sr-users August 2019

sr-users@lists.kamailio.org

65 participants
96 discussions

Re: [SR-Users] Question on Database Version Table

by JR Richardson

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > like Alex said, you can have two version tables, one for each versions > of kamailio you play with and use in config the core parameter to set > the name of that table. > > I am actually writing to say that address table never had accountcode > and notes columns. Likely they were added in your deployment for other > purposes or either the default column names were changed. The default > column names are listed at: > > - > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address > > These were the same for rather long time, iirc the recent version number > changed mainly due to with modifications done to the size, not due to > new columns. > > Cheers, > Daniel > > On 18.06.19 20:50, JR Richardson wrote: > > Hey All, > > > > I'm doing some upgrades and noticed database table version numbers > > have increased. > > > > database table adjustments: > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add > > accountcode and notes column > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add > > ruri_pattern and priority column > > > > I run mysql master/slave so I'm concerned about the master database > > not having the correct version numbers or can I just update the > > version number in the table when I do the upgrade of the kamailio > > versions? > > > > Thanks. > > > > JR

5 years

Re-invite after RTPengine pacemaker switching?

by Жан Базаров

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

5 years

Presence and BLF

by Bill Neely

Hi all: I am trying to set up presence and BLF notifies on Kamailo 5.2.4 I am following the instructions laid out in http://kb.asipto.com/kamailio:presence:k43-blf Don't know if these instructions are valid, but I am definitely not getting any light action (using Grandstream 2130 and Grandstream Wav for testing.) Entries in active_watchers appear to be correct. Here is my cfg file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v5.1 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** 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" #!define WITH_DEBUG #!define WITH_NAT #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_DLG 9 #!define FLT_DLGINFO 10 #!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=3 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* 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:62.151.xx.xx:50060 # allow http requests listen=tcp:127.0.0.1:8080 tcp_accept_no_cl=yes #!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 = "104.192.xx.xx" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.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 "acc.so" loadmodule "counters.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" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.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 # load the module loadmodule "xhttp.so" loadmodule "json.so" loadmodule "pua_json.so" loadmodule "dialog.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # 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) modparam("presence", "server_address", "sip:62.151.xx.xx:50060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!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) modparam("debugger", "log_level_name", "exec") #!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 { xlog("L_ALERT","ALERT: REQUEST ROUTE ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 $rm from $fu (IP:$si:$sp) $du\nMESSAGE BUFFER:\n$mb\n\n\nRU:$ru\n"); $var(i)=0; while($var(i)<$branch(count)) { xlog("$(branch(uri)[$var(i)])\n"); $var(i) = $var(i) + 1; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle registrations route(REGISTRAR); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); xlog("L_ALERT","ALERT: line 492******************************** $rm from $fu (IP:$si:$sp)\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_ALERT","ALERT: line 494******************************** $rm from $fu (IP:$si:$sp)\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } event_route[xhttp:request] { if ($hu != "/presence/") { xlog("L_INFO", "invalid request [$hu]"); xhttp_reply("404", "Not Found", "text/html", ""); return; } $var(call-id) = $(rb{json.parse,Call-ID}); xlog("L_INFO", "$var(call-id)|log|[$hu] payload $rb"); if ($(rb{json.parse,Event-Package}) == "dialog") { xlog("L_INFO", "$var(call-id)|log|received $(rb{json.parse,Event-Package}) update for $(rb{json.parse,From}) $rb"); pua_json_publish($rb); } xhttp_reply("200", "OK", "text/html", "$rb"); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); 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()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } 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 processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication 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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { xlog("ALERT:NATMANAGE********************* (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; #!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; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("corwie"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains # Routing to foreign domains route[SIPOUT] { xlog("ALERT: SIPOUT (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } route[TOAST] { rewritehostport("45.33.xx.xx:5060"); $fu = "sip:7028743833@62.151.182.83:5012"; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing 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 # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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 }

5 years, 2 months

Kamalio & Asterisk integration erroneous behavior, setup issue

by Aristeidis Tsitras

new to the area and trying to setup Kamailio with Asterisk in a single machine. All users will register to Kamailio's port and in case of need for media, it will be forwarded to Asterisk, that is my intention. All of my work is based on the following link https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Here is what i have done: - Debian 8, 64 bit machine with mysql and odbc - *root@kamast: ~ $ lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.11 (jessie) Release: 8.11 Codename: jessie root@kamast: ~ $ uname -a Linux kamast 3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux root@kamast: ~ $ * - Kamailio 5.2 installed from Kamailio's deb repository - Asterisk 13LTS installed from source - Used the same passwords such as kamailiorw and asterisk_password, since this is a test system, for proof of concept. I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202. Then created in sip.conf the same 3 users with the same credentials. Then on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a softphone. Problems: - Cannot see the users in the Asterisk's cli, sip show peers - I can see users only in Kamailio with kamctl ul show - A call between the extensions goes to voicemail. It never reaches the other destination eg 2200 calls 2201 and in Asterisk's console i am getting a message that 2201 is absent and it goes to voicemail. The same with any other extension. Attached you can find: 1. Kamailio.cfg 2. Asterisk's sip.conf 3. Asterisk's extension.conf 4. The import that i have done to mysql for the user creation. I would appreciate if someone could point me to the error and help me fix it please?

5 years, 2 months

$Au being set to $fu instead of $fU

by Joel Serrano

Hi, Testing on latest master deb nightly build from last night I've noticed that when $au is not available, $Au is being set to $fu (uri) instead of $fU (username). Docs: https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables ... $Au - Acc username: username for accounting purposes. It's a selective pseudo variable (inherited from acc module). It returns auth username ($au) if exists or From username ($fU) otherwise. ... I have the following log in several places in a test box: xlog("L_NOTICE", "[DEBUG] Au=$Au au=$au aU=$aU fU=$fU fu=$fu - \n"); And when I run through them (on both request and reply routes) I get the following: Aug 8 14:37:43 csbc01 csbc[9478]: NOTICE: {1 21564 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9479]: NOTICE: {1 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=1002366 aU=1002366 fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9484]: NOTICE: {2 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Those 3 logs belong to the same flow from different places in the config script. I would expect $Au to never have a "@some.domain" in it, so I assume in some place the replacement is not being done with $fU but with $fu. @Devs: I'm happy to create a GH issue for tracking etc if required, please let me know! Cheers, Joel.

5 years, 2 months

Planning the next major release - v5.3.0

by Daniel-Constantin Mierla

Hello, it's about the time to set the milestones to the next major release - v5.3.0. There are 6 new modules and many other enhancements to existing components in git master branch, also during the last irc devel meeting it was proposed to freeze after summer holidays. At this moment I would propose Wednesday, September 4, 2019 to freeze the development. Then the release should be about one month later, after testing interval. In case there are people willing a bit more time for development, we can freeze one week later, September 11, or another date -- your suggestions here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 years, 2 months

Kamailio stops processing requests over UDP

by Juha Heinanen

I'm returning to an old issue, where Kamailio stops processing incoming requests. Now it turned out that only requests over UDP are not processed. Requests over TCP work fine. Debug log doesn't show anything about the requests over UDP, but wireshark sees them coming. core.ps shows that main process - attendant and all udp receiver processes are alive. Any ideas what could cause Kamailio (5.2 version) to stop processing requests over UDP? What additional info should I try to get? -- Juha

5 years, 2 months

Unexpected BYEs from dialog module

by Alex Balashov

Hi, I am running 4.3.5:1b0c0a, which I am aware is an EOL'd release train, and have a problem with the dialog module I am baffled by. On many calls - I can't find any correlation to particular kinds of endpoints - I see spoofed BYEs come out of Kamailio after a minute and a half, as if the call had hit a dialog timeout or the dialog module's dead peer detection (OPTIONS pinging) kicked it out. The thing is, neither of those explanations seem to be borne out by the parameters of the calls under investigation. The dialog timeout on these calls is set to 7200, and the dialog keepalives aren't enabled in either direction. If they were, it'd be logged. Calls that are terminated in this manner also see the following log message: WARNING: dialog [dlg_req_within.c:214]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7f9997317e48 [3390:9644] with clid '1996679936_133218050(a)x.x.x.x' and tags '2bb17663-co4006-INS001' 'as7925780d' But I don't think this is unusual for a dialog-spoofed BYE; presumably this is due to a 200 OK for the BYE from both ends. So anyway, I am trying to track down anything else that could inadvertently cause dialog to hang up calls relatively quickly in that release, or inadvertently set the dialog timeout parameter to something lower than is apparent from the logging and the config. If anyone has any pointers, that would not go unappreciated! Many thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 years, 2 months

Issue with uac_replace

by David Costa

Greetings, I have a Kamailio proxy with uac_replace for To and From Headers. So far everything has worked correctly. However, I now have a client who encodes my Calling Number too and I'm getting some issues in the restore mechanics. I have the restore setup as "auto" and it's info is stored in "vsf" and "vst" parameters of Record-Route. On replies sent by my client I have the following Record-Route and From Header (Note : My Kamailio has two IPs which makes it insert two RR headers ): Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> From: "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this case the restore is done correctly When the client sends me a BYE request I have the following Route Header and To Header Route : <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2>,<sip:xxx.xxx.xxx. xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_o=128_11_Y;vsf=AA AAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcH wIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> To : "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this scenario, the To after restore is : To :"+351411110097" <sip:Q4525417L_<>Gxxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4, which throws an internal error on Kamailio about bad URI and also makes our MediaGateway reply with a "Bad Request". Can this error be caused by the restore of the uac_replace mechanics? Both Record Route URIs are on different headers while the both Route URIs are on the same Route header. Can this be the reason? If you need more information please let me know. Best Regards David Costa

5 years, 2 months

Re: [SR-Users] controlling presence with new module `pua_json`

by Henning Westerholt

Hello Bill, (please keep the list in CC) yes, this seems to be the case. Just looked briefly to the code, it does an assignment in some cases. E.g. if no Presentity* data is given it will just use the From* information. And then it should later output an error from pua_json_update_presentity(..) if there is not the necessary data. Do you see any error message like this? Another idea - does it works if you add more "missing" data in the curl requests? Cheers, Henning Am 28.08.19 um 21:16 schrieb bill(a)novatrope.us: > Henning: Good point. > > I do see a number of errors in the log. It looks like the json routine > is looking for data that is not in the http query as specified by the > author. > > > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Name: [update] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Package: [dialog] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-User] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-User: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-Realm] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-Realm: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): From: > [sip:1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-User: [1020101] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Realm: [<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To: > [sip:1020108@<IP of server>:50060] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > To-User: [1020108] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Call-ID: [1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Direction] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Direction: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): State: > [Confirmed] > 17(125579) DEBUG: presence [event_list.c:348]: search_event(): start > event= [dialog/5] > > On 8/28/19 12:00 PM, Henning Westerholt wrote: >> Hello Bill, >> >> just giving some generic debug pointers here: >> >> - do you see any errors in the kamailio log file? >> >> - do you see any SIP traffic (e.g. by observing with ngrep, wireshark >> etc..) to the phone after your curl command? >> >> Cheers, >> >> Henning >> >> Am 28.08.19 um 19:04 schrieb bill(a)novatrope.us: >>> Hi all: >>> >>> I am trying to use the procedure for setting BLF lights described in >>> https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… >>> >>> >>> WIthout success. >>> >>> My configuration is 1 Grandstream GXP2130 phone behind a NAT on a >>> public IP. (IPP in example below) >>> >>> Kamailio is running on public IP (IPK in example) listening on port >>> 50060 >>> >>> The command I am sending: >>> >>> curl -d >>> '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' >>> >>> http://localhost:8080/presence/ >>> {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} >>> >>> >>> >>> >>> Kamailio appears to be processing the command OK, so I think I am just >>> not setting it up properly> >>> >>> Anybody having any luck with this procedure, let me know how you are >>> getting it to work. >>> >>> >>> Bill >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

5 years, 2 months

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sr-users August 2019