21487d4bc09012-fab2. CSeq: 10 PUBLISH. Call-ID: 79a5f23d214a91de-9057@proxyip. Server: OmniVigil 5.2-rc1. Content-Length: 0.

1 0

Rewrite Prefix in Carrieroute
by Ali Jawad 04 Apr '13

04 Apr '13

Hi I do have a server with kamailio running on carrireroute, recently the need to rewrite prefixes arose, so I did enter the rewrite prefix into the relevant rows/columns in carrierroute tdb table and did reload but nothing changed. The call is matched to it's needed destination but the prefix is the same. In my config I do have the following line if(!cr_route("default", "ata.nymgo.com", "$rU", "$rU", "call_id")){ Regards

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New developer: Torrey Searle
by Daniel-Constantin Mierla 04 Apr '13

04 Apr '13

Hello, I want to announce that a new person got developer GIT write access to repository: Torrey Searle - he is going to add and maintain the sipt module, facilitating operations over SIP-T/SIP-I messages, the patch being on tracker: - http://sip-router.org/tracker/index.php?do=details&task_id=281&project=1 His git commit id is: tsearle My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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What IP in REGISTER message does Kamalio use to store location ?
by Khoa Pham 04 Apr '13

04 Apr '13

Hi, when client send REGISTER message to Kamalio. Kamailio will save that client's IP to database. But what is this IP ? Is is 1,2 or 3 ? 1. The IP from Via field 2. The IP from Contact field 3. The IP from the IP packet Many thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

3 6

Abort at qm_debug_frag function
by Bruno Bresciani 03 Apr '13

03 Apr '13

Kamailio is generating a abort at qm_debug_frag function (BUG: qm_*: prev. fragm. tail overwritten) but I can't understanding the meaning this abort... Someone can tell me the purpose of qm_debug_frag function, it checks if some improper memory handling happened and abort the program? Below is backtrace of abort: #0 0x00f41402 in __kernel_vsyscall () #1 0x009fec10 in raise () from /lib/libc.so.6 #2 0x00a00521 in abort () from /lib/libc.so.6 #3 0x081797b7 in qm_debug_frag (qm=0xb5f83000, f=<value optimized out>) at mem/q_malloc.c:158 #4 0x0817a6f3 in qm_free (qm=0xb5f83000, p=0xb6109a54, file=0x43760e "perms_db: src_ip.c", func=0x4378a0 "list_destroy", line=79) at mem/q_malloc.c:442 #5 0x0043563c in list_destroy (liste=0xb6107274) at src_ip.c:79 #6 0x00430a93 in reload_srcip_table () at db.c:163 #7 0x004313be in perms_db_srcip_reload (cmd_tree=0x835aeac, param=0x0) at fifo.c:107 #8 0x00335d97 in mi_fifo_server (fifo_stream=0x99561d8) at ../../lib/kmi/mi.h:77 #9 0x00337621 in fifo_process (rank=1) at mi_fifo.c:235 #10 0x003378a5 in mi_child_init (rank=0) at mi_fifo.c:199 #11 0x08120a02 in init_mod_child (m=0x82f4b88, rank=0) at sr_module.c:829 #12 0x081209dc in init_mod_child (m=0x82f4d1c, rank=0) at sr_module.c:826 #13 0x081209dc in init_mod_child (m=0x82f5164, rank=0) at sr_module.c:826 #14 0x081209dc in init_mod_child (m=0x82f79f4, rank=0) at sr_module.c:826 #15 0x081209dc in init_mod_child (m=0x82f7cf0, rank=0) at sr_module.c:826 #16 0x081209dc in init_mod_child (m=0x82f8174, rank=0) at sr_module.c:826 #17 0x081209dc in init_mod_child (m=0x82f8798, rank=0) at sr_module.c:826 #18 0x081209dc in init_mod_child (m=0x82f8b54, rank=0) at sr_module.c:826 #19 0x081209dc in init_mod_child (m=0x82f8eb4, rank=0) at sr_module.c:826 #20 0x081209dc in init_mod_child (m=0x82f90b4, rank=0) at sr_module.c:826 #21 0x081209dc in init_mod_child (m=0x82f952c, rank=0) at sr_module.c:826 #22 0x081209dc in init_mod_child (m=0x82fa980, rank=0) at sr_module.c:826 #23 0x081209dc in init_mod_child (m=0x82fab9c, rank=0) at sr_module.c:826 #24 0x081209dc in init_mod_child (m=0x82faff8, rank=0) at sr_module.c:826 #25 0x080ae175 in main_loop () at main.c:1624 #26 0x080b1256 in main (argc=3, argv=0xbf9a8f14) at main.c:2398 Best Regards

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Re: [SR-Users] sr-users Digest, Vol 95, Issue 17
by julian arsanches 03 Apr '13

03 Apr '13

Thanks a lot it was just what i need it , it worked like a charm. strip(3); On Wed, Apr 3, 2013 at 3:39 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: please help with some transformations. (Alex Balashov) > 2. Re: please help with some transformations. > (Daniel-Constantin Mierla) > 3. Abort at qm_debug_frag function (Bruno Bresciani) > 4. Re: Abort at qm_debug_frag function (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 03 Apr 2013 14:18:18 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] please help with some transformations. > To: sr-users(a)lists.sip-router.org > Message-ID: <515C726A.10104(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello Julian, > > On 04/03/2013 01:54 PM, julian arsanches wrote: > > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the sip > > uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com <mailto:sip%3A123xxx@domain.com> >>> > > changed to sip:xxxx@dmain.com <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so many > > errors that kamailio don't start, > > There is a core convenience function that can help with this, as long as > you're manipulating the request URI: > > strip(3); > > You referred to "the SIP URI", which, of course, is ambiguous since a > SIP request has many SIP URIs. > > -- Alex > > -- > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > United States > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ > > > > ------------------------------ > > Message: 2 > Date: Wed, 03 Apr 2013 20:18:44 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] please help with some transformations. > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <515C7284.5060905(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > if you just need to remove first 3 digits, use: > > strip(3); > > If the number of digits to strip is variables, use strip transformation: > > - http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#sstrip_len > > If does not work for you giving errors, paste here what you have in the > config for that part. > > Cheers, > Daniel > > On 4/3/13 7:54 PM, julian arsanches wrote: > > Hi am new to kamilio , but able to make calls, i need to do something > > really simple if someone can give me an example of it i will be > > gratefull. > > > > just before the call gets relayed i need to delete a prefix from the > > sip uri so it gets transformed to the same uir but less 3 digits. > > > > for example sip:123xxx@domain.com > > <mailto:sip%3A123xxx@domain.com> >>> changed to sip:xxxx@dmain.com > > <mailto:sip%3Axxxx@dmain.com> > > > > i tryed with transformations but still too new and does give me so > > many errors that kamailio don't start, > > > > tryed subs_uri but i dont know how to tell it to just take the first > > tree digits, it is usually done for adding prefixes. > > > > please help. > > > > thank you in advance. > > > > on my scenario i am having an asterisk that i dont control sending me > > an invite for with a prefix i need kamailio to forward that uri to > > another server minus the prefix. > > > > thank you. > > > > > > _______________________________________________ > > 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 > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > >

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please help with some transformations.
by julian arsanches 03 Apr '13

03 Apr '13

Hi am new to kamilio , but able to make calls, i need to do something really simple if someone can give me an example of it i will be gratefull. just before the call gets relayed i need to delete a prefix from the sip uri so it gets transformed to the same uir but less 3 digits. for example sip:123xxx@domain.com >>> changed to sip:xxxx@dmain.com i tryed with transformations but still too new and does give me so many errors that kamailio don't start, tryed subs_uri but i dont know how to tell it to just take the first tree digits, it is usually done for adding prefixes. please help. thank you in advance. on my scenario i am having an asterisk that i dont control sending me an invite for with a prefix i need kamailio to forward that uri to another server minus the prefix. thank you.

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Kamailio World Updates - April 3
by Daniel-Constantin Mierla 03 Apr '13

03 Apr '13

Hello, just a bit less than two weeks to the start of Kamailio World Conference, we are gearing up and already started preparing everything for the event. The agenda with presentations is available at: - http://conference.kamailio.com/k01/schedule/ Details about the speakers are available at: - http://conference.kamailio.com/k01/speakers/ Regarding Kamailio project, even there was an extended weekend with public holidays in around Easter, three new modules showed up - stun, dnssec and sipt - and a lot of new code made its way to old modules, among most active being snmpstats. If you haven't registered yet for the Kamailio World conference, be aware of that standard registration ends this week, on April 5, 2013. Actually the registration might be completely closed as it is very likely that the event will get full booked, even we extended the capacity about one week ago. It is a great opportunity to meet other hundreds people from real time communications services. Do not delay registration if you plan to attend the event, the form and more details at: - http://conference.kamailio.com/k01/registration/ Several companies coming with stands for presenting products, services and demos will welcome you to stop by just in front of the conference room - Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Evariste Systems to drop Kamailio market, become Acme Packet VAR
by Alex Balashov 02 Apr '13

02 Apr '13

For immediate release: ATLANTA, GA (1 April 2013)--Evariste Systems LLC, an Atlanta-based consultancy specialising in Kamailio-based VoIP infrastructure solutions for the ITSP and CLEC market, has announced that beginning in the second quarter of 2013, it will be abandoning its Kamailio-based technology portfolio to focus on its new role as a preferred VAR (Value Added Reseller) for Acme Packet (NASDAQ:APKT). "It is with a heavy heart that we abandon five years of Kamailio-oriented work and the Canonical SIP Routing Platform product derived from it," said Alex Balashov, the principal of the company. "However, the reality is that investment in open-source VoIP technology is a dead end. From a technological point of view, we have lagged very badly in meeting the needs of today's sophisticated VoIP market, and it's time to cut our losses. Asterisk, Kamailio, FreeSWITCH--all this stuff just hasn't kept up with the pace of evolution of 3GPP, ETSI, and ITU standards. We are tired of saying 'sorry, we don't support IMS or H.323' to our resultingly dwindling customer base. Does anyone actually run an all-SIP network?" Starting in early April, Evariste will begin providing value-added consultancy related to the implementation of the Acme Packet Net-Net Session Director. In Balashov's view, "the Net-Net SD is the only product capable of meeting the perimeter security, routing and peering needs of today's VoIP service delivery environment." Fred Posner, the director of Team Forrest, a Palner Group integration and consultancy operation based in the Jacksonville, Florida area, agreed: "SIP is a tiny piece of the telephony puzzle. The big boys of ClueCon [an interoperator revenue-sharing consortium] want DIAMETER-based interdomain peering policy control, H.323, MGCP, and IMS. IMS is pretty much how VoIP architecture is done now. We got out of the Asterisk business just in time, right before Mitel swallowed the PBX world. I'm glad to see Evariste is finally seeing the light, and I'm sure its shareholders are too." Posner also believes Evariste's lack of support for TDM interfaces accounted for dwindling market share. "Have you seen CSRP? It's SIP in, SIP out. Real inter-LATA haulers and application service providers use TDM and leave SIP for things like voicemail. I can't plug my DS3s into a SIP proxy, so I just don't think there was any real demand for the sort of thing they were doing." Noting Oracle's US$2.5bn acquisition of Acme Packet in early February, as well as its more recently announced buyout of Tekelec, a Siris Capital Group portfolio company, Balashov remarked: "The obvious shift to an Oracle-centric telephony paradigm was a kind of validation, if you will, of our decision to unload our dead weight and sign on to the revolution in unified communications." Sean McCord, of CyCORE Systems, an Atlanta-based software consulting house and long-time Evariste creditor, agreed that there was a natural synergy between Evariste's shift to Acme Packet and Oracle's dominance of telephony infrastructure. "Oracle is a forward-thinking telecom pioneer," McCord said. "The telephone is Oracle, and Oracle is the telephone." Balashov also noted that a tightening regulatory environment and new consumer protection rules helped hasten the decision to embrace the more professionalised Acme Packet product portfolio. John Knight, Senior Engineer at Hendersonville, NC-based Ringfree Communications, one of Evariste's oldest channel partners, said: "As one of Evariste's long-time disties, we were jittery about exposure to CALEA and the QA requirements of large call centers. We tried to make do, but at some point we just had to put the relationship on stop. I'm all in favour of open, but there's just no open-source software out there that does call recording, and that's the bottom line for us. In the end, we had to restructure some debt just to get bondholders to let us source a proprietary solution on tick." In a thematically related move, Evariste will be dropping its heavy use of the open-source PostgreSQL database manager for its rating and reporting tools. "The business case for standardising on Oracle's databases could not be clearer. With Oracle Database 11g's support of warehousing and OLTP, the real mystery is why we didn't go there sooner," said Balashov. Carlos Alvarez, a director at Televolve, a growing Phoenix-area VoIP operator, recently spearheaded a move away from Evariste's PostgreSQL- based call detail record (CDR) storage solution to one running atop Microsoft SQL Server 2008. Alvarez commented: "Evariste had a nice idea, in a cute, David-and-Goliath kind of way, but we're processing over five hundred phone calls a day now. Are we really going to store those kinds of volumes in an open-source database? Might as well just put it all in flat text files at that point. Phone service is an uptime game. You can't compromise on this stuff. What if someone needs to call 911?" Asked to summarise his expectations, Balashov said: "I hope this turns us around in a big way. We were wrong to think that nobody cared about stuff like P-CSCFs, or that you could deliver even rudimentary VoIP to the premise without the expansive feature set of a comprehensive solution like the Net-Net SBC. I can only hope the market forgives us for betting on 'SIP Express Router' and its ilk back in the day, and gives us a chance to do it right in round two." Fred Posner, of Team Forrest, added: "Besides, if you look at the Git repository, Kamailio hasn't had any code contributions in at least five years. It seems everyone's figured out this pure SIP stuff is defunct and hokey."

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ClueCon 2013 - Call For Speakers
by Michael Collins 02 Apr '13

02 Apr '13

[image: https://mail-attachment.googleusercontent.com/attachment/u/0/?ui=2&ik=43fec…]<http://www.cluecon.com> ClueCon - the open source IP communications conference by developers, fordevelopers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea fora technical presentation for ClueCon 2013 then we would like to hear about it. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon *loves *technical presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools, especially those in a production environment - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations - We are especially interested in WebRTC-related talks and demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include the following items: - Working title - Brief description of the talk (abstract) - Name of the presenter Don't delay! There are a limited number of openings. We will contact you as soon as your talk has been approved and will inform you of the scheduled time. ClueCon 2013 Registration Information ClueCon 2013 registration is now open!. Visit the registration page<http://www.cluecon.com/register?cc12cfs> for details. Be sure to book your room at the Hyatt Chicago Magnificent Mile<http://www.cluecon.com/hotels/>and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2013. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2013. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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alphabetic indexes generated for development version
by Daniel-Constantin Mierla 02 Apr '13

02 Apr '13

Hello, I generated the alphabetic indexes for development version, there are already new modules and code since release of 4.0. If anyone spots missing elements (mod params, functions, etc...), write back here so we fix the issue (could be wrong formatted xml docbook file for module readme). They are available at: - http://www.kamailio.org/wiki/#alphabetic_indexes Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

Are there any Blind Call Forwarding examples available for Kamailia 3.3.0?
by alans126 02 Apr '13

02 Apr '13

I've just configured Kamailia and users 10001,10002,10003 can give each other a call successfully. Now I want to forward every call to 10002 to 10003. The only example I can find is for OpenSER 1.X, the configure file seems different from Kamailia 3.3's. I tried to add the following code to the configure file and some other add_module parts to the other one, but it doesn't work... if(avp_db_load("$ru","$avp(s:callfwd)")) { $ru = $avp(s:callfwd); route(RELAY); exit; }. Can anyone provide me with the kamailia.cfg and kamctlrc files which implemented call forwarding unde Kamailia 3.3, please? I'm a newbie so I really hope to get your help. 2013-04-02 alans126

1 0

Support
by Maritza y Cristian Zuÿfffff1iga 02 Apr '13

02 Apr '13

Dear Sr, Is it possible to use QuteCom softphone with iptel.org SIP service? Regards Cristian A. Zúñiga

1 0

Is cvs repository of sip_route still alive?
by Yuming Zheng 02 Apr '13

02 Apr '13

Hi, export CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser cvs login OK, when try checkout the sip_route cvs co sip_router cvs [checkout aborted]: cannot stat /cvsroot/ser/locks: No such file or directory Although cvs view is working. I want go check some logs in the repository. BR, Frank.zheng

3 3

Wiki site registration updates
by Daniel-Constantin Mierla 01 Apr '13

01 Apr '13

Hello, as of yesterday, creation of new accounts for Kamailio's wiki site requires to answer a project related question. Captcha was useless as spam bots were lately going through it easily, creating accounts in a rate of approx 50 new registrations per day. The extra question is asked just after CAPTCHA, see it at: - https://www.kamailio.org/wiki/start?do=register Hopefully the questions are simple enough to allow good people to register and difficult enough for spambots to give up. It is not a very sophisticated system, let's see if there will be any efforts in reverse engineering to break in with bots. So far no new spammer account. If they will succeed, at least they learn something useful. If anyone has difficulties creating wiki accounts, write an email to sr-dev mailing list and it will be investigated. Cheers, Daniel PS. This registration system will last, is not for April 1. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Kamailio + ABills
by Victor V. Kustov 01 Apr '13

01 Apr '13

Hi. Can someone tips for Kamailio + ABillS works? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux

2 2

Yet another IPv6 question related with rtpproxy ipv4/ipv6 bridging
by Miguel Baptista 31 Mar '13

31 Mar '13

Hello, I have been doing some tests with kamailio and IPv6. My initial setup was IPv6 only and now am I extending it to a dual-stack environment. Well, and now I am starting to face some (interesting) challenges. So, the first step in the dual-stack environment was to install RTP Proxy and configured kamailio to use it. With this setup, the UAs locally registered were able to communicate with each-other no matter with address family (IPv4/IPv6) they were using. So far so good. But now I want to extend my tests a bit more ... I want to communicate with the "outside world" (using ENUM and domain based SIP URIs). Do I have a way to know if the "destination" is IPv4 or IPv6? Because I need that information in order to properly bridge the calls on rtpproxy. I tried to use the onsend_route but it didn't work. I mean, I am able to know if the "next-hop" is IPv6 or IPv4 but it seems that it is "too late" to use rtpproxy. BTW ... I am assuming that if the "next-hop" is IPv6 then the final user agent will also be IPv6 (the same for IPv4) Should I use some other approach? For example, failure route instead? Does anyone have a similiar setup? How are you solving this issue? - Miguel Baptista

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SIP contact matching - location db ops
by Daniel-Constantin Mierla 30 Mar '13

30 Mar '13

Hello, apparently all the db operations to update/delete contacts are done by matching on username, contact, call-id and domain (when use_domain=1) -- it is like this for way many years, therefore needs some debate if/how to change. While the module has a matching_mode parameter for various contact matching algorithms, it is used for in-memory matching only. - http://kamailio.org/docs/modules/stable/modules/usrloc.html#matching-mode IIRC, for initial SIP specs (with no GRUU extensions, etc), aor contact address should be used for matching. Call-Id has a relevance only when CSeq is lower than previous received REGISTER with same call-id. Do I miss any specs requirements? Following the report on tracker FS#278, matching on path value for db ops will be probably very inefficient. Matching on call id for this case should be sufficient if the REGISTER comes from different UA. Can it happen that same UA sends same REGISTER via two paths? With this question I mean if anyone is aware of an UA able to do this. Relying on previous hop address to use in the db key, would not be enough, as the different path can be in the hops before. I plan to keep for now call-id for db operations when matching mode involves path, one way to fix it properly could be usage of ruid as db key, but needs further analysis. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 3

Help with SIP over Websocket audio call
by Brad Johns 30 Mar '13

30 Mar '13

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. I don't know how to debug this further to find out what the problem is. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. With this JsSIP, I can do chat through Kamailio SIP over WebSockets. With this Kamailio server, SIP User Agent Clients work just fine to register and place SIP call with audio. It's just that WebRTC audio calls don't work with JsSIP sample application with Kamailio 4.0 websocket module. Kamailio websocket configuration borrowed from: https://gist.github.com/jesusprubio/4066845 Any help debugging this appreciated. Brad

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Help with SIP over Websocket audio call: 405 Method Not Allowed
by Brad Johns 30 Mar '13

30 Mar '13

Hi, New to Kamailio. I have my Kamailio 4.0 server with websocket support, and the users can register using the JsSIP Tryit sample WebRTC application. They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over websockets module. However, after registration, the users can't place an audio call. I see no ringing on the remote browser. Can anyone help with clues or debug? In Debug log I can see the websocket ws_frame.c decode the websocket message into SIP, and I see normal SIP call flow for an INVITE. However, nothing indicating a call. I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting an error response to browser UA of "405: Method Not Allowed". I've isolated it down to the this snippet in the kamailio.cfg for route[LOCATION]: $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", "TEST: Method Not Allowed"); exit; } } The switch case is returning -2, for some reason. Any help in debugging this appreciated.

3 5

sems.conf
by Maaz Bin Mahmood 30 Mar '13

30 Mar '13

Hi there, I am trying to try SEMS but after make install, file sems.conf is not available in /usr/local/etc/sems/ Can you please suggest a solution? as tutorial here ( http://ftp.iptel.org/pub/sems/doc/current/howtostart_noproxy.html ) asks to change sems.conf as a first step. Regards, Max

2 1

carrierroute module
by Angel Angelov 29 Mar '13

29 Mar '13

Hello, When I tried to start kamailio-4.0 with carrirroute module include, the follow error appeared 0(8732) ERROR: [sr_module.c:572]: ERROR: load_module: could not open module : /usr/local/kamailio-4.0/lib/kamailio/modules/carrierroute.so: undefined symbol: cfg_getnint . Do you need more information? Thanks in advanced. Regards. Angelov?

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IPv4 / v6 request to global parameter "TOS"
by Klaus Feichtinger 29 Mar '13

29 Mar '13

Hello list, in a bridging scenario with kamailio 3.3.4 and rtpproxy 1.2.1 for bridging signalling and media from an IPv4 to an IPv6 network and vice versa I found that the TOS value, which is set in kamailio.cfg, is used for IPv4 packets only. IPv6 packets have the traffic class value set to the default value 0x0. In other words: kamailio doesn´t use this variable for IPv6 packets. In the cookbook (http://www.kamailio.org/wiki/cookbooks/3.3.x/core#tos) I haven´t found any hint that it _is_ limited to IPv4 only ("...for the sent IP packages"). I know that the name "tos" may be misleading, as the original definition was outdated by dscp+ecn, but it was/is working fine now. However, as IPv6 is using dscp+ecn, too, I wonder if the tos variable should support IPv6 packets, too. Could anybody give me a hint? Is there maybe an alternative way to prioritise SIP in IPv6 with kamailio? Thanks in advance, Klaus Feichtinger

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wiki accounts cleanup
by Daniel-Constantin Mierla 29 Mar '13

29 Mar '13

Hello, as there were many spamming accounts created in the wiki, used to add inappropriate content (http://www.kamailio.org/wiki/start?do=recent), I went through them and deleted those having more like random name and random id for email addresses. Hopefully I haven't deleted any valid account, but if you try yours and does not work, try to create it again. If you still have troubles, write me mentioning the account id. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Re: [SR-Users] Iimitations of xcap server in kamailio
by David Thomson 29 Mar '13

29 Mar '13

Thanks Peter! I'll try that out!! Is there a doc that goes through how to tweak Kamailio into production mode? What about having two Kamailio servers sharing one db and allowing calls between the two servers to be completed? That's another hurdle I've yet to get over. Ttyl, Dave ------Original Message------ From: Peter Dunkley To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Iimitations of xcap server in kamailio Sent: Mar 28, 2013 10:20 PM There are a number of core parameter and modparam values you need to increase from their defaults in order to be able to handle large HTTP requests (and therefore large XCAP documents). For example, * tcp_rd_buf_size (core parameter) * tcp_wq_max (core parameter) * sql_buffer_size (core parameter) * buf_size (xcap_server modparam) Regards, Peter > Has anyone else noticed issues when adding a large number of contacts to > your buddy list? I was using Jitsi and at about 40 contacts I was > encountering issues in that the subsequent contacts would not get added to > my buddy list. > > Ttyl, > Dave > > > > _______________________________________________ > 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 > -- Peter Dunkley Technical Director Crocodile RCS Ltd _______________________________________________ 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

1 0

500 ERROR Reloading data when reloading hash table using FIFO
by kawarod 29 Mar '13

29 Mar '13

Dear list users, I encounter an issue when reloading a hash table using FIFO command. My hash is getting a list of "Username / Serial Number" from SQL for each phone connecting to Kamailio. Then I use this hash to authenticate the phone based on a X-SerialNumber attribute. This hash has to be updated once or twice a day, to accomodate new users or a change of the Phone. I'm using the following command for reloading: /opt/kamailio/sbin/kamctl fifo sht_reload hash_cpe_serial Everything is working fine, but after 25-26 reload I have an error like this: 500 ERROR Reloading data The only way to recover is do a restart with ./kamctl restart The hash size has approximatively 22 000 users. The hash has been defined like this in the configuration file: modparam("htable", "htable", "hash_cpe_serial=>size=14;dbtable=subscriber;") When using kamcmd, i have this: kamcmd> htable.listTables { name: ipban dbtable: dbmode: 0 expire: 300 updateexpire: 1 size: 256 } { name: hash_cpe_serial dbtable: subscriber dbmode: 0 expire: 0 updateexpire: 1 size: 16384 } When trying to dump with kamcmd, I have an error: kamcmd> htable.dump hash_cpe_serial error: 500 - Internal server error processing '{': buffer too small (overflow) (-2) But it works fine with: /opt/kamailio/sbin/kamctl fifo sht_dump hash_cpe_serial I have upgraded from 3.2.2 to 4.0.0 to be sure that it was not relative to the old version, but without success. Any idea on how to correct this. Kind regards, rod

1 1

kamailio 4.0.0; km_append_branch problem
by Pavel Miskov 29 Mar '13

29 Mar '13

Hi all, I installed Kamailio 4.0.0 from source an tried configuration from Kamailio 3.3.2. problem is with km_append_branch function. Not starting Kamailio400: invalid configuration file! -e 0(20346) ERROR: <core> [cfg.y:3431]: cfg. parser: failed to find command km_append_branch 0(20346) : <core> [cfg.y:3570]: parse error in config file /usr/local/kamailio400/etc/kamailio/kamailio.cfg, line 2448, column 22: unknown command, missing loadmodule? ERROR: bad config file (1 errors) configuration: ------------------ mpath="/usr/local/kamailio400/lib64/kamailio/modules_k/:/usr/local/kamailio400/lib64/kamailio/modules/" loadmodule "kex.so" ... km_append_branch(); route(RELAY); exit; } ... Please advise Best regards Pavel

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Iimitations of xcap server in kamailio
by David Thomson 29 Mar '13

29 Mar '13

Has anyone else noticed issues when adding a large number of contacts to your buddy list? I was using Jitsi and at about 40 contacts I was encountering issues in that the subsequent contacts would not get added to my buddy list. Ttyl, Dave

2 1

Homer / Kamailio / Freeswitch
by Antonio Teixeira 28 Mar '13

28 Mar '13

Hello. First of all i just did a fresh install of Homer and this thing is perfect only really missing the RTP analysis. Anyway I have a Freeswitch PBX with several internal profiles and also external profiles binded into external Ips ( Public not Private). Capture on the private network profiles works flawlessly when i try to track public interface , calls going into the VOIP Operators , there are no entries .... So : Public call data inside the DB , No Freeswitch sending data into Kamailio , Yes TCPDUMP on Kamailio Shows SIP Data from the PBX Interface , Yes , Kamailio complains with Mar 28 08:30:27 debian /usr/local/sbin/kamailio[23709]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#015#012> Mar 28 08:30:27 debian /usr/local/sbin/kamailio[23709]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed Problem Scenario : A call from a phone inside the network 1.1.1.0 Into the PBX 1.1.1.1 is tracked by Homer a call from the external IP Binded into the PBX lets say something like 80.0.80.8(External Public IP) , Call not tracked and Kamailio complains with parse_msg_error. Helper Data : Uname Linux debian 2.6.32-5-amd64 #1 SMP Mon Feb 25 00:26:11 UTC 2013 x86_64 GNU/Linux LSB Distributor ID: Debian Description: Debian GNU/Linux 6.0.7 (squeeze) Release: 6.0.7 Codename: squeeze Kamailio as of TODAY from GIT Homer as of TODAY from GIT WebHomer 3.3.1 From GIT Also Using the SOFIA SIPTRACE , problem shows both using the global sip trace or individual profile siptrace. Thanks for your time Antonio

2 1

base64 transformation
by Andreas Granig 28 Mar '13

28 Mar '13

Hi guys Maybe I missed something, but looks like there is no base64 encode/decode transformation in PV, right? I think we're going to add one, as it might be handy to do such transformations directly from within the config file. One case where this could be used is that some UAs are picky in regards to URIs as URI params, e.g. P-Foo: <sip:alice@example.org;x='sip:1.2.3.4:1234'> which could be changed into P-Foo: <sip:alice@example.org;x=c2lwOjEuMi4zLjQ6MTIzNAo=> which might be understood more easily (although it's still just a work-around for broken UAs, but that's how life is). Any objections against such a transformation? Andreas

3 2

Tag maximum size/length
by Grant Bagdasarian 28 Mar '13

28 Mar '13

Hello, Is there a maximum length/size defined for the From and To Tag? The RFC states: "it MUST be globally unique and cryptographically random with at least 32 bits of randomness". Though I can't find anything in the RFC about the maximum. The sip_capture table stores the to_tag and from_tag in a varchar(64). Does this mean the max is 64? Source: https://code.google.com/p/homer/source/browse/sql/create_sipcapture.sql Thanks, G

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Need help with Kamailio+RLS configuration.
by Aleksandrs Semenenko 27 Mar '13

27 Mar '13

Hello everyone, I am running Kamailio 3.3.2 server and I need to configure it to support RLS to avoid big amounts of SUBSCRIBE requests from clients. By this moment I have managed to configure XCAP server and learned the XML structure for press-rules, rls-services and resource-list documents. Also I have successfully put rls_update_subs and rls_handle_subscribe calls to kamailio.cfg, so now when I PUT new rls-services or resource-list document to XCAP I can see these calls triggered. The problem is that on client side I always get empty notification XML: <?xml version="1.0"?> <list uri="sip:user1-list@sip.server" xmlns="urn:ietf:params:xml:ns:rlmi" version="1" fullState="true"> <resource uri="sip:user2@sip.server"/> <resource uri="sip:user3@sip.server"/> </list> As I understand, user2 and user3 must have added user1 to pres-rules document? Correct? So it is… I assume there should be a call to rls_handle_notify function somewhere in kamailio.cfg, but where exactly? I have tried to add it after SUBSCRIBE condition, but it is never called in fact: if( is_method("SUBSCRIBE")) { xlog("BEFORE rls_handle_subscribe"); $var(ret_code)= rls_handle_subscribe(); xlog("AFTER rls_handle_subscribe"); if($var(ret_code)== 10) handle_subscribe(); t_release(); } else if (is_method("NOTIFY")) { xlog("BEFORE rls_handle_notify"); rls_handle_notify(); xlog("AFTER rls_handle_notify"); } Any comment would be appreciated! Regards, Alex.

1 1

SIP hard phones with TLS client certificate
by Daniel-Constantin Mierla 27 Mar '13

27 Mar '13

Hello, I am trying to build a list of SIP hardphones that have support for TLS with client side certificate (i.e., import custom ssl certificate in the phone specific for each SIP account). Anyone that can report back the vendor and model (eventually the minimum/current firmware version)? Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Implementing encryption
by Techie Sup 27 Mar '13

27 Mar '13

Hello, Any successful implementation of adding some customized security (encryption) over sip and rtp in kamailio? (non TLS) Also looking for assistance on the concept of rtp over tcp using kamailio and rtpproxy. Warm Regards Selvam -- Sent from my mobile device

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Re: [SR-Users] How to enable TLS in Kamailio ?
by David Thomson 27 Mar '13

27 Mar '13

Didn't read the whole thread, but, in some clients you need to specify the transport is TLS and the port is 5061 (assuming you use the default). Hth. Ttyl, Dave -----Original Message----- From: Khoa Pham <onmyway133(a)gmail.com> Date: Thu, 21 Mar 2013 03:52:12 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] How to enable TLS in Kamailio ? @Moacir, thanks 1. You said that " then use "listen="my ip address" and all enabled services will be bound to the especified IP address." So which port does Kamailio listen for TCP and TLS? How does client know which port to connect to Kamailio ? On Thu, Mar 21, 2013 at 12:41 AM, Moacir Ferreira <moacirferreira(a)hotmail.com <mailto:moacirferreira@hotmail.com> > wrote: Hummm. When I start playing with Kamailio I had some problems that were related to the compilation process not really to the final product. Now, if you comment out the listen statement then all server interfaces will listen for all enabled services (SIP-UDP, SIP-TCP and SIP-TLS if you enabled it). AS a troubleshooting suggestion, just comment out the "listen" statement and all enabled services will bind to all available IP interfaces. If it works, then use "listen="my ip address" and all enabled services will be bound to the especified IP address. Best regards, Moacir ---------------- From: oej(a)edvina.net <mailto:oej@edvina.net> Date: Wed, 20 Mar 2013 12:43:06 +0100 To: sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> Subject: Re: [SR-Users] How to enable TLS in Kamailio ? 20 mar 2013 kl. 10:33 skrev Khoa Pham <onmyway133(a)gmail.com <mailto:onmyway133@gmail.com> >: Hi Olle, I follow these 2 tutorials 1. http://nil.uniza.sk/network-security/tls/configuring-tls-support-kamailio-3… which only describes to listen on tls listen=tls:158.193.139.51:5061 <http://158.193.139.51:5061/> 2. http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates which only describes to listen on udp and tcp listen=udp:<ip-address-for-receiving-sip-requests>:5060 listen=tcp:<ip-address-for-receiving-sip-requests>:5060 It is not until I listen on both TCP and LTS does it work. You document http://kamailio.org/docs/modules/4.0.x/modules/tls.html seems to lack of these "listen" Right. But it's in the core cookbook. We should propably add the listen to the TLS docs too. Thanks! /O On Wed, Mar 20, 2013 at 4:10 PM, Olle E. Johansson <oej(a)edvina.net <mailto:oej@edvina.net> > wrote: 20 mar 2013 kl. 07:55 skrev Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com> >: Hello, when tls module is installed, a self signed pair of certificate-private key is generated in /usr/local/etc/kamailio If you need one that is signed by a trusted CA (e.g., Verisign), you will have to buy it. Cheers, Daniel On 3/19/13 8:08 AM, Khoa Pham wrote: Hi, I want to enable TLS in Kamailio, as in here http://kamailio.org/docs/modules/stable/modules/tls.html But how can I get the certificate and private key ? The documentation for the TLS module actually includes a quick howto. http://kamailio.org/docs/modules/4.0.x/modules/tls.html What part of this needs clarification? Please help us make the documentation better if there are parts you do not undertand or isn't explained. Thanks, /O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Khoa Pham HCMC University of Science Faculty of Information Technology _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Khoa Pham HCMC University of Science Faculty of Information Technology

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kamailio > user agent client...
by Douglas Adami 27 Mar '13

27 Mar '13

hello guys, I'm new here, already working with Asterisk for some time and decided to learn more about kamailio, I'm studying your settings and features and want to start a project to integrate with Asterisk, but leaving the asterisk for voicemail and IVR. http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb I have a question, can I use kamailio as client for the various gateways and voip providers, saw that possibility in module UAC, but how do I authenticate to multiple servers? anyone have any sample settings? Hugs to all! Doug

2 1

SIP over WebSockets error: bad request
by Brad Johns 26 Mar '13

26 Mar '13

Hi, Wondering if anyone could please help me debug what is wrong with my kamailio.cfg for Websockets support. I'm using JsSIP sample application, and I've compiled Kamailio 4.0 for WS support. My config is borrowed from link this link [1]. Wondering if anyone can help me debug this. Error is bad request, from syslog: Mar 26 19:26:42 ace /usr/local/sbin/kamailio[25774]: ERROR: <core> [tcp_read.c:1296]: ERROR: tcp_read_req: bad request, state=7, error=4 buf:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:8888#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:8888#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012 Any ideas? Can send my kamailio.cfg if necessary. [1] Kamailio sample config for WebSockets https://gist.github.com/jesusprubio/4066845

2 3

Dialog module - shared state
by Charles Chance 26 Mar '13

26 Mar '13

Hi all, I was wondering, how are others sharing dialog module (particularly profile) state between multiple servers? Thanks in advance, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

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Kamailio World Updates - March 26
by Daniel-Constantin Mierla 26 Mar '13

26 Mar '13

Hello, the agenda is pretty much set for Kamailio World Conference, see it at: - http://conference.kamailio.com/k01/schedule/ The event starts in three weeks from now, we are glad to be able to host a great group of speakers. There are well known names in the industry able to share insights of the telecom and RTC/RCS market as well as talks about new projects that have potential to grow and help building new services. Check the speakers list at: - http://conference.kamailio.com/k01/speakers/ Regarding Kamailio project, two weeks after release of v4.0.0, the development branch is very active, with two new modules already imported there (app_java - embedded execution of java code; cnxcc - call control/prepaid system module), other new modules in developers personal branches and new features to old modules (e.g., snmpstats, usrloc). If you haven't registered yet for the Kamailio World conference, be aware of registration timeline updates - standard registration period has been shortened, last date for it being April 5, 2013. As the number of participants exceeded initial estimation, the company taking care of offering drinks and meals needs to know attendees' details more in advance. Registration form and more details at: - http://conference.kamailio.com/k01/registration/ Take in cosideration also the companies coming with demos, presenting products or services: Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. You may want to reserve some time for private discussions with them, contact us for further details. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio on AWS
by Felipe Ramos 26 Mar '13

26 Mar '13

Can anyone please point me on a configuration example for kamailio when it is behind NAT, i need to change the contact ip from internal to external on sdp and proxy all media from outside to internal asterisk farm. Please any help will be appreciated. most of documentation available is when client is behind Nat but not when Server behind Nat like is the case with Amazon aws vpc instances. Again thank you before hand.

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Debugging and syslog
by Daniel W. Graham 26 Mar '13

26 Mar '13

Hello, Reaching out for some assistance with debugging, I am not seeing any xlog messages in my log file. If there is any other info I didn't include that would be helpful please let me know. Any ideas are appreciated. Using Kamailio 3.3.2 & Cent OS 6 Rsyslog.conf -------------------- *.info;mail.none;authpriv.none;cron.none;local0.none /var/log/messages # Kamailio local0.* -/var/log/kamailio Kamailio CFG ------------------------------ #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif log_facility=LOG_LOCAL0 loadmodule "xlog.so" Example xlog- xlog("L_INFO", "No more gateways \n"); Thanks, Dan-

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KAMAILIO Config Hot/Live reload
by Omar 26 Mar '13

26 Mar '13

Kamailio Team. Was this feature added already or we still need to restart kamailio after we modify the Kamailio.cfg. O

3 4

str2ip6 bug is also in 3.3.3
by Juha Heinanen 26 Mar '13

26 Mar '13

i tried to compile latest 3.3 and found that it failed to compile due to same str2ip6 bug that was fixed in 4.0: dns_cache.c: In function ��dns_cache_do_request��: dns_cache.c:1894: warning: implicit declaration of function ��str2ip6�� -- juha

2 1

Help improving the modules readmes & docs
by Daniel-Constantin Mierla 25 Mar '13

25 Mar '13

Hello, I thought some people may have spare cycles and are willing to help with the documentation. Here is some updates I would like to get for modules: 1) adding ids for parameters, functions, and other sections of the docbook xml files. Now we generate the alphabetic indexes pointing to the online html readmes, would be better to point directly to the exact section. Practically, for example, sections such as next from async module: <section> <title><varname>workers</varname> (int)</title> should become: <section id="async.p.workers"> <title><varname>workers</varname> (int)</title> To get unique ids over all files, I suggest using following pattern: [module name] [dot] [type] [dot] [title] The [type] should be: - p - parameters - f - functions - m - mi commands - r - rpc commands - s - statistics 2) Add RPC commands to readmes -- some of modules already do, but many don't. There is a perl script that should get the list of RPC commands from source code, but it requires some patch to a perl lib, not working on all systems. Moreover, the output is quite minimalistic, not easy to improve the content or add examples. Last generated index is: - http://www.kamailio.org/docs/docbooks/3.4.x/rpc_list/rpc_list.html 3) Document selects -- although their name are quite suggestive, a bit of text around won't harm. - http://www.kamailio.org/wiki/cookbooks/devel/selects Because 1) and 2) requires git access, the easiest way to contribute is to provide patches to the xml files. If anyone commits to do an extensive work on these tasks in short term, we may eventually arrange for git access. The 3) is in the wiki, everyone can register an account and then update the pages. I would recommend these guidelines to be followed from now on by developers when adding new elements to the module documentation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Acknowlidging BYE in advance using t_reply
by Efelin Novak 25 Mar '13

25 Mar '13

Hi, I'd like to ask you whether it is a good idea to acknowledge a BYE message in advance using t_reply("200", "OK") on my border proxy. Not acknowledged BYE messages by clients cause me problems in the middle of my network. Is there any problem with this behavior from Kamailio side, from script side or from SIP side? Do you have any experience with such logic manipulation? I am aware that this is not a proxy function, but it would solve many problems. My script is as follows: if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... t_newtran(); t_reply("200", "OK"); } route(RELAY); } My test shows that 200 OK from client are absorbed. Also if there is no reply from a client, Kamailio does not generate 408 Request Timeout. I'm using older version kamailio 3.1.0 (i386/linux) 21a375. Thanks for an advice Efelin

2 1

Does Kamailio work with ICE ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, 1. Does Kamailio work with ICE? What does Kamailio do with ICE candidates in SDP ? 2. Does rtpproxy acts as TURN server ? Many thanks. Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 1

Store GeoIP in Accounting
by Ali Jawad 25 Mar '13

25 Mar '13

Hi I am storing accounting entries in the db, and I am also inserting some custom AVP values, however I also need to store the GeoIP value that I am extracting using gip. I did try the below, but it is not working, any other ideas how I could achieve this ? modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;CountryCode=$gip(src=>cc)") Regards

2 1

How to enable TLS in Kamailio ?
by Khoa Pham 25 Mar '13

25 Mar '13

Hi, I want to enable TLS in Kamailio, as in here http://kamailio.org/docs/modules/stable/modules/tls.html But how can I get the certificate and private key ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

7 14

Problems with Mar portion of registration
by mrichardson 22 Mar '13

22 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } }

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Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 22 Mar '13

22 Mar '13

Hi Barry, Thanks for trying to help me out. Unfortunately your suggestion didn't work out. Not sure why. I tried a few other functions like t_relay_to. Guess I still have a lot to learn about this kamailio stuff in more complex setups. Any other ideas, anyone? Ttyl, Dave -----Original Message----- From: Barry Flanagan <barry(a)flanagan.ie> Date: Thu, 21 Mar 2013 12:01:18 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name On 20 March 2013 03:05, David Thomson <jdavidthomson(a)hotmail.com <mailto:jdavidthomson@hotmail.com> > wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com <http://siptest.testdomain.com> with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Hi I think you can use the t_replicate function within your register route so that each server will replicate the registration request to the other. I did this many years ago on OpenSER using the following just before the consume_credentials() line. On each of the servers you want to check that the src_ip is not that of the OTHER sever so they don't keep sending back and forth forever :-) So in the example below 10.0.1.1 is the IP of the other server. --------8<----------------------------------------- if (!src_ip==10.0.1.1) { t_replicate("udp:10.0.1.1:5060 <http://10.0.1.1:5060> "); xlog("L_INFO", "Replicating $ct, URI = $ru"); }; consume_credentials(); --------8<----------------------------------------- Have a look at http://www.kamailio.org/docs/modules/3.3.x/modules/tm.html#t_replicate Hope this helps. -Barry

1 0

GeoIP Issue
by Ali Jawad 21 Mar '13

21 Mar '13

Hi I did setup geoIP module and got the latest geoipcity free dat file and csv file and did setup all parameters in kamailio, when doing a test the instead of showing the country name the code is showing the region name, while region name is empty "I only need the country name", see : Config : if(geoip_match("$si", "src")) xlog("L_ERR","SIP message from: $gip(src=>cc) $gip(src=>regn) $gip(src=>regc)\n"); Log : SIP message from : A2 <null> <null> >From the CSV file of geoipcity just for cross reference : locId,country,region,city,postalCode,latitude,longitude,metroCode,areaCode 190132,"FR","A2","Lézardrieux","",48.7851,-3.1059,, As you can see above A2 is the region it is the 3rd column, src=>CC should show FR instead of A2. Working on Kamailio 3.3.4. Is there anything I missed here ? Regards

1 2

2 kamailio servers sharing one database and one dns name
by David Thomson 21 Mar '13

21 Mar '13

hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows:User 1 registers to Server1User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060>...ignoring. Any ideas? ttyl,Dave

3 2

New developer: Hugh Waite
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

New developer: Carlos Ruiz Díaz
by Daniel-Constantin Mierla 21 Mar '13

21 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Carlos Ruiz Díaz - he is going to add and maintain the credit control module, that was announced earlier this year on mailing list - the module is part of a framework that has three parts: - The module itself: https://github.com/caruizdiaz/cnxcc (to be added in Kamailio git repository) - The web interface: https://github.com/caruizdiaz/cnxcc-web - A helper script: https://github.com/caruizdiaz/cnxcc-db-sync His git commit id is: caruizdiaz My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Need help with port fowards.
by Mini UN 21 Mar '13

21 Mar '13

Hi, was on the chat channel, so emailed as no activity. I have a new Cisco/Linksys EA-2700 and port forwarding is the same as my old Linksys WRT-610N. However my VoIP and website don't work. I tried DMZ aswell. Confused? Unfortunately, this router isn't compatibel for DD-WRT firmware. Thanks if anyone answers.

3 2

kamailio 4.0.0 and websockets
by Kelvin Chua 20 Mar '13

20 Mar '13

Hi guys, i have yet to finish my readings on the websocket standards but just wanted to fire away with this question. is the behavior of protocol conversions between UDP and TCP the same as if you include websockets? i have this twinkle issue (an old SIP stack) Received from: udp:192.168.122.100:5060 INVITE sip:kelvin@192.168.122.1 SIP/2.0 Record-Route: <sip:192.168.122.100;r2=on;lr=on> Record-Route: <sip:192.168.122.100:8080;transport=ws;r2=on;lr=on> Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0 Via: SIP/2.0/WS z173czhz21tk.invalid;rport=54765;received=192.168.122.1;branch=z9hG4bK3818745 Max-Forwards: 16 To: sip:kelvin@192.168.122.100 From: sip:kelvin2@192.168.122.100;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Contact: <sip:kelvin2@z173czhz21tk.invalid ;alias=192.168.122.1~54765~5;transport=ws;ob> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.2.1 Content-Length: 2103 v=0 o=- 3148117784 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio m=audio 55736 RTP/SAVPF 103 104 111 0 8 106 105 13 126 c=IN IP4 192.168.122.1 a=rtcp:55736 IN IP4 192.168.122.1 a=candidate:2625852906 1 udp 2113937151 <cut off> --- +++ 16-3-2013 10:41:25.584732 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100> From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Content-Length: 0 --- +++ 16-3-2013 10:41:25.589231 INFO SIP ::send_sip_udp Send to: udp:192.168.122.100:5060 SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745 To: <sip:kelvin@192.168.122.100>;tag=pxtmo From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq Call-ID: 69hbgnng64at9p07r2j4 CSeq: 9406 INVITE Server: Twinkle/1.4.2 Warning: 302 X340precise "Incompatible transport protocol" Content-Length: 0 Kelvin Chua

2 1

SIP debug.
by Victor V. Kustov 20 Mar '13

20 Mar '13

Good day. How could I debug SIP dialogs? -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.2-calculate GNU/Linux

4 5

NAT & Firewall Issues
by Amr Elsharqawi 20 Mar '13

20 Mar '13

Hello, I am facing NAT & Firewall problems how I can solve this issue I've tried RTPProxy but voice quality is bad. any tutorial, please Amr

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Re: [SR-Users] dialog ka_timer and ka_interval problem
by dolphin 20 Mar '13

20 Mar '13

Hello, Thank you for your reply I follow your suggestions to modify SHM_MEMORY ,PKG_MEMORY value and the system only two uac register. See the program after I start kamailio. shm memory 1G pkg memory 10M sh-3.00# ps -ef | grep kamailio root 23627 1 0 12:57 ? 00:00:00 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio root 23635 23627 0 12:57 ? 00:00:03 /home/pkg/kamailio-3.3.4/sbin/kamailio -f /home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio This problem is still the same last crazy the sending options message for uac But the /var/log/syslog not the same as follows Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x77052ae4 tl->next=(nil) tl->prev=(nil) Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime 2013/3/20 Rinor Hoxha <rinorhoxha(a)gmail.com> > Hi, > I had this "issue" some times ago. It turned out that I needed to increase > the value of SHM_MEMORY, since by mistake I forgot a 0 on the value (1024 > vs. 10240). However this happened only when high loads on system. > I needed also to tune PKG_MEMORY to my needs. ( both on > /etc/default/kamailio ) > > Check your settings and see if this helps. > > Br, Rinor > > > > On Tue, Mar 19, 2013 at 4:07 PM, dolphin <dolphinctk(a)gmail.com> wrote: > >> hi , >> >> i use kamailio 3.3.4 version >> >> I set and load dialog module >> >> modparam("dialog", "ka_timer", 10) >> modparam("dialog", "ka_interval", 20) >> >> use dlg_set_property("ka-src"); >> >> When uac A call uac B connection success , After 20 seconds, sip proxy >> send the OPTIONS message uac A >> >> Crazy sent to fill my bandwidth , I use ngrep tools see sip server >> >> U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 >> OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. >> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. >> To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. >> From: sip:xxxxxxx@xxx.xxx.xxx.xxx >> :5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. >> CSeq: 1 OPTIONS. >> Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. >> Content-Length: 0. >> User-Agent: SIP Server. >> >> and /var/log/syslog >> >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> dialog [dlg_req_within.c:381]: failed to send the BYE request >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) >> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: >> tm [uac.c:338]: t_uac: Error while building message >> >> Please help me >> >> >> >> >> _______________________________________________ >> 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 >> >

1 0

Re: [SR-Users] 2 kamailio servers sharing one database and one dns name
by David Thomson 20 Mar '13

20 Mar '13

Thanks for the idea. After sending the original email I got user2 to call through to user1. Zrtp even was negotiated. So I think usrloc is sort of sharing the registration information but sometimes the call doesn't get sent through under certain circumstances. Maybe it's because user1 is behind a nat and rtpproxy is being used to keep its connection alive while user2 is not nat'd. I think it is almost working. -----Original Message----- From: David | StyleFlare <david(a)styleflare.com> Date: Wed, 20 Mar 2013 03:27:06 To: <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns name I dont think it works like that, obviously the server 1 does not have a connection if the user is registered on server 2. On 3/19/13 11:05 PM, David Thomson wrote: hi, I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode. The scenario is as follows: User 1 registers to Server1 User 2 registers to Server2 User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2: WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060> <udp:xxx.xxx.xxx.xxx:5060> ...ignoring. Any ideas? ttyl, Dave _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

sip-router mailing list
by Nikita Kozlov 19 Mar '13

19 Mar '13

Hello, I don't know if it's something wanted, but the email header "List-Id" has changed for both sr-dev and sr-users. The modifications : List-Id: "SIP Router - Kamailio" -> List-Id: "Kamailio $SER$ - Users Mailing List" List-Id: "Development mailing list of the sip-router project" -> List-Id: "Kamailio $SER$ - Development Mailing List" Regards, -- Nikita

2 1

dialog ka_timer and ka_interval problem
by dolphin 19 Mar '13

19 Mar '13

hi , i use kamailio 3.3.4 version I set and load dialog module modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 20) use dlg_set_property("ka-src"); When uac A call uac B connection success , After 20 seconds, sip proxy send the OPTIONS message uac A Crazy sent to fill my bandwidth , I use ngrep tools see sip server U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296 OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0. To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e. From: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O. CSeq: 1 OPTIONS. Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY.. Content-Length: 0. User-Agent: SIP Server. and /var/log/syslog Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: dialog [dlg_req_within.c:381]: failed to send the BYE request Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370) Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR: tm [uac.c:338]: t_uac: Error while building message Please help me

2 1

Kamailio not increasing cseq
by Camila Troncoso 19 Mar '13

19 Mar '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * * * When I make a call from one client connected to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing though a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , because the second gateway is also in the cloud, but when this happens, FreeSWITCH answer with : “482 Request merged” because it detects that is the same call. This is because the second INVITE has the same Call-ID and same Cseq. Kamailio *is not* increasing CSeq. Is there a way to resolve this? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 5

CentOS 6 crash
by Krishna Kurapati 19 Mar '13

19 Mar '13

Daniel, Thanks for the help. YES after disabling TLS for Kamailio, it is working fine. Is there a solution to have both TLS for Kamailio and SSL for LUA? Thanks Krish Kura

2 1

Multiple calls uac_replace_from()
by Alex Balashov 19 Mar '13

19 Mar '13

Forgive me if I failed to follow a recent discussion to its conclusion, but what is the bottom line on calling uac_replace_from() consecutively, in the course of processing one request? Is it possible, while using restore_mode 'auto'? Is there a correct way to do it, such as calling msg_apply_changes() after the first call but not the second one? If not, what is the alternative? -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

4 7

Kamailio 4.0 crashing in LUA on CentOS 6 - What could be wrong?
by Krishna Kurapati 19 Mar '13

19 Mar '13

36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd474542320 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): sqlops 9(8567) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 36(8606) DEBUG: <core> [sr_module.c:674]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/lib64/kamailio/modules/db_mysql.so] 36(8606) DEBUG: <core> [db.c:213]: using db bind api for db_mysql 36(8606) DEBUG: <core> [db.c:327]: connection 0x7fd4745424f0 found in pool 36(8606) DEBUG: <core> [sr_module.c:892]: DEBUG: init_mod_child (32): app_lua 36(8606) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 39(8615) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 34(8604) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 40(8616) DEBUG: app_lua [app_lua_sr.c:51]: someone probing from lua 41(8617) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 41(8617) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 6, pid 8558 (shutting down?) 0(8543) ALERT: <core> [main.c:788]: child process 8558 exited by a signal 11 0(8543) ALERT: <core> [main.c:791]: core was generated And here is backtrace done. Loaded symbols for /usr/lib64/lua/5.1/ssl.so Core was generated by `/usr/local/sbin/kamailio -L /usr/local/lib64/kamailio/modules/:/usr/local/lib64'. Program terminated with signal 11, Segmentation fault. #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 269 FRAG_END(next)->prev_free=prev; Missing separate debuginfos, use: debuginfo-install lua-5.1.4-4.1.el6.x86_64 (gdb) backtrace #0 qm_detach_free (qm=0x7fd370304000, size=32) at mem/q_malloc.c:269 #1 qm_malloc (qm=0x7fd370304000, size=32) at mem/q_malloc.c:386 #2 0x00007fd47052ae71 in shm_malloc (size=24) at ../../mem/shm_mem.h:262 #3 ser_malloc (size=24) at tls_init.c:285 #4 0x0000003928c5d95e in CRYPTO_malloc (num=24, file=0x3928d24a84 "o_names.c", line=195) at mem.c:306 #5 0x0000003928c6085f in OBJ_NAME_add (name=0x3928d27c52 "RSA-SHA512", type=1, data=0x3928d27c56 "SHA512") at o_names.c:195 #6 0x0000003928cc076a in EVP_add_digest (md=0x3928f84440) at names.c:99 #7 0x000000392b442f18 in SSL_library_init () at ssl_algs.c:165 #8 0x00007fd36ea1d3c9 in luaopen_ssl_core () from /usr/lib64/lua/5.1/ssl.so #9 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #10 0x00007fd36f678a54 in ?? () from /usr/lib64/liblua-5.1.so #11 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #12 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #13 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #14 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #15 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so #16 0x00007fd36f673d56 in lua_call () from /usr/lib64/liblua-5.1.so #17 0x00007fd36f68cf2c in ?? () from /usr/lib64/liblua-5.1.so #18 0x00007fd36f6785d1 in ?? () from /usr/lib64/liblua-5.1.so #19 0x00007fd36f683249 in ?? () from /usr/lib64/liblua-5.1.so #20 0x00007fd36f678a9d in ?? () from /usr/lib64/liblua-5.1.so ---Type <return> to continue, or q <return> to quit--- #21 0x00007fd36f678137 in ?? () from /usr/lib64/liblua-5.1.so #22 0x00007fd36f6781b2 in ?? () from /usr/lib64/liblua-5.1.so #23 0x00007fd36f673b61 in lua_pcall () from /usr/lib64/liblua-5.1.so #24 0x00007fd36f89e69d in lua_sr_init_child () at app_lua_api.c:227 #25 0x00000000004e4aa3 in init_mod_child (m=0x7fd474485060, rank=-1) at sr_module.c:893 #26 0x00000000004e4a24 in init_mod_child (m=0x7fd474485980, rank=-1) at sr_module.c:890 #27 0x00000000004e4a24 in init_mod_child (m=0x7fd474485bf0, rank=-1) at sr_module.c:890 #28 0x0000000000489d15 in fork_process (child_id=-1, desc=<value optimized out>, make_sock=<value optimized out>) at pt.c:347 #29 0x0000000000466dec in main_loop () at main.c:1700 #30 0x0000000000468afe in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566

2 1

Survey on SIP overload control
by Yang Hong 18 Mar '13

18 Mar '13

Dear all, SIP overload may block too many calls unnecessarily. "Requirements for Management of Overload in the Session Initiation Protocol. IETF RFC 5390." http://www.ietf.org/rfc/rfc5390.txt Build-in overload mechanism cannot prevent overload effectively. Therefore, IETF SIP Overload Control (soc) Working Group works on IETF RFC "SIP Overload Control" currently. http://tools.ietf.org/html/draft-ietf-soc-overload-control-12 IETF SIP Overload Control discussion archives http://www.ietf.org/mail-archive/web/sip-overload/current/threads.html#00874 We have published a survey on SIP overload control algorithms (including IETF RFC "SIP Overload Control"). http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI… http://arxiv.org/abs/1210.1505 http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67… We have also used control theorectic approaches to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - redundant retransmission ratio control and round trip delay control (IEEE Globecom 2010 and ICC 2011). "Mitigating SIP Overload Using a Control-Theoretic Approach" IEEE Globecom 2010 http://www.researchgate.net/publication/221284946_Mitigating_SIP_Overload_U… http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5683124&url=http%3A%2… Implementation of implicit SIP overload control in the real system "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach" http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15# "Design Of A PI Rate Controller for Mitigating SIP Overload" IEEE ICC 2011 http://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Contr… http://features.rr.com/article/01vBgtc8TR17z?q=Skype http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=5963029&url=http%3A%2… Best regards, Winston Hong Software Engineer InBay Technologies Inc. Ottawa, Canada K2K 1Y3 http://www.inbaytech.com/

1 0

Kamailio, NAT and RTPProxy
by Amr Elsharqawi 18 Mar '13

18 Mar '13

Dears, Finally I've got Kamailio 4 running on FreeBSD 9 with MySQL. in the beginning it was working fine with NAT enabled, but after I've configured NAT & RTPProxy users can register, ring but voice is very bad. here is my kamailio.cfg #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/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 # # *** To enable nat traversal execute: #!define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _._._._ -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://sip:s510Sg4lrw@localhost/sip" #!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=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:XXXXXXXXX:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" 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_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/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" #!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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:22222") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!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 { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) 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 }

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Runaway TCP Log
by Krishna Kurapati 18 Mar '13

18 Mar '13

Hi, My Log filled to 20G overnight with these Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files Mar 18 09:59:44 sipprod /usr/local/sbin/kamailio[26157]: ERROR: <core> [tcp_main.c:4049]: WARNING: handle_new_connect: error while accepting connection(24): Too many open files^C I looked at the code segment it is here where it is logging. new_sock=accept(si->socket, &(su.s), &su_len); if (unlikely(new_sock==-1)){ if ((errno==EAGAIN)||(errno==EWOULDBLOCK)) return 0; LOG(L_ERR, "WARNING: handle_new_connect: error while accepting" " connection(%d): %s\n", errno, strerror(errno)); return -1; } I am using 3.3.x Is this an issue with Kamailio or something else? Appreciate any help regarding this matter Krish Kura

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Kamailio strange behavior : bad Via, bad port
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. The INVITE goes through Kamailio. In the Kamalio 's log file, I see it say "bad Via", bad port. And it doesnot reply A with 100 Trying not forwarding INVITE to B. The INVITE message (espcially the Via header) is OK. This problem sometimes happens. Do you know why ? Please help. Thanks -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Kamailio not send 100/180 for INVITE !
by Khoa Pham 18 Mar '13

18 Mar '13

Hi, When A calls B. Kamailio will handle A's INVITE message. But I see that B doesnot receive the INVITE. And A doesnot receive any 100/180 privisional message. Why does Kamailio not send 100/180 ? Please help -- Khoa Pham HCMC University of Science Faculty of Information Technology

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How does Kamailio handle INVITE message by default ?
by Khoa Pham 17 Mar '13

17 Mar '13

Hi, After asking many questions, I haven't got any clues about how Kamailio handles INVITE message by default, in terms of modifying c= line in SDP According to rtpproxy flow http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf When client register, SIP proxy will call nat_uac_test() to detected if client is NATed or not, then save this info. When client A calls client B, the INVITE message will go through SIP proxy. Here the SIP proxy can do 3 things (as in section "INVITEs behind NAT" in the pdf). 1. Add an SDP command direction:active to the SDP content 2. Change the c= line to a.b.c.d 3. Force RTP to go through a proxy by changing the c-line to c=IN IP4 address-of-proxy and the m-line to m=audio port-on-proxy RTP/AVP 0 101. When will SER do 2, 3 ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

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Kamailio only allowing traffic with machines in the dispatcher.cfg?
by Benjamin Henrion 16 Mar '13

16 Mar '13

Hi, Does someone know/uses a simple rule so that Kamailio only exchanges traffic with machines in the dispatcher? Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

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Re: [SR-Users] sr-users Digest, Vol 94, Issue 51 - Problem with Mar portion of registration with Kamailio ims modules
by mrichardson 15 Mar '13

15 Mar '13

Date: Fri, 15 Mar 2013 08:44:13 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Problems with Mar portion of registration [IMS] To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <5142D14D.7080808(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello, I haven't played with IMS modules, but you said is causing scscf to die -- is that a kamailio instance? What means 'to die' -- does it crash? If yes, can you get the syslog messages and the gdb backtrace? Cheers, Daniel ============================================ *Response:* ============================================ It appears any way we force the scscf to process a mar causes a crash. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio.pid -m 64 -M 4 -u kamailio -g kamailio'. Program terminated with signal 11, Segmentation fault. #0 async_cdp_callback (is_timeout=<value optimized out>, param=0x7f641b158e70, maa=0x7f641b159040, elapsed_msecs=<value optimized out>) at cxdx_mar.c:414 414 start_reg_await_timer(avlist[0]); //start the timer to remove stale or unused Auth Vectors Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 libxml2-2.7.6-8.el6_3.4.x86_64 nss-pam-ldapd-0.7.5-18.el6.x86_64 pcre-7.8-6.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) syslog: 7(2211) DEBUG: tm [t_lookup.c:1657]: DEBUG: t_lookup_ident: transaction found 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_experimental_result_code: Failed finding avp 7(2211) INFO: ims_auth [cxdx_avp.c:139]: cxdx_get_sip_number_auth_items: Failed finding avp 7(2211) DEBUG: <core> [msg_translator.c:206]: check_via_address(10.12.92.191, 10.12.92.191, 0) 7(2211) DEBUG: <core> [mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called 7(2211) DEBUG: tm [t_reply.c:1547]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 7(2211) DEBUG: tm [t_reply.c:706]: DEBUG: reply sent out. buf=0x7f64e586b950: SIP/2.0 403 Forbidde..., shmem=0x7f64de66c1b0: SIP/2.0 403 Forbidde 7(2211) DEBUG: tm [t_reply.c:716]: DEBUG: _reply_light: finished 0(2204) DEBUG: <core> [parser/msg_parser.c:623]: SIP Request: 0(2204) DEBUG: <core> [parser/msg_parser.c:625]: method: <REGISTER> 0(2204) DEBUG: <core> [parser/msg_parser.c:627]: uri: <sip:scscf.somecompany.com:6060> 0(2204) DEBUG: <core> [parser/msg_parser.c:629]: version: <SIP/2.0> 0(2204) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field: cseq <CSeq>: <5> <REGISTER> 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=10 0(2204) DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: get_hdr_field: <To> [34]; uri=[sip:bob@somecompany.com] 0(2204) DEBUG: <core> [parser/msg_parser.c:192]: DEBUG: to body ["bob" <sip:bob@somecompany.com> ] 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.929b9c33.1>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=2 0(2204) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: this is the first via 0(2204) DEBUG: <core> [receive.c:149]: After parse_msg... 0(2204) DEBUG: <core> [receive.c:190]: preparing to run routing scripts... 0(2204) DEBUG: maxfwd [mf_funcs.c:85]: value = 15 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK1bd1.a11580a2575452ad784ca00f869952fb.0>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers: this is the second via 0(2204) DEBUG: <core> [parser/parse_via.c:1284]: Found param type 232, <branch> = <z9hG4bK-333530-035cb8a7773aa017e8e89aa8db3c4f01>; state=16 0(2204) DEBUG: <core> [parser/parse_via.c:2672]: end of header reached, state=5 0(2204) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: Via found, flags=1000 0(2204) DEBUG: <core> [parser/msg_parser.c:204]: DEBUG: get_hdr_body : content_length=0 0(2204) DEBUG: <core> [parser/msg_parser.c:106]: found end of header 0(2204) DEBUG: <core> [parser/parse_to.c:176]: DEBUG: add_param: tag=1f9be4d3 0(2204) DEBUG: <core> [parser/parse_to.c:799]: end of header reached, state=29 0(2204) DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 0(2204) DEBUG: siputils [checks.c:103]: no totag 0(2204) DEBUG: <core> [select.c:425]: Calling SELECT 0x7f64e5839ea0 0(2204) DEBUG: tm [t_lookup.c:1095]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_lookup.c:1164]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil 0(2204) ERROR: <script>: Enter register block 0(2204) DEBUG: tm [t_lookup.c:1395]: DEBUG: t_newtran: msg id=5 , global msg id=5 , T on entrance=(nil) 0(2204) DEBUG: tm [t_lookup.c:534]: t_lookup_request: start searching: hash=7601, isACK=0 0(2204) DEBUG: tm [t_lookup.c:492]: DEBUG: RFC3261 transaction matching failed 0(2204) DEBUG: tm [t_lookup.c:716]: DEBUG: t_lookup_request: no transaction found 0(2204) DEBUG: tm [t_hooks.c:374]: DBG: trans=0x7f64de66c4d0, callback type 1, id 0 entered 0(2204) DEBUG: ims_auth [authorize.c:567]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:597]: Nonce or response missing: nonce len [1910316154], response16 len[-440163170] 0(2204) INFO: ims_auth [cxdx_mar.c:81]: created AVP successfully : [maa_return_code] - [-2] 0(2204) DEBUG: ims_auth [authorize.c:298]: Need to challenge for realm [somecompany.com] 0(2204) DEBUG: ims_auth [authorize.c:305]: Checking if REGISTER is authorized for realm [somecompany.com]... 0(2204) DEBUG: ims_auth [authorize.c:435]: Suspending SIP TM transaction 0(2204) DEBUG: ims_auth [authorize.c:1271]: Sending MAR 0(2204) DEBUG: cdp [session.c:276]: adding a session with id scscf.somecompany.com;1782936534;5 0(2204) DEBUG: cdp [routing.c:80]: get_first_connected_route in list 0x7f64de63dbc0 for app_id 16777216 and vendor_id 10415 0(2204) DEBUG: cdp [routing.c:88]: The peer hss.somecompany.com state is opened 0(2204) DEBUG: cdp [routing.c:63]: Checking if peer hss.somecompany.com handles application 16777216 for vendord 10415 0(2204) DEBUG: cdp [routing.c:90]: The peer hss.somecompany.com matches - will forward there 0(2204) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event Send_Message 0(2204) DEBUG: cdp [peerstatemachine.c:1096]: Snd_Message called to peer [hss.somecompany.com] for request with code 303 0(2204) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 0(2204) DEBUG: cdp [session.c:306]: no session found 0(2204) DEBUG: cdp [diameter_msg.c:83]: AAABuildMsgBuffer(): len=348 0(2204) DEBUG: cdp [receiver.c:967]: peer_send_msg(): Pipe push [0x7f64de66edd0] 0(2204) DEBUG: ims_auth [cxdx_mar.c:516]: Successfully sent async diameter 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) 0(2204) DEBUG: <core> [xavp.c:447]: destroying xavp list (nil) 0(2204) DEBUG: <core> [receive.c:293]: receive_msg: cleaning up 10(2214) DEBUG: cdp [receiver.c:743]: select_recv(): There is something on the send pipe 10(2214) DEBUG: cdp [receiver.c:756]: select_recv(): Send pipe says [0x7f64de66edd0] 8 10(2214) DEBUG: cdp [diameter_msg.c:412]: AAAFreeMessage: Freeing message (0x7f64de66edd0) 303 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [1] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:567]: receive_loop(): [hss.somecompany.com] Recv Version 1 Length 172 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [2] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 10(2214) DEBUG: cdp [receiver.c:1061]: receive_message(): [hss.somecompany.com] Recv msg 303 10(2214) DEBUG: cdp [peerstatemachine.c:91]: sm_process(): Peer hss.somecompany.com State I_Open Event I_Rcv_Message 10(2214) DEBUG: cdp [session.c:297]: called get session with id scscf.somecompany.com;1782936534;5 and hash 111 10(2214) DEBUG: cdp [session.c:306]: no session found 10(2214) DEBUG: cdp [receiver.c:104]: --- Receiver [cdp_receiver_peer=hss.somecompany.com] Serviced Peers: --- 10(2214) DEBUG: cdp [receiver.c:110]: Peer: [hss.somecompany.com] TCP Socket: [15] Recv.State: [0] 10(2214) DEBUG: cdp [receiver.c:112]: -------------------------------------------------------- 0(2204) ALERT: <core> [main.c:788]: child process 2212 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on 0(2204) ALERT: <core> [main.c:788]: child process 2211 exited by a signal 11 0(2204) ALERT: <core> [main.c:791]: core was generated 0(2204) INFO: <core> [main.c:800]: INFO: dont_fork turned on, living on This is the modified register block to get a 401 unauthorized to even be generated (probably broken).... route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); if (!ims_www_authenticate(NETWORKNAME)) { ims_www_challenge("$td"); exit; } if ($avp(maa_return_code) != 1 ) { if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } ============================================ On 03/15/2013 03:06 AM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. FreeBSD build from git: i use that patch (Victor V. Kustov) > 2. Re: Problems with Mar portion of registration [IMS] > (Daniel-Constantin Mierla) > 3. Re: UAC module and SRV records (Daniel-Constantin Mierla) > 4. Re: registrar and check registered (Daniel-Constantin Mierla) > 5. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > 6. Re: FreeBSD build from git: i use that patch (Victor V. Kustov) > 7. Re: FreeBSD build from git: i use that patch > (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 15 Mar 2013 11:43:55 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <20130315114355.2409d851@bkstv> > Content-Type: text/plain; charset=US-ASCII > > diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c > index faabd85..24686e7 100644 > --- a/modules/ims_icscf/cxdx_lir.c > +++ b/modules/ims_icscf/cxdx_lir.c > @@ -51,6 +51,11 @@ > #include "mod.h" > #include "location.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable > int create_lia_return_code(int result) { > int rc; > diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c > index bd5216c..8b6b79c 100644 > --- a/modules/ims_icscf/scscf_list.c > +++ b/modules/ims_icscf/scscf_list.c > @@ -47,6 +47,11 @@ > #include "db.h" > #include "../../lib/ims/useful_defs.h" > > +#if defined (__OS_freebsd) > +#include "sys/limits.h" > +#define MAXINT INT_MAX > +#endif > + > extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list > > extern struct tm_binds tmb; //Structure with pointers to tm funcs > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.3-calculate GNU/Linux > > > > ------------------------------ > > Message: 2 > Date: Fri, 15 Mar 2013 08:44:13 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Problems with Mar portion of registration > [IMS] > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D14D.7080808(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > I haven't played with IMS modules, but you said is causing scscf to die > -- is that a kamailio instance? What means 'to die' -- does it crash? If > yes, can you get the syslog messages and the gdb backtrace? > > Cheers, > Daniel > > On 3/13/13 3:01 PM, mrichardson wrote: >> The MAR portion of registration is either not happening or failing >> causing scscf to die. Using Kamailio 4.0 with the ims_auth module. >> We're trying to determine what >> the REGISTER block should look like and what ims_auth parameters are >> necessary for a successful MAR challenge with auth vector delivery. We >> currently only get the UAR by ICSCF then the SAR by SCSCF. >> >> Module parameters: >> # -- ims_auth params -- >> modparam("ims_auth", "name", URI) >> modparam("ims_auth", "registration_default_algorithm", >> REG_AUTH_DEFAULT_ALG) >> modparam("ims_auth","ignore_failed_auth",0) >> #!ifdef CXDX_FORCED_PEER >> modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) >> #!endif >> modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) >> modparam("ims_auth", "av_request_at_once", 1) >> modparam("ims_auth", "registration_qop", "auth") >> >> route[REGISTER] { >> xlog("L_ERR", "Enter register block"); >> t_newtran(); >> >> ims_www_authenticate(NETWORKNAME); >> #check to see if user is authenticated - ie sip header has auth >> information - (already challenged) >> if ($avp(maa_return_code) == 1) { >> # user has not been authenticated. Lets send a challenge >> via 401 Unauthorized >> ims_www_challenge("$td"); >> exit; >> } else { >> # We need to check if this user is registered or not >> if (!impu_registered("location")) { >> save("location"); >> if ($avp(saa_return_code) == 1) { >> isc_match_filter_reg("0","location"); >> exit; >> } >> } else { >> save("location"); >> if($avp(saa_return_code) == 1) { >> isc_match_filter_reg("1","location"); >> exit; >> } >> } >> } >> } >> >> >> 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer >> hss.ims.somecompany.com:3868 connected >> 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: >> ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : >> [saa_return_code] - [-2] >> 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: >> cxdx_get_experimental_result_code: Failed finding avp >> 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP >> successfully : [saa_return_code] - [1] >> 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't >> release transaction >> >> _______________________________________________ >> 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 > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 3 > Date: Fri, 15 Mar 2013 08:48:53 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] UAC module and SRV records > To: "Olle E. Johansson" <oej(a)edvina.net> > Cc: "Kamailio $SER$ - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D265.3060104(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 3/13/13 10:00 AM, Olle E. Johansson wrote: >> 13 mar 2013 kl. 09:35 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> On 3/13/13 9:25 AM, Olle E. Johansson wrote: >>>> 13 mar 2013 kl. 09:08 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>> >>>>> Hello, >>>>> >>>>> do you set the port in r-uri/dst-uri? >>>> No. That would invalidate SRV record lookup, right? >>> yes. >>> >>> Have you watched the dns traffic? Is directly A/AAAA query? Afaik, this function should use just the usual logic to determine the next hop based on dst-uri/r-uri. >> I will debug in more detail. Had to set an outbound URI with a hostname and a port number to get it to work. > There could be the possibility that some of the functions in the chain > of uac_req_send() set port to default 5060 if not set explicitly, but > from sending point of view, it should the same as for the rest of > messages. Thus a look over the code should be done as well. I'll do it > soon if nobody gets the time meanwhile. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 4 > Date: Fri, 15 Mar 2013 08:53:29 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] registrar and check registered > To: "Victor V. Kustov" <coyote(a)bks.tv>, sr-users(a)lists.sip-router.org > Message-ID: <5142D379.20402(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > On 3/13/13 10:00 AM, Victor V. Kustov wrote: >> Hello, Daniel. Thanks for reply. >> >>> do you want to check if caller (origin) is registered or if callee >>> (destination) is registered? Like one can call to pstn only if has a >>> phone registered for itself? >> In our scheme all calls forwards to PSTN GW, if that call from our to >> our users, GW turn call back and kamailio receive it as incoming. >> So, no "local calls" in this scheme. >> >> For outgoing calls (to PSTN) i want check caller (origin) is registered. >> For incoming calls (from PSTN) i want callee is registered. >> >> Need cache registrations (DB, memcached or something). I try that way: >> 1. REGISTER - if ok, than we save("location"). >> 2. Other methods - check !registered("location") or !lookup("location"). >> But in INVITE seems both !registered("location") >> and !lookup("location") return TRUE. I dont understand why. >> >> In topicstart letter i citate part of my cfg: in INVITE i check >> registration again and it work - registered users may call to PSTN, >> unregged may not. But no cache for registration and look as overhead. >> Must be more "beauty" solution. > when you need to check if caller is registered use: > > - registered("location", "$fu") > See more details at: > - http://kamailio.org/docs/modules/stable/modules/registrar.html#id2512620 > > If used without the second parameter is checking if R-URI ($ru) is > registered. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 5 > Date: Fri, 15 Mar 2013 08:55:28 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Message-ID: <5142D3F0.4090408(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > thanks for checking on bsd, can you please sent the patch as attachment? > Because it can saved locally and applied easier. > > Cheers, > Daniel > > On 3/15/13 8:43 AM, Victor V. Kustov wrote: >> diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c >> index faabd85..24686e7 100644 >> --- a/modules/ims_icscf/cxdx_lir.c >> +++ b/modules/ims_icscf/cxdx_lir.c >> @@ -51,6 +51,11 @@ >> #include "mod.h" >> #include "location.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable >> int create_lia_return_code(int result) { >> int rc; >> diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c >> index bd5216c..8b6b79c 100644 >> --- a/modules/ims_icscf/scscf_list.c >> +++ b/modules/ims_icscf/scscf_list.c >> @@ -47,6 +47,11 @@ >> #include "db.h" >> #include "../../lib/ims/useful_defs.h" >> >> +#if defined (__OS_freebsd) >> +#include "sys/limits.h" >> +#define MAXINT INT_MAX >> +#endif >> + >> extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list >> >> extern struct tm_binds tmb; //Structure with pointers to tm funcs >> >> > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > - http://conference.kamailio.com - > > > > > ------------------------------ > > Message: 6 > Date: Fri, 15 Mar 2013 12:01:37 +0400 > From: "Victor V. Kustov" <coyote(a)bks.tv> > Subject: Re: [SR-Users] FreeBSD build from git: i use that patch > To: sr-users(a)lists.sip-router.org > Message-ID: <20130315120137.72b404c4@bkstv> > Content-Type: text/plain; charset="utf-8" > > ? Fri, 15 Mar 2013 08:55:28 +0100 > Daniel-Constantin Mierla <miconda(a)gmail.com> ?????: > >> Hello, >> >> thanks for checking on bsd, can you please sent the patch as >> attachment? Because it can saved locally and applied easier. >> > > > > -- > SY, > Victor > JID: coyote(a)bks.tv > JID: coyote(a)bryansktel.ru > I use FREE operation system: 3.8.2-calculate GNU/Linux >

1 0

FreeBSD build from git: i use that patch
by Victor V. Kustov 15 Mar '13

15 Mar '13

diff --git a/modules/ims_icscf/cxdx_lir.c b/modules/ims_icscf/cxdx_lir.c index faabd85..24686e7 100644 --- a/modules/ims_icscf/cxdx_lir.c +++ b/modules/ims_icscf/cxdx_lir.c @@ -51,6 +51,11 @@ #include "mod.h" #include "location.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + //we use pseudo variables to communicate back to config file this takes the result and converys to a return code, publishes it a pseudo variable int create_lia_return_code(int result) { int rc; diff --git a/modules/ims_icscf/scscf_list.c b/modules/ims_icscf/scscf_list.c index bd5216c..8b6b79c 100644 --- a/modules/ims_icscf/scscf_list.c +++ b/modules/ims_icscf/scscf_list.c @@ -47,6 +47,11 @@ #include "db.h" #include "../../lib/ims/useful_defs.h" +#if defined (__OS_freebsd) +#include "sys/limits.h" +#define MAXINT INT_MAX +#endif + extern int scscf_entry_expiry; //time for scscf entries to remain the scscf_list extern struct tm_binds tmb; //Structure with pointers to tm funcs -- SY, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.8.3-calculate GNU/Linux

3 4

registrar and check registered
by Victor V. Kustov 15 Mar '13

15 Mar '13

Hi. I try to use Kamailio 4.1 git version as registrar and PSTN gw. Need accept register from users. Its OK, I made it and use save() function - from keep registrations in case of kamailio restart. Need forward calls to gw, but only for registered users. If I understand correctly in INVITE enough check registered("location"), isnt it? but it's not work, i need verify auth again. Routing part of my config: route { # initial sanity checks - skip in letter # We are registrar! if (is_method("REGISTER")) { sl_send_reply("100", "Trying"); # send "Trying, dont threat on me if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } # we must check auth right now # WORK: if (!www_authorize("$fd", "subscriber")) if (!auth_check("$fd", "subscriber", "1")) { # WORK: www_challenge("$fd", "0"); auth_challenge("$fd", "0"); # send me requisites plz exit; } else { # ok. auth accepted, save in location if (save("location")) #if (save("location","0x04","$fd")) { xlog("L_WARN", "Save OK in location sip:$au@$fd"); } else { xlog("L_WARN", "Save FAIL in location sip:$au@$fd"); } } xlog("L_WARN", "REGISTERED method done"); exit; } if (is_method("INVITE")) { if ($si == $sel(cfg_get.pstn.gw_ip)) { # incoming calls xlog("L_WARN", "Incoming call"); } else { # outgoing calls xlog("L_WARN", "Outgoing call"); # check auth # we must check auth right now if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } consume_credentials(); ### THAT FUNCTIONS ALWAYS WRITE IN LOGS # "!registered" "!lookup" for both registered and unregged users # if (!registered("location")) xlog("L_WARN","!registered"); # if (!lookup("location")) xlog("L_WARN", "!lookup"); # number change - FIX late # check may be used dialplan automatically? # route $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_WARN", "Rewrite to $ru, send to PSTN."); if (!t_relay()) { sl_reply_error(); } } } else { # reply 2 keepalive and drop all # if (!t_relay()) { sl_reply_error(); } } } -- SY, Victor V. Kustov +7 (952) 966 76 24 JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.7.7-calculate GNU/Linux

2 3

UAC module and SRV records
by Olle E. Johansson 15 Mar '13

15 Mar '13

Hi! When using the UAC module to send a request to a domain that only exists in SRV records, I get an error message that no A or AAAA records can be found for the domain. Is this by design or possibly a bug? Anyone else that have seen this? /O

2 5

Problems with Mar portion of registration
by mrichardson 15 Mar '13

15 Mar '13

The MAR portion of registration is either not happening or failing causing scscf to die. Using Kamailio 4.0 with the ims_auth module. We're trying to determine what the REGISTER block should look like and what ims_auth parameters are necessary for a successful MAR challenge with auth vector delivery. We currently only get the UAR by ICSCF then the SAR by SCSCF. Module parameters: # -- ims_auth params -- modparam("ims_auth", "name", URI) modparam("ims_auth", "registration_default_algorithm", REG_AUTH_DEFAULT_ALG) modparam("ims_auth","ignore_failed_auth",0) #!ifdef CXDX_FORCED_PEER modparam("ims_auth", "cxdx_forced_peer", CXDX_FORCED_PEER) #!endif modparam("ims_auth", "cxdx_dest_realm", NETWORKNAME) modparam("ims_auth", "av_request_at_once", 1) modparam("ims_auth", "registration_qop", "auth") route[REGISTER] { xlog("L_ERR", "Enter register block"); t_newtran(); ims_www_authenticate(NETWORKNAME); #check to see if user is authenticated - ie sip header has auth information - (already challenged) if ($avp(maa_return_code) == 1) { # user has not been authenticated. Lets send a challenge via 401 Unauthorized ims_www_challenge("$td"); exit; } else { # We need to check if this user is registered or not if (!impu_registered("location")) { save("location"); if ($avp(saa_return_code) == 1) { isc_match_filter_reg("0","location"); exit; } } else { save("location"); if($avp(saa_return_code) == 1) { isc_match_filter_reg("1","location"); exit; } } } } 12(22978) INFO: cdp [receiver.c:910]: peer_connect(): Peer hss.ims.somecompany.com:3868 connected 0(22966) ERROR: <script>: Enter register block 0(22966) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [-2] 5(22971) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_experimental_result_code: Failed finding avp 5(22971) INFO: ims_registrar_scscf [cxdx_sar.c:79]: created AVP successfully : [saa_return_code] - [1] 5(22971) WARNING: tm [t_lookup.c:1559]: WARNING: script writer didn't release transaction

2 1

A weird message looping scenario
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, We have an ugly message looping scenario where a SIP request message received at Kamailio is forwarded to the switch/router and back to Kamailio in a loop until the Max-forwards header value becomes 0 or the transaction times out. After receiving an incoming INVITE request from a local user, Kamailio forwards the INVITE request to another proxy located in a different network. The call is established successfully. During the call setup the external proxy sends an OPTIONS request to Kamailio (actually to the external public IP address from where it received INVITE request from Kamailio). The OPTIONS request arrives at Kamailio. But the request uri contains the mapped public address instead of the local ip address where Kamailio is listening. So Kamailio forwards the request to the public IP address (i.e. the switch) which again sends it back to Kamailio again and it keeps looping. Here is the message flow. 192.168.1.3 is the local user registered to Kamailio which is running at 192.168.1.5. IP address of the external proxy server is 10.139.90.161 and 10.139.90.137 is the public IP address of the switch/router. NAT isn't enabled at Kamailio. Internet Protocol Version 4, Src: 192.168.1.3 (192.168.1.3), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@video-conf SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Server: kamailio (3.3.3 (i386/linux)) Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.161 (10.139.90.161) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1002@vc-vcs-control.air2gnd.net SIP/2.0 Message Header Record-Route: <sip:192.168.1.5;lr=on> Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1001@192.168.1.3> Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Subject: Phone call Content-Length: 510 Message Body Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf> Call-ID: 1110783207 CSeq: 20 INVITE Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bKd6e1.f7985e64.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@ video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: <sip:1002@10.139.90.161:5060> Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 192.168.1.3 (192.168.1.3) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 101 Dialog Establishement Message Header Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK1941685709 Record-Route: <sip:192.168.1.5;lr=on> From: "xxxx" <sip:1001@192.168.1.5>;tag=256977615 To: <sip:1002@video-conf>;tag=689763673 Call-ID: 1110783207 CSeq: 20 INVITE Contact: sip:1002@10.139.90.161:5060 Content-Length: 0 P-Received: 10.139.90.137 Internet Protocol Version 4, Src: 10.139.90.161 (10.139.90.161), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.139.90.161:5060;rport;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.161>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "uuuu" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 69 Content-Length: 0 Internet Protocol Version 4, Src: 192.168.1.5 (192.168.1.5), Dst: 10.139.90.137 (10.139.90.137) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 Internet Protocol Version 4, Src: 10.139.90.137 (10.139.90.137), Dst: 192.168.1.5 (192.168.1.5) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: OPTIONS sip:10.139.90.137:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK8ba7.0b94bab7.0 Via: SIP/2.0/UDP 192.168.1.5;received=10.139.90.137;branch=z9hG4bK8ba7.fa94bab7.0 Via: SIP/2.0/UDP 10.139.90.161:5060;rport=5060;branch=z9hG4bK231479102 From: "yyyy" <sip:1002@10.139.90.100>;tag=537974226 To: <sip:1001@192.168.1.5> Call-ID: 1856507851 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 68 Content-Length: 0 This message forwarding goes on for a long time in a loop, each time with an extra Via header and Max-Forwards value decremented by one. After sometime it eventually times out and Kamailio starts sending 408 response for the OPTIONS request which again keeps looping. Probably adding an alias for the public IP address i.e., 10.139.90.137 in the Kamailio config file would solve the problem. But this IP address is not fixed and may change and I guess there is no way Kamailio can learn the public IP address on its own. Any help or reference to previous posts if it is already discussed and solved will be much helpful. Regards, Ajay

2 1

What factor determine NATed or not ?
by Khoa Pham 15 Mar '13

15 Mar '13

Hi, When we send REGISTER or INVITE to Kamailio, which part of the request message does Kamailio use to determine if client is NATed or not ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 2

search_hf returns always true
by Ajay Sabat 15 Mar '13

15 Mar '13

Hi, I want to check whether a parameter exists in a header and I am using the search_hf() function defined in textops module. I am surprised that is always returning true whether the searched pattern exists or not. To be specific, I need to check if the "received" parameter exists in the top most via header and this is how I do it. The statements inside the if block get executed all the time irrespective of whether the received parameter exists in the top most via or not. Any help what am I doing wrong here? if (search_hf("Via", "received=", "f")) { xdbg("received param $sel(via.received) in top most via header \n"); # perform necessary action } Case 1: no received parameter in via Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK39bc.c1076d93.0 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK56870034 And the corresponding debug message from Kamailio is: ./sbin/kamailio[22600]: DEBUG: <script>: received param <null> in top most via header Case 2: received parameter exists Via header in response message looks like: Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK006e.18417697.0;received=10.139.90.137 Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;branch=z9hG4bK960971659 And the corresponding debug message: ./sbin/kamailio[22600]: DEBUG: <script>: received param 10.139.90.137 in top most via header Thanks, Ajay

2 1

testing Supported and Require keywords
by Juha Heinanen 14 Mar '13

14 Mar '13

is there currently a proper way to test if a particular keyword appears in Supported or Require header or is a regex test on $hdr(X)[*] the only possibility? -- juha

3 3

Kamailio World Updates - March 14
by Daniel-Constantin Mierla 14 Mar '13

14 Mar '13

Hello, the groups of speakers at Kamailio World Conference got new members, among them Prof. Dr. Thomas Magedantz from Technische Universitat Berlin/FhG Fokus and Alan Duric, a pioneer in VoIP, among other positions being co-founder of Telio, more details at: - http://conference.kamailio.com/k01/speakers/ It is about one month till the start of the event, agenda is pretty much done, with possibility of few updates in the near future. Regarding Kamailio project, Monday we released v4.0.0, with really good feedback so far. Thank you again to everyone making it possible. New contributions started to show up on master branch, once you are done upgrading to v4.0.0, don't forget to give a try to development version and check the list of new additions. If you haven't registered yet for the Kamailio World conference, you should hurry up, the speakers and the other participants form an excellent group of professionals working with rich communication services and products, it is your chance to meet with them and get updated on the field: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ We are glad to announce the first group of participating companies with all day availability for demos and presentations of their products and services: Sipwise, Sipgate, NG Voice, FhG Fokus and Asipto. The list will be updated soon with more participants. Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

I added the append_hf() inside the if condition in oureply_route[] block as given below, and it is working. I still want to learn how to store data in a variable to use it later in another block. Thanks, Ajay Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio with PostgreSQL as persistence
by Hoai-Anh Ngo-Vi 13 Mar '13

13 Mar '13

Dears, I have tried out Kamailio 3.3.4 with PostgreSQL as persistence but it does not work. make FLAVOUR=kamailio include_modules="db_postgres" cfg was OK. I found /usr/local/lib/kamailio/modules/db_postgres.so after installation. I have successfully set up database openser using /usr/local/sbin/kamdbctl create with DBENGINE=PGSQL in /usr/local/etc/kamailio/kamctlrc. I have edited /usr/local/etc/kamailio/kamailio.cfg with: #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB ... #!ifdef WITH_PGSQL #!define DBURL "postgres://openser/openserrw@localhost:5432/openser" #!endif ... #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif Run /etc/init.d/kamailio start|debug I got Listening on .... kamailio error, failed to start. Does someone know some hints or have a working example? When I comment out the first 3 define kamailio starts fine. Thank you for your help. -- Ngô Vi Hoài Anh Biedenkopfer Weg 13 60489 Frankfurt am Main 069 74 22 36 63 0172 53 97 0 82

2 2

Re: [SR-Users] how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi Daniel, Thanks for your response. But I don't want the IP in the received parameter of the incoming request. Instead, I want to read it (i.e. the public IP address of my own network where the local user and Kamailio are located) from the incoming response received from another proxy located outside the network. And I need that because Kamailio strips the top most via (where the received parameter is present ) in the incoming response before sending it back to the local user, which I guess is correct. But the user needs to know that IP address. So we are planning to send that IP address in a custom header in the response back to the user. After going through some Kamailio documents I found that $sel(via.received) would return the IP address in top most via of a message. Can I store it in a variable inside onreply_route[MANAGE_REPLY] block and use that variable later in some appropriate route block to insert a new header in the response to the local user? Can I do something as given below to fetch and store the IP address from received parameter in the via header of incoming response onreply_route[MANAGE_REPLY] { ... if(search_hf("Via", "received", "f")) { xdbg("received param exists on top most via header \n"); $var(public_ip) = $sel(via.received); } ... } and insert a new header as given below in an appropriate route block which is executed while sending response back? append_hf("My-custom-hdr: $var(public_ip) \r\n"); I am not sure if this logic would work. Is the $var() variable retained through two different sip transactions? And which route block should be used to insert the header as mentioned above? Please help. Thanks, Ajay Hello, the received parameter is constructed from source IP of the incoming request. You get it via $si script variable. Cheers, Daniel On 3/12/13 7:22 PM, Ajay Sabat wrote: > > Hi, > > I want to extract the IP address in the received parameter of the top > most via header of a response and use that in a custom header while > sending response back to the caller. Can somebody please tell how to > implement that in the Kamailio configuration file? > > I need to handle a NAT traversal scenario without enabling NAT in > Kamailio which is used as the proxy server. The caller which is in the > same network as Kamailio first sends a OPTIONS request to the callee > before sending the INVITE. It uses the public IP address returned in > the custom header of the OPTIONS response to build the contact header > and SDP of the INVITE request. In fact, it first looks for the IP > address in the received parameter of the via header to do that. But > Kamailio removes the top most via with received parameter and then > sends the OPTIONS response back to the caller. So we are using a > custom header for that purpose. > > I would appreciate other suggestions to achieve this in Kamailio, may > be by enabling NAT or some other alternative. > > Thanks, > > Ajay > > > > _______________________________________________ > 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Need help understanding/separating signaling from media
by Grant Bagdasarian 13 Mar '13

13 Mar '13

Hello, I want to separate signaling and media in our architecture but I'm not entirely sure if the following will work (note that it is still in a design phase and these kinds of constructs are still new to me): User Agent 1-------------- Kamailio -------------- User Agent 2 | | | | | | -------------- Media Server Cluster -------------- The signaling will go through Kamailio but the media stream should pass through one of the media servers in the cluster for each call. Is it possible to achieve this by having Kamailio instruct the media server which ports to open and alter sdp information? I think that the RTP Proxy module does precisely this, but what I still don't understand is how the rtp stream is passed between these two agents for each call. Also, is RTP proxy the best way to do this or are there any other third party media servers which I could use? I don't need any fancy functionality, just basic media handling/pass-through. Regards, Grant

2 8

Send Bye to all calls in one profile
by Camila Troncoso 13 Mar '13

13 Mar '13

Hi, I have a kamailio proxy server that´s controlling the limit of calls that clients can make with profile grouping. In other side I also limit clients call duration base on users credit ( DATABASE) and *timeout *feature . Before the calls get answer I calculate how much time the client has left for this call and set the timeout variable. When the time is reached the proxy sends BYEs two both sides of this call. I want when the call timeout, as before explain, all other calls in the same profile end too, because the client has expired his credit. Is there a way to achieve this? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

4 6

java code to interact with kamailio
by Rupayan Dutta 13 Mar '13

13 Mar '13

Dear Sir, I want to authenticate users as well as getting buddy list from kamailio using java .Is there any existing library already in place. Rupayan Dutta

3 2

how to extract the ip address in the via header received paramter
by Ajay Sabat 13 Mar '13

13 Mar '13

Hi, I want to extract the IP address in the received parameter of the top most via header of a response and use that in a custom header while sending response back to the caller. Can somebody please tell how to implement that in the Kamailio configuration file? I need to handle a NAT traversal scenario without enabling NAT in Kamailio which is used as the proxy server. The caller which is in the same network as Kamailio first sends a OPTIONS request to the callee before sending the INVITE. It uses the public IP address returned in the custom header of the OPTIONS response to build the contact header and SDP of the INVITE request. In fact, it first looks for the IP address in the received parameter of the via header to do that. But Kamailio removes the top most via with received parameter and then sends the OPTIONS response back to the caller. So we are using a custom header for that purpose. I would appreciate other suggestions to achieve this in Kamailio, may be by enabling NAT or some other alternative. Thanks, Ajay

2 1

Malformed PUBLISH from pua_dialoginfo (4.0.0)
by Charles Chance 13 Mar '13

13 Mar '13

Hi all, We are testing 4.0.0 in the lab and since upgrading, PUBLISHes generated by pua_dialoginfo appear to be malformed (content-length mismatch). Please see attached output from Wireshark for both the PUBLISH and the 400 response from Kamailio. PUBLISH is being sent from one Kamailio to another - nothing has changed other than upgrading from 3.3.3 to 4.0.0. Has anyone else noticed the same? Regards, Charles -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

2 5

Failover on carrierroute
by Ali Jawad 13 Mar '13

13 Mar '13

Hi I did setup carrierroute with failover using t_on_failure, I do have 4 ".25 ratio" primary gateways for one carrier and 4 ".25 ratio"secondary gateways for a secondary carrier. However, given the current setup, if one gateway is down carrieroute does use the secondary carrier if one gateway of the primary carrier is down. What I need is to only go to the secondary carrier if all the gateways of the primary carrier are down. I thought of setting up 4 t_on_failure branches with carrierroute but that will be troublesome to maintain. Is there another method for this ? Regards

2 2

High availability
by Moacir Ferreira 12 Mar '13

12 Mar '13

Hi, What would be the community's architecture advice for deploying Kamailio in a high availability environment? I am looking for something simple, still efficient. Thanks,

3 3

Porting kamailio4 to OpenWrt
by Jiri Slachta 12 Mar '13

12 Mar '13

Hello everyone, I am new to this list and I am a newbie when it comes to Kamailio details, so sorry for any of my misunderstandings. I am the author of kamailio3 package in OpenWrt and currently I am trying to port recent major release of kamailio to OpenWrt. Let's jump directly to my question. The current state of my Kamailio4 package is that all modules which does not depend on any external libraries, are succesfully built. Any module that depends on external library is not built at all. But if I pass LD="$(TARGET_CC)" to the linker via: make -C/path/to/module LD="$(TARGET_CC)" then the module is succesfully compiled. I build kamailio4 via macro Build/Compile in following Makefile - http://liptel.vsb.cz/svn/besip/Trunk/packages-trunk/net/kamailio4/Makefile My question is - has anything changed in build procedures or variables significantly? It seems that LD variable is not passed into Makefiles of specific modules. Am I doing anything wrong? I am open to any suggestions. Thank you! ~ Jiri Slachta

2 1

Fwd: Re: memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

---------- Forwarded Message ---------- Subject: Re: memcached module problems Date: Dienstag, 12. März 2013, 14:50:41 From: Dragos Oancea <droancea(a)yahoo.com> To: Bis: Henning Westerholt <henning.westerholt(a)1und1.de> Thank you Henning, that was a good hint. I've noticed that kamailio and my perl script (which is using Cache::Memcached) set the value in memcached somehow differently. kamailio: set 888_HA1 1 10800 32. 6fa22552ea04d443ab5864a7b3c2168c. perl script: set 888_HA1 0 0 32. 6fa22552ea04d443ab5864a7b3c2168c. VAR_VAL_STR is 1. I will investigate further. Regards, Dragos ps: I will also subscribe to sr-users . From: Henning Westerholt <henning.westerholt(a)1und1.de> To: Dragos Oancea <droancea(a)yahoo.com> Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 12, 2013 9:35 AM Subject: RE: memcached module problems Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt -------------------------------------------------------------

1 0

File kamailio_fifo is removed...
by Bruno Bresciani 12 Mar '13

12 Mar '13

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

2 2

Kamailio MySQL DB Creation
by Amr Elsharqawi 12 Mar '13

12 Mar '13

Hello Kamailio Community!! I am newbie to Kamailio & SIP. And I am interested in Kamailio. I've used asterisk for some time then I heard about FreeSWITCH unfortunately FreeSWITCH was not good enough for me like Asterisk. I've found NAT problem with Asterisk Finally I've decided to work with Kamailio as SIP server & Asterisk as PBX. to be fair I need to learn more about FreeSWICH to test it well. My Question is about Kamailio installation; I am using FreeBSD 9 attached my uname -a FreeBSD PBX.XXXXX 9.1-RELEASE FreeBSD 9.1-RELEASE #0 r243826: Tue Dec 4 06:55:39 UTC 2012 root@obrian.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 I've used gmake for building, compiling and installation. Everything goes smooth EXCEPT mysql db installation when I use sh /usr/local/sbin/kamdbctl create I am getting the following error: /usr/local/lib/kamailio//kamctl/kamdbctl.base: 36: Syntax error: Missing '}' Thanks a lot for your valuable time and that amazing project (Kamailio). Amr

3 6

Strange TLS issue
by Jirka Jirout 12 Mar '13

12 Mar '13

Hello, I have a very strange issue - we are running kamailio 3.3.2 as a part of Sip Provider CE and some of our clients have issues with TLS connectivity. 1. some clients such as Jitsi on Mac OS X and Bria or idoubs on iOS can always connect 2. imsdroid on Android can sometimes connect and works (about 10 % of attempts) When I run tcpudmp on the server during the unsuccessfull attempts, I see something like this: > 1 1 0.0347 (0.0347) C>S SSLv2 compatible client hello > Version 3.1 > cipher suites > TLS_RSA_WITH_AES_128_CBC_SHA > 1 2 0.0360 (0.0013) S>C Handshake > ServerHello > Version 3.1 > session_id[0]= > cipherSuite TLS_RSA_WITH_AES_128_CBC_SHA > compressionMethod NULL > 1 3 0.0361 (0.0001) S>C Handshake > Certificate > 1 4 0.0361 (0.0000) S>C Handshake ServerHelloDone > 1 5 0.0782 (0.0420) C>S Alert > level fatal > value unexpected_message > 1 0.0785 (0.0002) S>C TCP FIN > 1 6 0.1092 (0.0307) C>S Handshake > ClientKeyExchange > 1 7 0.1092 (0.0000) C>S ChangeCipherSpec > 1 8 0.1092 (0.0000) C>S Handshake What I find particularly strange is that the client sends fatal alert and then attempts to continue with handshake. What exactly does this "unexpected_message" mean? Is the client letting the server know the last message was unexpected? Or is this telling me the server had not expected the message? It cannot be a certificate problem, can it? Considering other clients were fine, I blamed our code at first, but I am no longer sure that is the (only) reason for this behavior. I decided to start from scratch and wrote an extremely simple application in plain C, that connects to the TLS port using openssl (which imsdroid also uses), sends some query and then prints response. It works fine on https sites, even on the same server kamailio is running and with the same certificates (our certificates come from Verisign CA G3). When I run ssldump, I can see handshake is fine, socket opened, the client is sending application data. I can see data coming in on the server using tcpdump. However, the data never reaches Kamailio - no sign of the packets in the log and no answer whatsoever (I use a fake REGISTER message so I would expect to get something like "Unauthorized" back). The same happens, when I try to use openssl s_client to connect to the site and send "register" message (because R at the start of packet triggers renegotiation in s_client, uff). My TLS settings in Kamailio are pretty much default, only using my own key and certificate. regards, jjj

4 5

Re: [SR-Users] memcached module problems
by Henning Westerholt 12 Mar '13

12 Mar '13

Hi Dragos, (setting sr-users to CC) Ok, If you’ve set this by other means you need to set the appropriate “Kamailio” type flag to the value as well, that Kamailio can parse it correctly. You can get the appropriate value if you grep for “VAR_VAL_STR” in the source code. Your perl script function call should also export the appropriate flag setting from the memcache library function call, then just add it here. Best regards, Henning From: Dragos Oancea [mailto:droancea@yahoo.com] Sent: Monday, March 11, 2013 8:24 PM To: Henning Westerholt Subject: Re: memcached and other things Hi Henning, Other means. a perl script. Dragos > 0(21130) DEBUG: memcached [mcd_var.c:87]: allocate new memcache request at > 0x7f747fb798f0 > 0(21130) DEBUG: memcached [mcd_var.c:93]: allocate new > memcache result at 0x7f747fb79a70 0(21130) > DEBUG: memcached [mcd_var.c:103]: result: 'just_A_text' for key 888_HA1 with > flag 0 > 0(21130) ERROR: memcached [mcd_var.c:146]: could not convert string > 'just_A_text' to integer value > 0(21130) DEBUG: memcached [mcd_var.c:159]: free memcache request and result > at 0x7f747fb798f0 The issue is that the module somehow does not evaluate the string variable _from memcached_ as a string, it tries to evaluate it as integer. This obviously fails. Has this value set from kamailio or is this from some external application? Then probably this application does not set the proper flags which kamailio expects. Regards, Henning Westerholt

1 0

RPMs for v4.0.0
by Daniel-Constantin Mierla 12 Mar '13

12 Mar '13

Many RPM packages are are now available on OpenSUSE build server (for Centos 5.5 and 6, RedHat 5 and 6, Fedora 16, 17 and 18, OpenSuse 11.4, 12.1, 12.2 and 12.3, on i586 and x86_64 architectures). Links to repositories are at: - http://www.kamailio.org/wiki/packages/rpms There is listed the link to Peter's RPM repository as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Routing calls
by mark＠brightvoip.co.uk 12 Mar '13

12 Mar '13

Hi, I'm looking for pointers on the following scenario. I have two sites (A and B) and sip traffic is sent from A to B. At each site there are a number of SIP (Asterisk) servers and a Kamailio server. The two sites have various network routes (X, Y & Z) between them. I have been asked to load balance the inter-site SIP traffic across the available links with a 20/20/60 traffic split (dispatcher, carrierroute). Each Kamailio server has multiple IP addresses and these are used to direct traffic to specific routes (Kamailio at site A IP 111.111.111.1 to Kamailio site B 111.111.111.2 defines route X for example). For any given call, the complete call (SIP and both RTP streams) should travel wholly over one route. I would welcome any suggestions about how best to route the whole call (sip & rtp) over a specific route, I was working on a solution with multiple RTPProxy instances, but was wondering if this is the best solution. Thanks in advance. Mark

1 0

Kamailio v4.0.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Kamailio v4.0.0 is out – a special major release, shifting up first number of versioning string! It is the landmark for the end of Kamailio-SER integration, bringing out as well a very large set of new features and improvements, like websocket support or dozens of IMS extensions. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.0.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 1

releasing v4.0.0 tomorrow
by Daniel-Constantin Mierla 11 Mar '13

11 Mar '13

Hello, the plan is to release final Kamailio v4.0.0 tomorrow, Monday, March 11, 2013. If any developer/packager has something to add to the release, please commit by 12:00GMT to GIT branch 4.0. After that, announce first on sr-dev and wait for an ack, to avoid packaging inconsistent content. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 1

kamailio is returning 500
by chanea 11 Mar '13

11 Mar '13

Dear all While lots of request come to my Kamailio 3.1.x in the same time , server is returning "500, Server Internal Error" to some of the request. What should i do if i want to reduce the number of "500" response? chanea

3 7

SIP and XMPP server integration
by Vinay V 11 Mar '13

11 Mar '13

Hi All, I am in process of integrating a sip server (Lync) with XMPP server (kamailio). Can a kamailio server be used as a sip to xmpp gateway and vice versa. If i am using any other xmpp server and want to integrate with lync will kamailio can be used as a gateway. This integration is maily for IM and presence. If it can be done what configuration should be done to kamailio.conf file. Please provide the inputs. I am very much need of it. If anybody has done already the above testing please provide DOCS. Thanks in advance, Vinay

2 2

Problem with frequent use of FIFO commands
by Mino Haluz 11 Mar '13

11 Mar '13

Hi, I have some nagios scripts binded to kamctl fifo commands, as well as some web scripts that are directly calling kamctl fifo statements. I do not know why, but it sometimes returns ** ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo [1] => ** ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config [2] => ** ERROR: and also have loaded the mi_fifo module. when it is called with web script with www-data account. From shell it works everytime, but with web 20% of results are errors. Other 80% are sucessful, so no problem with permissions. Does mi_fifo support multiple access, are there any other limits in this regard ? Within OS level? kamailio-3.3.4 Thanks, Mino

2 1

Kamailio 4.0 install guide
by Jesús Pérez Rubio 11 Mar '13

11 Mar '13

Hi, I think there is a little error on this guide about 4.0 version install: http://www.kamailio.org/wiki/install/4.0.x/git First it is said this: "8. Edit configuration file To fit your requirements for the VoIP platform, you have to edit the configuration file. /usr/local/etc/kamailio/kamailio.cfg" At the end it is said that only this line of init.d script have to be changed: DAEMON=/usr/local/sbin/kamailio I would add this one too: CFGFILE=/usr/local/etc/kamailio/kamailio.cfg Another option would be to work directly with "/etc/kamailio/kamailio.cfg" Anyway, I take this oportunity to thank you for your great job and to notice that this version 4.0 seems to work stil better with our QoffeSIP Javascript SIP stack. :) -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com

2 1

no destination sets in dispatcher module
by Ali Jawad 11 Mar '13

11 Mar '13

Hi I am using carrieroute to distribute traffic, I am not sure if there is a way for carrieroute to detect if a destination is down or not, but I read that the dispatcher module does that so I did compile it into the existing kamailio installation and added the following settings : loadmodule "dispatcher.so" modparam("dispatcher","db_url", "mysql://connectionstring") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) and under ROUTE if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; xlog("L_ERR","************************AFTER TOO MANY HOPS******************************************* cr\n"); ds_select_dst("1", "0"); forward(); # t_relay(); All I get is too many hops in ngrep and kamailio reports : (26467) ERROR: dispatcher [dispatch.c:1564]: no destination sets In the DB I do have a setid 1 and sip:ip.of.carrer:5060 as destination. Regards

2 1

why use received as RURI for nathelper OPTIONS ping?
by Andrew Pogrebennyk 11 Mar '13

11 Mar '13

Some Fritzbox devices don't accept the OPTIONS ping generated by kamailio with RURI: sip:11.22.3.4:5060. In the location table we have: received: sip:11.22.3.4:5060 contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967 path: <sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'> Kamailio version is 3.3.2 and the nathelper config is nothing fancy: modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sipwise.local") modparam("nathelper", "received_avp", "$avp(s:received)") What I don't understand is why kamailio sets RURI of the OPTIONS message to value of "received" instead of the contact. I suspect a bug in the parser somewhere along these lines: > rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), > ((unsigned int)(unsigned long)timer_idx)*natping_interval+iteration, > natping_processes*natping_interval); > if (rval != 0) { > pkg_free(buf); > goto done; > } > } > > if (buf == NULL) > goto done; > > cp = buf; > while (1) { > memcpy(&(c.len), cp, sizeof(c.len)); > if (c.len == 0) > break; > c.s = (char*)cp + sizeof(c.len); > cp = (char*)cp + sizeof(c.len) + c.len; > memcpy( &send_sock, cp, sizeof(send_sock)); > cp = (char*)cp + sizeof(send_sock); > memcpy( &flags, cp, sizeof(flags)); > cp = (char*)cp + sizeof(flags); > memcpy( &(path.len), cp, sizeof(path.len)); > path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ; > cp = (char*)cp + sizeof(path.len) + path.len; > > /* determin the destination */ > if ( path.len && (flags&sipping_flag)!=0 ) { > /* send to first URI in path */ > if (get_path_dst_uri( &path, &opt) < 0) { > LM_ERR("failed to get dst_uri for Path\n"); > continue; > } > /* send to the contact/received */ > if (parse_uri(opt.s, opt.len, &curi) < 0) { > LM_ERR("can't parse contact dst_uri\n"); > continue; > } > } else {

4 5

scheduling inside SER
by irma zujovic 11 Mar '13

11 Mar '13

Hello, I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. My question is : where inside SER configuration scripts, the *scheduling *and *queueing *of SIP messages is done? I am interested in a special function or part of the written program. I hope that engineers that worked on developing of SER could help me with this issue. Thank you in advance Irma

3 2

INVITE messages not authenticated (default configuration)?
by Paul Belanger 08 Mar '13

08 Mar '13

Greeting, Hopefully, I'm understanding the following default kamailio.cfg[1] file. Over the weekend, I was attached by SipVicious. Following along with the example Daniel[2] create with kamailio and asterisk, I have almost the same setup. Rather then storing my SIP profiles in Asterisk database, I have then in Kamailio. To my point, the attacker was actually able to by pass any sort of authentication, but simply sending an INIVTE message: ./svmap.py -e 18885551234 kamailio.example.org -m INVITE Which kamailio, forwarded to Asterisk and because there is no additional auth within asterisk, was able to hit the asterisk context for getting processed (they did not get out to the real world). However, my question is.... why do we not authenticate INVITE messages? If my understanding is correct, if would require something like the following: if (is_method("INVITE")) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } } If so, why not also do it in the default configuration file? [1] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=et… [2] http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

6 8

Update existing module or create new?
by Charles Chance 08 Mar '13

08 Mar '13

All, We are working on a memcached module which uses the libmemcached library, as opposed to the (no longer under active development) libmemcache that the existing module depends on. Am I correct in thinking that in this instance, it would be better to create a brand new module (with a different name) rather than modify the existing one? In addition to changing the dependency, we will also be exporting new commands and potentially changing the format of the exported PVs. Regards, Charles -- www.sipcentric.com <http://www.sipcentric.com/> Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

4 8

Asterisk realtime with kamailio Load balancing issue for sip user
by Prakash N 07 Mar '13

07 Mar '13

Hi All, We have finished the Kamailio & Asterisk real time integration and load balancing also done using dispatcher module. Queue and voice mails are load balancing as well.When we are calling extension to extension it is showing in all the servers.It seems extension are not load balancing as per our knowledge. I have attached the kamailio.cfg for your reference,Find my coding below as mentioned. *# -- dispatcher params for DB support --* *modparam("dispatcher","db_url", "mysql:// openser:openserrw@192.168.1.170/openser")* *modparam("dispatcher", "table_name", "dispatcher")* *modparam("dispatcher", "setid_col", "setid")* *modparam("dispatcher", "destination_col", "destination")* *modparam("dispatcher", "flags_col", "flags")* *modparam("dispatcher", "priority_col", "priority")* * * * ----------------------------------------------------------------------------------------- * *# Dispatch requests* *route[DISPATCH] {* *if ( method=="INVITE" ) {* *# dst_select( "GROUP", "HASH METHOD")* * ds_select_dst("1","4");* * sl_send_reply("100","Trying");* * forward();#uri:host, uri:port);* * exit();* *}}* Kindly suggest the solution for the same. Thanks in advance. Regards, N.Prakash

3 10

Re: [SR-Users] scheduling inside SER
by Jiri Kuthan 07 Mar '13

07 Mar '13

not sure if there is a point in doing that, at least for incoming traffic since it already happens in kernel, unless you want to do it in kernel. jiri On 3/7/13 2:25 PM, irma zujovic wrote: > Thank you for your answer. > > So is there any way to produce some kind of algorithm for prioritization of SIP messages inisde FOKUS IMS Core platform by configuring SER? That was my first > goal, when asking the previous question. I tought that scheduling and buffering of messages inside FOKUS IMS is done inside SER, and tought about priorituzation > inside SER. > > If you can help me in any way i would be very greatful. > > Thank you in advance > > Irma > > On Tue, Mar 5, 2013 at 4:13 PM, Jiri Kuthan <jiri(a)iptel.org <mailto:jiri@iptel.org>> wrote: > > On 3/5/13 2:36 PM, irma zujovic wrote: > > Hello, > > I am using FOKUS IMS Core open source platform, and trying to investigate scheduling and queuing inside IMS. FOKUS IMS uses SER. > > My question is : where inside SER configuration scripts, the *scheduling * and *queueing *of SIP messages is done? I am interested in a special function > or part > of the written program. > > > this is not done in SER, it is done in kernel. kernel dispatches incoming traffic to > one of multiple processes listening on sockets. Same for outgoing traffic -- SER leaves > it to kernel's decision. > > jiri > > > I hope that engineers that worked on developing of SER could help me with this issue. > > Thank you in advance > > Irma > > > _________________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

1 0

Wrong ACK packet sent from Cisco gw
by Nawfel Oujdi 07 Mar '13

07 Mar '13

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

2 1

installing v4.0.x from git
by Daniel-Constantin Mierla 07 Mar '13

07 Mar '13

Hello, I created the wiki page with the how-to install kamailio v4.0.x from git. The tutorial is available at: - http://www.kamailio.org/wiki/install/4.0.x/git It should make life easier for those willing to give a try to v4.0.0 these days. Anyhow, a review of the content is very appreciated. Feel free to enhance the content by adding to the wiki page or reply here with your ideas to improve. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

4 7

Fwd: kamailio rls mod
by Aleksandrs Semenenko 06 Mar '13

06 Mar '13

Hello, I'm looking for assistance in RLS module setup for kamailio 3.3.2 The problem is that I need to handle resource-list updates on my kamalio server. For testing purposes I use Jitsi client. Expected: 1) When a new contact is added to Jitsi or an old one is removed, Jitsi sends updated resource-list XML to caps. - This works fine, I can see PUT request in Wireshark and I can see following log from kamailio.cfg: xlog("===== xhttp put: refreshing resource-list for $var(uri)\n"); rls_update_subs("$var(uri)", "presence"); which hopefully means that rls_update_subs function has been called. 2) After this function was called, I expect SUBSCRIBE messages in backend, not in Jitsi, but I still see Jitsi sending SUBSCRIBE for each contact. What am I doing wrong? Any help would be useful! Regards, Alex.

4 3

New developer: Hugh Waite
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, I want to announce that a new person got developer GIT write access to repository: Hugh Waite - he is quite old contributor by now, with many patches and activity on mailing lists. He is part of the Crocodile RCS team, which developed very important features for Kamailio so far, like websocket and outbound extensions, Lua API and presence/rls improvements. His git commit id is: hpw My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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early rpm and deb packages from branch 4.0
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, if anyone is preferring to use packages for pre-release testing of v4.0.0, I created rpms for centos, redhat, opensuse and fedors, plus packages for debian squeeze. You can find them on opensuse build service servers: - https://build.opensuse.org/package/show?package=kamailio40&project=home%3Ak… - https://build.opensuse.org/package/show?package=kamailio40-d6.0&project=hom… I made them mainly to be sure there are no issue compiling on common linux OSes. Debian packages should be available at the usual repository for more versions (including ubuntu), courtesy of Jon Bonilla. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Kamailio Geoip
by Ian French 06 Mar '13

06 Mar '13

Hey List, I'm looking at an article by Daniel, http://by-miconda.blogspot.ie/2010/10/best-of-new-in-kamailio-310-5-geoip-a…. In it he's uses the below to block source IP addresses by country, How could this be used to block multiple country sources?? I've tried a couple of things aleardy but have failed geoip_match("$si", "src"); if($gip(src=>cc)=="CA") { xlog("SIP message from Canada (ip: $si) - drop it\n"); send_reply("403", "Forbidden"); exit; } Thanks in advance Ian

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Kamailio World Updates - March 06
by Daniel-Constantin Mierla 06 Mar '13

06 Mar '13

Hello, the schedule for Kamailio World Conference has been updated, among topics of the recently added talks: North American VoIP market, Jitsi softphone, usage in embedded systems and enterprises. Check the current schedule at: - http://conference.kamailio.com/k01/schedule/ The group of speakers for the two days is filled up at this moment, as soon as we received the profiles and presentation summaries, the schedule will get final. Regarding Kamailio project, these are the last days before the next major release, version 4.0.0 is planned to be out next week. This week is CeBIT in Hanover, many community members will visit the event in various days (to my knowledge, Thursday and Friday will be the busiest). If you want face to face meetings, announce yourself on our forums. I plan to be there tomorrow. It is just a bit over one month till the event starts, if you haven't registered yet, you should hurry up, there is no similar event in Europe this year that gives the chance to meet and learn from many professionals working with rich communication services and products: - http://conference.kamailio.com/k01/registration/ Also, if you consider sponsoring or participate to surrounding exhibition space, please get in touch with us, you will get an exceptional visibility and additional benefits at the event: - http://conference.kamailio.com/k01/sponsors/ Do not forget that there are open positions to work with VoIP/SIP and Kamailio, just in case you look for new challenges in your career, check: - http://conference.kamailio.com/k01/career-jobs/ Looking forward to meet many of you in Berlin, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Simple look table
by Gareth Rylance 05 Mar '13

05 Mar '13

Is it possible to have a custom look-up table to map ids to other ids in the configuration file? E.G. TestLookValues key | value 1 |100 2 | 100 3 | 600 lookup("TestLookValues", "3") gives $var(lookupval) equal to 600 It is similar to MTree functionality only I am looking for an exact match not just a prefix TestLookValues prefix | value 1 |100 2 | 100 3 | 600 mt_match("TestLookValues", "3003)", "0") gives $var(mtval) = 600 I would need $var(mtval) = -1 (not found) Thanks Gareth

2 1

Troubles with call between 2 users
by TF Julian Plata 05 Mar '13

05 Mar '13

Hello, I'm Julian Plata and I have some troubles to do a call between 2 users: julianplata(a)iptel.org since Media5 in my phone, and marinastereo(a)iptel.org in a Comrex box. When I try to connect, Media5 shows a message "The caller can not be found". Also, in the Iptel User Management the users julianplata@iptel & marinastereo@iptel are shown as online. Could you help me to fix the issue ? Could you verify if the accounts are OK for establish a connection?? Thanks, Best Regards, Julian Plata -- ------------------------------------------------------------------------------------ La informacion contenida en este E-mail es confidencial y solo puede ser utilizada por el individuo o la compania a la cual esta dirigido. Si no es el receptor autorizado, cualquier retencion, difusion, distribucion o copia de este mensaje es prohibida y sera sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo al remitente y borrar el mensaje recibido inmediatamente. Cualquier punto de vista expresado en este mensaje es unicamente de quien lo envia excepto donde especificamente se indique que es la posicion de la Armada Nacional de Colombia (ARC). ------------------------------------------------------------------------------------ Apaga la pantalla de su computador cuando salgas a almorzar. No malgastemos la Energía. Cuidemos lo que es de Todos. ------------------------------------------------------------------------------------ Antes de imprimir este e-mail piensa bien si es necesario hacerlo: 1 hoja de papel tamaño carta = 2 litros de agua + 10 gramos de materia prima (decolorantes, fijadores, pigmentos) + 1 bombillo de 40 Vatios prendido por 1 hora + 14 gramos de madera pura.

2 1

Kamailio problem with both clients in the same NAT
by mohavere36 04 Mar '13

04 Mar '13

4 4

releasing v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, from my point of view, the last to sort out for v4.0.0 is packaging (I just sent a dedicated email to decide on that). Personally I have been testing v4.0.0 thoroughly in various cases, no critical issue being on my list at this moment. Also, v4.0 is running for quite some time on voipuser.org (our usual life testing platform). I propose releasing v4.0.0 next Monday, on March 11, 2013. The second option, if more people prefer it, would be Thursday, March 14, 2013. Opinions? If anybody has other dates in mine, let us know here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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packaging v4.0.0
by Daniel-Constantin Mierla 04 Mar '13

04 Mar '13

Hello, (cross-posting as this decision may be done properly with feedback from both developers and users community). Now that we have some band new modules but also several that were not packaged by Kamailio, just by ser (due to single modules directory), I was thinking we may want to reconsider a bit the packages we do for v4.0.0. 1) A new package is related to the bunch of IMS modules, that is pretty much obvious. 2) There are few modules targeting development samples or troubleshooting: print, print_lib and malloc_test - I think they should not be packaged. 3) I would create a special package for the modules using uid-based database schema: db2_ops uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db uri_db 4) Maybe we should merge some packages by dependencies: - a) some depend on libxml2: cpl-c xhttp_pi xmlrpc xmlops - b) some depend on libssl: tls auth_identity outbound 5) Some experimental modules with external deps will stay unpackaged like so far: db_oracle Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

When to rewrite o= in SDP ?
by Khoa Pham 04 Mar '13

04 Mar '13

Hi, I read in http://kamailio.org/docs/modules/1.4.x/nathelper.html that Kamailio might rewrite ip address in o= in SDP to facilitate NAT traversal. It may decide to use to original IP or use IP of RTP server. What factors affect its decision ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

2 3

UnixODBC cant insert sip_capture record on sql server
by Grant Bagdasarian 03 Mar '13

03 Mar '13

Hello, We are using SQL Server 2008 as our database. When the sip_capture module tries to insert a row, the following error is given: db_unixodbc [con.c:220]: unixodbc:SQLExecDirect=42000:1:156:[FreeTDS][SQL Server]Incorrect syntax near the keyword 'authorization'. Using Kamailio 3.3.4. Is there a way to force brackets around the column values? Regards, Grant

2 3

uac_replace_from / New URI shorter than old URI
by Ronald Voermans 03 Mar '13

03 Mar '13

Hi, this last weekend we went live from the old 3.0 version to the latest stable release of Kamailio (3,3,0). All went well, except for one issue. We make use of uac_replace_from in our config, in case a client needs CLIR: uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); Since the upgrade we receive the following error: uac [replace.c:521]: new URI shorter than old URI Which causes the call to be disconnected after about 20/30 seconds. This, because the ACK is received by Kamailio, but cannot be 'linked' to the transaction i guess. What is exactly causing this issue and, more importantly, is there any way to resolve this? Thanks in advance, Regards, Ronald Voermans

2 4

Which is first, t_on_reply or callbacks in C ?
by David 03 Mar '13

03 Mar '13

Hello, Can someone tell me which is executed first ? the on reply route setup with t_on_reply() or the callbacks in the code that allows the module dialog to insert the confirmed dialog into the dialog table ? Thanks, David

2 1

not able to install Kamailio
by Julián Quintero 03 Mar '13

03 Mar '13

Hello thereI hope that you can help me, I have an issue installing Kamailio, when I issue th command "make all" I receive the following error resolve.h:471:4: error: #error neither gethostbyname2 or getipnodebyname present make: *** [action.o] Error 1 I am using debian squeeze, please let me know if there is something missing in my system or what is not allowing to install the SIP Server THanks Julián

3 3

t_lookup.c: WARNING: script writer didn't release transaction - problem
by Efelin Novak 03 Mar '13

03 Mar '13

Hi folks, we have a serious issue with tm module. In Fail route we would like to send a reply 404 when a cr_route function fails(no destination found). We use t_reply("404", "No route to destination"). However after the call is disconnected with a 404 reply the tm module writes a following warning WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Very soon after this error message is shown a segfault occurs. Log is as follows: Mar 3 15:26:46 hostname kamailio[791]: ERROR: carrierroute [cr_func.c:594]: desired routing domain doesn't exist, prefix 0039333333333, carrier 14, domain 2 Mar 3 15:26:46 hostname kamailio[791]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction //ACK is received Mar 3 15:26:52 hostname kernel: [771134.093250] kamailio[796]: segfault at 4 ip 0815c182 sp bf8ace00 error 6 in kamailio[8048000+1ea000] Mar 3 15:26:53 hostname kamailio[803]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 15 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:785]: child process 796 exited by a signal 11 Mar 3 15:26:53 hostname kamailio[790]: ALERT: <core> [main.c:788]: core was generated Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Mar 3 15:26:53 hostname kamailio[802]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[799]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[801]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[798]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[800]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[797]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[792]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[803]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[795]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[791]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[794]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[793]: INFO: <core> [main.c:851]: INFO: signal 15 received Mar 3 15:26:53 hostname kamailio[790]: INFO: <core> [mem/f_malloc.c:532]: freeing a free fragment (0x886b3ad8/0x886b3ae0) - ignore Mar 3 15:26:53 hostname kernel: [771135.681720] kamailio[790]: segfault at 7dd193e0 ip 0816f5cd sp bf8acc20 error 4 in kamailio[8048000+1ea000] Currently we have fixed it using a custom route route(T_REPLY); . . . route[T_REPLY] { sl_send_reply("404","No route to destination"); } This works correctly with no error or a warning message or a segfault. We are using kamilio version: kamailio 3.3.4 (i386/linux) f8c8f2 Is this behavior a bug or should I dig deeper into the tm documentation? Kind regards Efelin

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Karl Smith
by Karl Smith 02 Mar '13

02 Mar '13

witwcvrf ictniydtoz.vsd/http://www.loudounwalkin.com/sq/f7nax0z0n2ouq77adgkgejjvnz.g… yklxx.vsd/knc hj.vsd/

1 0

RPM packages for v3.3.4
by Daniel-Constantin Mierla 01 Mar '13

01 Mar '13

Hello, the RPM packages for latest minor release v3.3.4 were created on openSuse build service. More details and download links at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Karl Smith
by Karl Smith 01 Mar '13

01 Mar '13

http://www.louiselemelin.com/sinkzm/hhir5ew9drwu4s54d429jdnj.d?a6dfinu09fqc… Smith3/1/2013 3:00:01 PM

1 0

How to let Kamailio to always use RTP proxy
by Khoa Pham 01 Mar '13

01 Mar '13

Hi, Supposed A call B in a non-NAT environment. How to let Kamailio to always force A, B to use RTP proxy ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

4 4

Help with Asterisk RT integration
by Jon Morby 28 Feb '13

28 Feb '13

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

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Re: [SR-Users] SBC
by Keith 28 Feb '13

28 Feb '13

Hi All, So following a guide I have successfully setup Kamailio and FreeSwitch together working as an SBC. SIP registration with Kamailio and using FreeSwitch as the media server which in turn authenticates with users in Kamailio. I have managed to pass calls between users etc. So the next challenge is bolting 2 Asterisks servers behind this SBC and load balancing the media. As the SBC will be public facing I also have to consider NAT which I believe will be configured in FreeSwitch. I have had some fantastic advice in the past so just interested to get peoples views again. Thanks. -- Keith

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Re: [SR-Users] [sr-dev] openser db on an external mysql server
by Prakash N 28 Feb '13

28 Feb '13

Hi Govoiper Free Memory on my server 2309 MB total used free shared buffers cached Mem: 2795 486 2309 0 70 180 -/+ buffers/cache: 234 2561 Swap: 2927 0 2927 *Kamailio error:* * * ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed *tail -f /var/log/messages:* Feb 12 23:41:28 localhost /usr/local/sbin/kamailio[4280]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: usrloc [ul_mod.c:407]: child(1): failed to connect to database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4281]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module usrloc (/usr/local/lib64/kamailio/modules_k/usrloc.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to MySQL server on '192.168.1.170' (4) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [db.c:320]: could not add connection to the pool Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: auth_db [authdb_mod.c:174]: unable to connect to the database Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [sr_module.c:896]: init_mod_child(): Error while initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: ERROR: <core> [pt.c:350]: ERROR: fork_process(): init_child failed for process 3, pid 4283, "udp receiver child=2 sock=127.0.0.1:5060" Feb 12 23:41:30 localhost /usr/local/sbin/kamailio[4283]: : <core> [main.c:1625]: main_loop: Cannot fork With Regards N.Prakash ---------- Forwarded message ---------- > From: SamyGo <govoiper(a)gmail.com> > Date: Tue, Feb 12, 2013 at 5:49 PM > Subject: Re: [SR-Users] [sr-dev] openser db on an external mysql server > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: Muhammad Shahzad <shaheryarkh(a)gmail.com>, sr-dev(a)lists.sip-router.org, > Prabhakaran B <prabhakaran(a)tevatel.com>, Indu Priya <indu(a)tevatel.com> > > > Hi Prakash, > > Please paste All ERROR appearing and Warnings when you start your > kamailio. What is the free memory situation on your server ? > > Regards, > Sammy > > > > > On Tue, Feb 12, 2013 at 5:12 PM, Prakash N <prakash.n(a)tevatel.com> wrote: > >> Hi Muhammad, >> >> Yes, I have define Mysql on top in Kamailio.cfg >> >> DBURL "mysql://openser:openserrw@192.168.1.170/openser" >> >> >> I have attached Kamailio.cfg ,please advice >> >> With Regards >> >> N.Prakash >> >> >> On Tue, Feb 12, 2013 at 5:37 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com>wrote: >> >>> What is your connection string in kamailio i.e. DBURL? Also make sure >>> you have defined WITH_MYSQL on the top of kamailio.cfg to load necessary >>> drivers. >>> >>> #!define WITH_MYSQL >>> >>> Thank you. >>> >>> >>> On Tue, Feb 12, 2013 at 12:56 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>> >>>> Hi Shahzad , >>>> >>>> It is not ,We can able to connect remote mysql ( please find >>>> the detail below),moreover we can able create openser db from Kamailio >>>> server to remote mysql server >>>> >>>> *Mysql server : 192.168.1.170* >>>> *Kamailio server : 192.168.1.101* >>>> >>>> >>>> *Mysql output :* >>>> * >>>> * >>>> *mysql -u openser -h 192.168.1.170 -p* >>>> Enter password: >>>> Welcome to the MySQL monitor. Commands end with ; or \g. >>>> Your MySQL connection id is 68 >>>> Server version: 5.1.67 Source distribution >>>> >>>> Copyright (c) 2000, 2012, Oracle and/or its affiliates. All rights >>>> reserved. >>>> >>>> Oracle is a registered trademark of Oracle Corporation and/or its >>>> affiliates. Other names may be trademarks of their respective >>>> owners. >>>> >>>> Type 'help;' or '\h' for help. Type '\c' to clear the current input >>>> statement. >>>> >>>> mysql> >>>> >>>> With Regards >>>> >>>> N.Prakash >>>> >>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Muhammad Shahzad <shaheryarkh(a)gmail.com> >>>>> Date: Tue, Feb 12, 2013 at 5:16 PM >>>>> Subject: Re: [sr-dev] [SR-Users] openser db on an external mysql server >>>>> To: Development mailing list of the sip-router project < >>>>> sr-dev(a)lists.sip-router.org> >>>>> Cc: Daniel-Constantin Mierla <miconda(a)gmail.com>, >>>>> sr-users(a)lists.sip-router.org, Indu Priya <indu(a)tevatel.com>, >>>>> Prabhakaran B <prabhakaran(a)tevatel.com> >>>>> >>>>> >>>>> Its not a Kamailio error, it is problem with connecting to mysql >>>>> server. Error 4 means No MySQL Server Running On Specified Host. >>>>> >>>>> May be MySQL server is down or there is some firewall running or MySQL >>>>> not listening on IP you specified. >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> On Tue, Feb 12, 2013 at 12:11 PM, Prakash N <prakash.n(a)tevatel.com>wrote: >>>>> >>>>>> Dear Daniel, >>>>>> >>>>>> Thanks for your quick response >>>>>> >>>>>> We can able to create operseb db on remote mysql server ,but >>>>>> Kamailio not getting start >>>>>> >>>>>> Error detail: >>>>>> >>>>>> ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to >>>>>> MySQL server on '192.168.1.170' (4) >>>>>> ERROR: <core> [db.c:32 0]: could not add connection to the pool >>>>>> ERROR: auth_db [authdb _mod.c:174]: unable to connect to the database >>>>>> ERROR: <core> [sr_modu le.c:896]: init_mod_child(): Error while >>>>>> initializing module auth_db (/usr/local/lib64/kamailio/modules_k/auth_db.so) >>>>>> >>>>>> >>>>>> I have attached my kamailo.cfg ( we followed Kamailio and >>>>>> Asterisk real time installation) ,please advice >>>>>> >>>>>> With Regards >>>>>> >>>>>> N.Prakash >>>>>> >>>>>>> >>>>>>> ---------- Forwarded message ---------- >>>>>>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>>>>>> Date: Tue, Feb 12, 2013 at 2:41 PM >>>>>>> Subject: Re: [SR-Users] openser db on an external mysql server >>>>>>> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >>>>>>> Users Mailing List" <sr-users(a)lists.sip-router.org>, >>>>>>> sr-dev(a)lists.sip-router.org, miconda(a)gmail.com >>>>>>> Cc: Prabhakaran B <prabhakaran(a)tevatel.com> >>>>>>> >>>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> kamailio database is created with: >>>>>>> >>>>>>> kamdbctl create >>>>>>> >>>>>>> the host of mysql server has to be specified in kamctlrc file via >>>>>>> DBHOST variable - kamctlrc is located in same directory as kamailio.cfg. >>>>>>> >>>>>>> In kamailio.cfg, update the db_url value (or DBURL define) to >>>>>>> reflect the mysql server host. >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> >>>>>>> On 2/12/13 4:16 AM, Prakash N wrote: >>>>>>> >>>>>>> Hi All, >>>>>>> >>>>>>> I am new this Kamailo ,ther is any step by step document to set >>>>>>> up openser db on an external mysql server ? >>>>>>> >>>>>>> With Regards >>>>>>> >>>>>>> N.Prakash >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> Kamailio World Conference, April 16-17, 2013, Berlin >>>>>>> - http://conference.kamailio.com - >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sr-dev mailing list >>>>>> sr-dev(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Muhammad Shahzad >>>>> ----------------------------------- >>>>> CISCO Rich Media Communication Specialist (CRMCS) >>>>> CISCO Certified Network Associate (CCNA) >>>>> Cell: +49 176 99 83 10 85 >>>>> MSN: shari_786pk(a)hotmail.com >>>>> Email: shaheryarkh(a)googlemail.com >>>>> >>>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >> >> >> _______________________________________________ >> 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 >> >> > >

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GIT master branch is open for development of v4.1.x
by Daniel-Constantin Mierla 27 Feb '13

27 Feb '13

Hello, the GIT master branch is again open for new development. Common testing period of releasing v4.0.0 has ended, if you have pending commits or personal branches to merge, you can go ahead. v4.0.0 will be released out of newly created branch 4.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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