- sr-users - kamailio.org

TLS & log files
by Tributh 13 Apr '15

13 Apr '15

Hi, is there any way to configure the tls curve like secp384r1 in the config? I use actually version 4.2.4 On the other side i can't see any connection Information in the logs. I would like to see something like: "Connected with TLSv1.1 using cipher ECDHE-RSA-AES256-GCM-SHA384 and curve secp384r1" With this logs the variety of the offered ciphers can be shortened to the most secure and only needed by the clients. Regard T.Tributh

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Kamailio LDAP authentication
by Marek Moravcik 11 Apr '15

11 Apr '15

Hello, I'd like to authenticate Kamailio users in LDAP. But it looks like Kamailio need to download password from LDAP and authenticate user on it's own. Is there any possibility to send password to LDAP and let LDAP to say, if the user can be sign in? Thanks Marek Moravčík

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In acc for 5xx calls result in "Unknown column 'time_attr' in 'field list'"
by Ethan Brooks 10 Apr '15

10 Apr '15

I am seeing a very strange error saying that time_attr is an unknown column and the insert is failing. What is strange is that it is in the acc table and is being populated for all successful calls. It seem to only happen for 5xx unsuccessful calls and the INVITE is not recorded. 4xx response are recorded. *config* modparam("acc", "time_mode", 1); modparam("acc", "failed_transaction_flag", 3) *syslog* Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'time_attr' in 'field list' Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 9 20:40:52 a-deploy-callprocessing3 /usr/sbin/kamailio[9732]: ERROR: acc [acc.c:504]: acc_db_request(): failed to insert into database *Database Fields:* `time_attr` int(11) DEFAULT NULL, `time_exten` int(11) DEFAULT NULL, *kamailio -v* version: kamailio 4.2.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_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 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 12:08:27 Feb 16 2015 with gcc 4.9.1

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weight of rtpengine rtp proxy?
by Juha Heinanen 10 Apr '15

10 Apr '15

rtpengine readme tells in section 2: The balancing inside a set is done automatically by the module based on the weight of each RTP proxy from the set. i have not found in rtpengine_sock module parameter description, how is the weight of a rtp proxy is specified specified. it contains an example: modparam("rtpengine", "rtpengine_sock", "1 == udp:localhost:12221 udp:localhost:12222") how can i tell what the weight of each of two proxies is? -- juha

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Re: [SR-Users] fifo error and kamailio wont start
by Bruno Emer 10 Apr '15

10 Apr '15

Ok Joli! Here is the way I do this. At the begining of your kamailio.cfg, just after "#!KAMAILIO" insert this define: #!define WITH_DEBUG In your global parameters, you must now define what happens if "WITH_DEBUG" is configurated, so add these lines: #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif On Modules Selection, load the module debbuger.so: #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif And set the module specific parameters: #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif Maybe some of these informations can be there, but I am not sure about the default configuration file. After these changes, restart your kamailio and you should see a lot of information in your screen. Then, just try to find the error there. I just asked something here in this forum as well, and posted my .cfg file there. If you want to check the places where I put these lines, take a look at the topic "MULTIDOMAIN Environment". After discover the error and finish your troubleshooting, you can just comment the line #!define WITH_DEBUG putting one more # at the beginning of it and you will not see the debug information anymore. *Bruno Emer*

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Re: [SR-Users] fifo error and kamailio wont start
by Bruno Emer 10 Apr '15

10 Apr '15

Hello Joli! To see what is going on, is a good idea to set debug=4 so you can see exactly what is going on. If you don't know how to do this, please let me know and I can try to help you. Bruno Emer

2 1

fifo error and kamailio wont start
by Joli Martinez 10 Apr '15

10 Apr '15

Hello, I am installing the latest version of Kamailio on CentOS 7 from Git. Everything compiled correctly, but when I try to start it I am getting a fifo error and Kamailio wont start [error] kamailio: : <core> [cfg.y:3428]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 296, column 39: Can't initialize shared memory kamailio: ERROR: mi_fifo [mi_fifo.c:271]: mi_destroy(): FIFO stat failed: Bad address kamailio: Starting kamailio: kamailio.service: control process exited, code=exited status=1 systemd: Failed to start LSB: Kamailio (OpenSER) - the Open Source SIP Server. systemd: Unit kamailio.service entered failed state. [/error] any suggestions on how to fix this?

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Keeping track of routing/dialog attribute value pairs
by Dirk Teurlings - SIGNET B.V. 10 Apr '15

10 Apr '15

Hello, Throughout the routes of kamailio we use multiple variables. These consist of $var, $avp, $dlg_var and probably more. What's the policy for these variables when used. When, if even, are they being cleaned up after a call has been terminated. Is this something we need to be doing ourselves with functions like avp_delete("foo")? Many thanks! Dirk

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MULTIDOMAIN Environment
by Bruno Emer 10 Apr '15

10 Apr '15

Hello all! I have a scenario here where I need to put more than two companies into the same Kamailio server and to accomplish this I'm using a multi-domain configuration. For now, I am able to create users in different domains and register all of them. The users are able to place and receive internal calls (as I am using this just to get internal communications working), so I am ok with this part of my configuration. The problem that I'm facing now is that users from domain "A" are able to place calls to users from domain "B". I need to deny calls between different domains, and I don't know the best way to achieve this. Here are my domains create (output from "kamctl domain show"): domain:: enterprise.com did:: enterprise.com domain:: enterprise2.com did:: enterprise2.com And here are my users created: username: bob domain: enterprise.com username: alice domain: enterprise2.com So, can you please help me to get this? Here is my .cfg file: #!KAMAILIO #!define WITH_PGSQL #!define WITH_AUTH #!define WITH_USRLOCDB ##!define WITH_NAT ##!define WITH_DEBUG #!define WITH_MULTIDOMAIN ##!define WITH_SIPTRACE #!define WITH_SQLOPS #!define WITH_XMLRPC ##!define WITH_603 #!define WITH_TLS # #!substdef "!MY_WS_PORT!80!g" #!substdef "!MY_WSS_PORT!81!g" #!substdef "!MY_WS_ADDR!eth1:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:eth1:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS #!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_PGSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "postgres://kamailio:kamailiorw@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 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=no tcp_accept_no_cl=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=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 tcp_rd_buf_size=6144 listen=eth1 #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=eth1 #!endif #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.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 #!ifdef WITH_SIPTRACE loadmodule "siptrace.so" #!endif #!ifdef WITH_SQLOPS loadmodule "sqlops.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!endif # ----------------- setting module-specific parameters --------------- # ---- sip_trace params ---- #!ifdef WITH_SIPTRACE modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost /kamailio") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_sl_acks", 0) #!endif # ----- 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) # 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", 300) modparam("registrar", "min_expires", 30) # 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) #modparam("auth_db", "use_domain", 1) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "connection_timeout", 60) modparam("tls", "tls_debug", 10) modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #modparam("tls", "verify_certificate", 1) #modparam("tls", "require_certificate", 0) #!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 #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(REQINIT); #!ifdef WITH_SIPTRACE if($hdr(X-Omnz-SipTrace) !=$null) { sip_trace(); setflag(22); } #!endif #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) fix_nated_register(); else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # 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. #!ifdef WITH_603 if (is_method("INVITE")) { if($hdr(X-Omz-int-Id) ==$null) { # request with no Username in RURI sl_send_reply("603","Declined"); exit; } } #!endif 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()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif 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("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { 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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_WEBSOCKETS onreply_route { if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { xlog("L_DBG", "WebSocket\n"); xlog("L_DBG", " Host: $hdr(Host)\n"); xlog("L_DBG", " Origin: $hdr(Origin)\n"); if ($hdr(Host) == $null ) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information abou the # successful connection exit; } } xhttp_reply("404", "Not found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif

2 1

The caller doesn't receive reject message when the callee has more one register location
by xuefeng zhang 10 Apr '15

10 Apr '15

Dear Sir/Madame, I find out a caller doesn't receive reject message when a callee has been registered more times with different device then keep only one callee's device is available.I use the linphone's app to test it,It always reproduce on 4.2.4 stable version.The linphone's app reject a new call that will send 603 message,but the caller can't receive the 603 message.The caller will be continue at calling state. So I add a source code "*should_relay=branch" at 1384 line number in t_reply.c file. Hope you can change it. Thanks! Xuefeng Zhang

2 3

apache 2.4 wstunnel
by Juha Heinanen 10 Apr '15

10 Apr '15

apache 2.4 has capability to tunnel websocket requests: https://httpd.apache.org/docs/2.4/mod/mod_proxy_wstunnel.html the tunnel also automatically upgrades the connection to websocket connection meaning that ws_handle_handshake() is not needed. has anyone managed to configure kamailio to handle sip requests via apache wstunnel? -- juha

4 9

Cannot register Aastra 6757i
by David Matichuk 10 Apr '15

10 Apr '15

I set up a new account and I am unable to register my phone. It shows "no Service" I'm using an Aastra 6757i. David

1 0

DTLS not working with kamailio
by renuka ghate 09 Apr '15

09 Apr '15

I have configured the kamailio server for TLS.. but my soft phone supports only DTLS, how do i configure kamailio to support DTLS. regards, Renuka

4 3

Re: [SR-Users] Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 09 Apr '15

09 Apr '15

I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users(a)lists.sip-router.org Subject: Avoiding "script writer didn't release transaction" warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

1 0

Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 09 Apr '15

09 Apr '15

Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

1 0

Work with radius
by Никитенко Виталий 09 Apr '15

09 Apr '15

Hi. I have scheme: User -> Kamailio -> Asterisk -> PSTN | V Radius Server At the beginning dialog (INVITE) kamailio send to radius server START packet which indicates Called, Calling, Accounting Session ID, connect_time, setup_time. When the terminating call (BYE) kamailio forms STOP package. How can I get on kamailio Called, Calling, connect_time that I have in kamailio for START packet if I only match Accounting Session ID? Thank.

3 2

missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 08 Apr '15

08 Apr '15

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 <http://10.0.1.30:5061/> kamailio server 2(k2): 10.0.1.32:5061 <http://10.0.1.32:5061/> sip client c1: sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30 <mailto:sip%3A72316@10.0.1.30>, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 <http://10.0.1.30:55999/> failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999 <http://10.0.1.30:55999/>: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); 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); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } 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); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Thanks

3 5

rtpstat counters
by Andres 08 Apr '15

08 Apr '15

In the documentation at: http://kamailio.org/docs/modules/4.2.x/modules/rtpproxy.html I see a variable called $rtpstat which "Returns the RTP-Statistics from the RTP-Proxy. The RTP-Statistics from the RTP-Proxy are provided as a string and it does contain several packet-counters." But I cannot find out where exactly it describes the packet counters. For example the xlog entry: log("L_ALERT", "STATS-rtpproxy,$rtpstat\n"); Will yield an output like: STATS-rtpproxy,60 596 660 1256 0#012 I need to find the documentation for those counters. Any ideas? Thanks, -- Technical Support http://www.cellroute.net

1 0

Variable interpolation in xavp identifiers
by Alex Balashov 08 Apr '15

08 Apr '15

Hello, There seems to be an implementational difference in interpolation of PVs in AVP and XAVP identifiers and/or field names. The pseudovariables documentation[1] on classical AVPs says: The 'id' can be: ... pseudo variable - if value of pv is integer, id is integer, if string, id is string So, for example, this composition of variables works: $var(f) = 'donkey'; $avp($var(f) = 'eehaw'; xlog("L_INFO", "Donkey says: $avp(donkey)\n"); It correctly prints: Donkey says: eehaw Making some assumptions, I tried this, and it did not work similarly as expected: $var(f) = 'donkey'; $xavp(r=>$var(f)) = 'eehaw'; xlog("L_INFO", "Donkey says: $xavp(r=>donkey)\n"); This prints: Donkey says: <null> And a PV dump confirms that the PV value is not interpolated: Mar 31 18:26:15 localhost /usr/local/sbin/kamailio[2399]: INFO: debugger [debugger_api.c:1799]: dbg_dump_json(): {"$xavp(r)":[{"$var(f)":["eehaw"]}]} Interpolation of variables does not work for either field or stem names. Is this a design principle of XAVPs, or an oversight? Thanks, -- Alex [1] http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#avp_id_-_avps -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

BYE not forwarded
by Grant Bagdasarian 08 Apr '15

08 Apr '15

Hello, I have the following setup: UAC (Bria) ------ Kamailio -------- Freeswitch Kamailio is used to authenticate the client using basic authentication. Upon authentication the INVITE is sent to Freeswitch for further handling. Kamailio stays in the path using record-route. At this point, the authentication and calling FS works fine. For now, FS just answers, plays a file and hangsup the call. Answering the call and playing a file works fine. When FS hangs up the call, the BYE message is sent back to Kamailio. The problem is the BYE message from FS contains a Route header which points to Kamailio again. Which causes Kamailio to send the BYE message back to itself. Kamailio then replies with a 404. The BYE is never sent to the UAC. When I remove the record_route() for INVITEs, the call works fine, but Kamailio isn't kept in the signaling path. The BYE is sent directly to the UAC. T 10.0.5.52:54822 -> 10.0.5.50:5060 [AP] BYE sip:grant@10.0.0.40:5060;transport=tcp SIP/2.0. Via: SIP/2.0/TCP 10.0.5.52;rport;branch=z9hG4bK99X8a4Q8BFy1a. Route: <sip:10.0.5.50;transport=tcp;lr=on>. Max-Forwards: 70. From: <sip:123456789@10.0.0.40>;tag=aey7rKe7DmQte. To: <sip:grant@10.0.0.40>;tag=dbc9c27b. Call-ID: NmQ4YjYyMDFkNmMwZjg0ZmU5NjQ3ODhjYzU0MDBmNGI. CSeq: 73909300 BYE. User-Agent: CM SBC. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY. Supported: timer, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Here is the WITHINDLG route: route[WITHINDLG] { xlog("L_INFO", "[R-ROUTE-WITHINDLG:$ci] !>" "Route:WITHINDLG;UnixTime:$TV(un);RequestURI:$rU;RequestMethod:$rm;FromTag:$ft;ToTag:$tt;CallID:$ci;From:$fU;To:$rU\r\n"); if (!has_totag()) { return; } # Sequential request within a dialog should take the path determined by record-routing if (loose_route()) { route(RELAY); exit; } else { if (is_method("ACK")) { if (t_check_trans()) { # No Loose-Route, but stateful ACK. Must be ACK after 487 route(RELAY); exit; } else { # ACK without matching transaction exit; } } sl_send_reply("404","Not here"); } exit; } I'm using TCP as the transport. Kamailio: 4.2.3 Freeswitch: 1.5.15 I've attached a SIP trace from the point of view of Kamailio. UAC: 10.0.0.40 Kamailio: 10.0.5.50 Freeswitch: 10.0.5.52 What is going on here? The whole Record-Route thing confuses me a lot. Regards, Grant

3 5

Fritzbox issues kamailio 4.0.7 regarding stale nonce?
by Daniel Tryba 08 Apr '15

08 Apr '15

Did anyone notice issues with Fritzbox 7390 devices? When upgrading from 4.0.1 to 4.0.7 some of these devices fail to register after some time. They keep trying to reuse an old nonce and ignore the new one in the 401 response to the REGISTER. It will try 11 times to register with the same cseq, nonce and nc, then it will try to register (11 times) with cseq+1, nc+1 and still the old nonce. So www_auth fails with -4: stale nonce. I could find anything in kamailio auth module to accept this behavior. -- Telefoon: 088 0100 700 Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024

2 4

Kamailio V 2.4.2 Dialog_ng crash and IMS_USRLOC_SCSCF doc
by Yasin CANER 08 Apr '15

08 Apr '15

3 6

Re: [SR-Users] sr-users Digest, Vol 119, Issue 4
by Yufei Tao 08 Apr '15

08 Apr '15

Hi Daniel Thanks for the reply! Here are things we use: * keepalived for HA * Ubuntu 12.04 There seems to be problems when Kamailio operates on both UDP and TCP at the same time. But I'll upgrade Kamailio from v4.0.0. to v4.0.7 first and see what we get, since there are a few bug fixes to do with TCP between these two versions. Cheers, Yufei On 4 April 2015 at 11:00, <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: SIP messages over UDP with sizes over MTU (Yufei Tao) > 2. Re: SIP messages over UDP with sizes over MTU > (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 3 Apr 2015 12:04:49 +0100 > From: Yufei Tao <yufei.tao(a)gmail.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU > Message-ID: > < > CAJwP0iTBGk8S91CheS+xAON2fJgyKZH9HFcwTbvcLygmzLNd+g(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi Daniel > > We don't have pike module enabled. > > When the problem occurred on an VIP, we observed these: > * Kamailio stopped responding to any messages that were sent to the VIP, > not just OPTIONS > * The OPTIONS messages are not big. But the other SIP messages, e.g. some > of the INVITE/OK that came from some SBCs can be big > * netstat showed the Recv-Q on that VIP had a lot of bytes accumulated, > while Kamailio was not seeing/reading them > * Kamailio responded fine on its real IP, when I sent OPTIONS pings to it > using sipsak > * After restarting Kamailio it started working. But after 1 week or so the > same problem happened again > > Since this only happened after running for 1 week or so, we didn't have any > traces to show what exactly happened at the particular time when it > happened. It is possible some SIP messages may have come in fragmented and > some are just too big, depending on the route they came from etc. So I was > wonder if it was possible that the UDP receive buffer was filled somehow > with messed-up messages. Is there anyway to check this? Any suggestions > where I should start looking please? Or is it generally a bad idea to use > UDP when there are messages that may be too big, either fragmented or not? > > Since the it is running in the production environment, I'd like to get some > confidence that a Kamailio upgrade will fix the problem first before I > change anything there. > > Cheers, > Yufei > > > > Date: Wed, 01 Apr 2015 16:27:44 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU > Message-ID: <551C0060.3010002(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > first, not related to the topic you reported, I would recommend at least > upgrading to latest 4.0.x, there were many fixes from v4.0.0 and there > are no changes to be done to config or database -- simply deploy latest > 4.0.x and restart. > > Back to this topic. Is all the other traffic handled apart of big OPTIONS? > > Do you have pike module enabled? If yes, can you double check and be > sure that the SBC is not blacklisted (traffic from it should not be > handled via pike_check_req()). > > Cheers, > Daniel > > On 01/04/15 15:42, Yufei Tao wrote: > > Hi > > > > We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP > > traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio > > responds to the OPTIONS pings with OK, i.e. in the main route block: > > if (is_method("OPTIONS")) > > { > > sl_send_reply("200","OK"); > > exit; > > } > > > > All works fine for a week or so, then Kamailio stopped responding to > > the OPTIONS pings on the VIP it listens to. But it still respond to > > OPTIONS pings that are sent to its real IP. The real IP is not used > > for receiving/sending any traffic while only the VIP is. So it seems > > that Kamailio is still working, but maybe having problems with the > > receiving buffer for the VIP? > > > > We do see that some SIP messages sent to Kamailio's VIP are too big > > (sometimes over 1500 bytes). My question is, in this case, what would > > be expected to happen? Is it possible somehow the receiving buffer for > > the VIP got messed up by the big UDP messages? Any one seen similar > > problems? What is the suggested solution? > > > > We're considering moving to TCP. But since this is production > > environment, I want to get some confidence that the problem we saw was > > likely to have been caused by the UDP message being too large. > > > > Cheers, > > Yufei >

1 0

File tranfere with Kamailio server(postgres)
by Vivek Bohra 08 Apr '15

08 Apr '15

Hi all, I want to know that file transfere service implementation by webrtc client api can be implemented with kamailio server. Best Regards vivek

2 4

Re: [SR-Users] Reg. invalid cseq error. (Daniel-Constantin Mierla)
by Badri Ranganathan 08 Apr '15

08 Apr '15

Thanks a lot Daniel. It was indeed the Cseq coming out of order from my box. Your comment helped me in identifying the problem. Regards, Badri. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: 03 April 2015 11:00 To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 119, Issue 3 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: Make deb (Daniel-Constantin Mierla) 2. Re: [sr-dev] Announcement: Kamailio is now systemd-rtc-server (Alex Balashov) 3. Re: Kamailio coredump when "INVITE" enter in failure route (Jos? Seabra) 4. Re: Kamailio coredump when "INVITE" enter in failure route (Daniel-Constantin Mierla) 5. Re: Upgrade from Sip Express Router 0.9.6 (Andres) 6. Dialog based CDRs and failed dialogs (Mickael Marrache) 7. Re: Upgrade from Sip Express Router 0.9.6 (Daniel-Constantin Mierla) 8. Re: Dialog based CDRs and failed dialogs (Daniel-Constantin Mierla) 9. Re: Upgrade from Sip Express Router 0.9.6 (Andres) 10. Re: Dialog based CDRs and failed dialogs (Mickael Marrache) 11. Kamailio v4.2.4 Released (Daniel-Constantin Mierla) 12. Reg. invalid cseq error. (Badri Ranganathan) 13. Re: Reg. invalid cseq error. (Daniel-Constantin Mierla) 14. Re: Dialog based CDRs and failed dialogs (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Thu, 02 Apr 2015 12:07:06 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: jay binks <jaybinks(a)gmail.com> Cc: "Kamailio $SER$ - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Make deb Message-ID: <551D14CA.4030201(a)gmail.com> Content-Type: text/plain; charset="utf-8" Do you see db_cassandra being compiled in the logs printed by 'make deb'? Cheers, Daniel On 02/04/15 03:37, jay binks wrote: > I have managed to confirm that "make deb" does not actually build > db_cassandra but it does make a deb for it :) ( has changelog etc but > no db_cassandra.so file ) > > why does it not build the .so when calling make deb ?? > can someone point me in the direction to debug that part. > > doesnt seem to be related to the debian rules or control file .. > > is there something like modules.lst for when using "make deb" ? > > > On 2 April 2015 at 00:24, Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> wrote: > > The spec files to build debian packages are kept inside > 'pkg/kamailio/deb/' directory, groupped per distribution name > inside specific sub-directories (using debian and ubuntu version > names -- the one 'debian' is the generic one, but for a specific > version you should choose the one matching the name). > > Then I am not that familiar with debian spec files to be able to > advice by heart, however you should find easily on the web many > sources of information about the files inside those folder and how > to build deb packages. > > Say you want to build for debian wheezy: > > - edit the files inside pkg/kamailio/deb/wheezy/ to fit your needs > - in the root directory with kamailio sources do: > > ln -s pkg/kamailio/deb/wheezy debian > make deb > > And you should get the packaging process started. > > Cheers, > Daniel > > > On 01/04/15 14:46, jay binks wrote: >> Sorry, this confused me a little.. >> can you give an example of what you mean ? >> >> On 1 April 2015 at 22:40, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> you have to edit spec files and remove the other modules in >> the folder matching your debian distribution inside >> 'pkg/kamailio/deb/'. This is the only way I know to change >> the packaging content for debs. Then make a symlink inside >> root folder of kamailio server named 'debian' to the >> respective folder inside 'pkg/kamailio/deb/' and run 'make deb'. >> >> Cheers, >> Daniel >> >> >> On 01/04/15 13:50, jay binks wrote: >>> is there any way to make deb packages for only 1 of the >>> modules ?? >>> currently it takes a long time to run "make deb" >>> >>> im simply trying to build and package db_cassandra ( need >>> the package for my environment ) >>> >>> As a side note, is there any chance db_cassandra could be >>> added to the kamailio deb packages ? >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> >>> _______________________________________________ >>> 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 >> >> -- >> Daniel-Constantin Mierla >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda >> Kamailio World Conference, May 27-29, 2015 >> Berlin, Germany - http://www.kamailioworld.com >> >> >> _______________________________________________ >> 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 >> >> >> >> >> -- >> Sincerely >> >> Jay > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda > Kamailio World Conference, May 27-29, 2015 > Berlin, Germany - http://www.kamailioworld.com > > > > > -- > Sincerely > > Jay -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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