sr-users December 2017

sr-users@lists.kamailio.org

73 participants
101 discussions

blf with kamailio not sending NOTIFY
by Ravi Kuru 12 Dec '17

12 Dec '17

Hi, I set up BLF but not working, phone sending SUBSCRIBE message then kamailio is sending 202 OK but not sending NOTIFY message. What could be the issue. Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg [root@rpx1 kamailio]# vi [root@rpx1 kamailio]# cd [root@rpx1 ~]# vi a Request-Line: SUBSCRIBE sip:8665336444_202@67.227.65.10 SIP/2.0 Method: SUBSCRIBE Request-URI: sip:8665336444_202@67.227.65.10 Request-URI User Part: 8665336444_202 Request-URI Host Part: 67.227.65.10 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 66.46.51.154:21389;branch=z9hG4bK1022076515;rport Transport: UDP Sent-by Address: 66.46.51.154 Sent-by port: 21389 Branch: z9hG4bK1022076515 RPort: rport From: <sip:8665336444_201@67.227.65.10>;tag=1691126593 SIP from address: sip:8665336444_201@67.227.65.10 SIP from address User Part: 8665336444_201 SIP from address Host Part: 67.227.65.10 SIP from tag: 1691126593 To: <sip:8665336444_202@67.227.65.10> SIP to address: sip:8665336444_202@67.227.65.10 SIP to address User Part: 8665336444_202 SIP to address Host Part: 67.227.65.10 Call-ID: 1930543912-5060-5(a)BA.B.B.BEC CSeq: 20031 SUBSCRIBE Sequence Number: 20031 Method: SUBSCRIBE Contact: <sip:8665336444_201@66.46.51.154:21389> Contact URI: sip:8665336444_201@66.46.51.154:21389 Contact URI User Part: 8665336444_201 Contact URI Host Part: 66.46.51.154 Contact URI Host Port: 21389 Proxy-Authorization: Digest username="8665336444_201", realm="67.227.65.10", nonce="Wil7QVopehWJO/uGzcWNmD3vlFL39ztq", uri=" sip:8665336444_202@67.227.65.10", response="c7bf71e9f7f4d9720655531cbd768cff", algorithm=MD5 Authentication Scheme: Digest Username: "8665336444_201" Realm: "67.227.65.10" Nonce Value: "Wil7QVopehWJO/uGzcWNmD3vlFL39ztq" Authentication URI: "sip:8665336444_202@67.227.65.10" Digest Authentication Response: "c7bf71e9f7f4d9720655531cbd768cff" Algorithm: MD5 X-Grandstream-PBX: true [Expert Info (Note/Undecoded): Unrecognised SIP header (X-Grandstream-PBX)] [Message: Unrecognised SIP header (X-Grandstream-PBX)] [Severity level: Note] [Group: Undecoded] Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.29 Expires: 1800 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Session Initiation Protocol (202) Status-Line: SIP/2.0 202 OK Status-Code: 202 [Resent Packet: False] [Request Frame: 295] [Response Time (ms): 0] Message Header Via: SIP/2.0/UDP 66.46.51.154:21389 ;branch=z9hG4bK1022076515;rport=21389;received=66.46.51.154 Transport: UDP Sent-by Address: 66.46.51.154 Sent-by port: 21389 Branch: z9hG4bK1022076515 RPort: 21389 Received: 66.46.51.154 From: <sip:8665336444_201@67.227.65.10>;tag=1691126593 SIP from address: sip:8665336444_201@67.227.65.10 SIP from address User Part: 8665336444_201 SIP from address Host Part: 67.227.65.10 SIP from tag: 1691126593 To: <sip:8665336444_202@67.227.65.10 >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-fbce SIP to address: sip:8665336444_202@67.227.65.10 SIP to address User Part: 8665336444_202 SIP to address Host Part: 67.227.65.10 SIP to tag: a6a1c5f60faecf035a1ae5b6e96e979a-fbce Call-ID: 1930543912-5060-5(a)BA.B.B.BEC CSeq: 20031 SUBSCRIBE Sequence Number: 20031 Method: SUBSCRIBE Expires: 1800 Contact: <sip:67.227.65.19:5060> Contact URI: sip:67.227.65.19:5060 Contact URI Host Part: 67.227.65.19 Contact URI Host Port: 5060 Server: kamailio (5.0.3 (x86_64/linux)) Content-Length: 0 this is the my kamailio.cfg Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_PSTN #!define WITH_NAT #!define WITH_ANTIFLOOD #!define WITH_PRESENCE # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB "kamailio.cfg" 1232L, 34401C Last login: Mon Dec 11 10:56:53 on ttys000 raveechdransmbp:~ rkuru$ ssh -p44422 kuru.ravi(a)67.227.65.10 kuru.ravi(a)67.227.65.10's password: Last login: Sun Dec 10 10:25:45 2017 from 216.123.66.49 [kuru.ravi@rpx1 ~]$ su - Password: Last login: Sun Dec 10 10:25:53 EST 2017 on pts/0 [root@rpx1 ~]# cd /usr/local/etc/kamailio/ [root@rpx1 kamailio]# vi kamailio.cfg # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #mhomed=1 #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "unixodbc://telcan:t3lc4n@localhost/MySQLkamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "unixodbc://test:test@localhost/MySQLasterisk" #!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 ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="rp-na1.sipconfig.net" alias="na-rpx1.sipconfig.net" /* 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:38.64.154.234:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: #pstn.gw_ip = "67.227.65.2" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "67.227.65.10" desc "PSTN GW Address" #!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 #!ifdef WITH_ASTERISK asterisk.bindip = "67.227.65.10" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "67.227.65.10" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "db_unixodbc.so" #loadmodule "mi_fifo.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" #loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "sqlops.so" loadmodule "dialog.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 #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ modparam("registrar", "append_branches", 1) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 4) modparam("registrar", "retry_after", 60) # max value for expires of registrations #modparam("registrar", "min_expires", 0) modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 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) #modparam("registrar", "retry_after", 30) # max value for expires of registrations #modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU #modparam("registrar", "gruu_enabled", 1) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) 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", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") ##!ifdef WITH_ASTERISK #modparam("auth_db", "user_column", "name") #modparam("auth_db", "password_column", "sippasswd") #modparam("auth_db", "db_url", DBASTURL) #modparam("auth_db", "version_table", 0) ##!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) ##!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # 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:67.227.65.10:5060" ) 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) # ----- 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:127.0.0.1:7722") # ----- 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", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 3) modparam("pike", "reqs_density_per_unit", 1000) modparam("pike", "remove_latency", 10) # ----- 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 # ------- Load-balancer params ------ modparam("dispatcher", "db_url","unixodbc://test:test@localhost /MySQLkamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@67.227.65.10") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); xlog("L_INFO","AFTER LOCATION to ROUTE DEVICE ------- : $ci"); ####route(RELAY); } 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(); } xlog("L_INFO","----- This is to device < $du > asterisk server -------"); if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); 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()) 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 statelessy 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_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$"){ if(sd_lookup("speed_dial")){ route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("REFER")){ xlog("L_INFO"," WITH_ASTERISK WITHINDLG SIP REFER METHOD -------"); } #!endif xlog("L_INFO"," FROM ASTERISK TO DEVICE -------"); $avp(oexten) = $rU; if (!lookup("location")) { xlog("L_INFO","LOOK UP LOCATION FAILED -------"); $var(rc) = $rc; xlog("L_INFO"," *********** $var(rc) *****************"); #route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } xlog("L_INFO","LOCATION FROM ASTERISK TO DEVICE -------"); # 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] { # 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; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ##!ifdef WITH_ASTERISK # if (!auth_check("$fd", "sipusers", "1")) { ##!else if (!auth_check("$fd", "subscriber", "1")) { ##!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("L_INFO","SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { #route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk # Send to Asterisk route[TOASTERISK] { # only local users allowed to call if(is_method("REFER")){ xlog("L_INFO"," SIP REFER METHOD -------"); } if(from_uri!=myself) { xlog("L_INFO","[$fU@$si:$sp]{$rm} *************** From Asterisk Box to Device$du\n"); }else{ if(is_method("INVITE")){ xlog("L_INFO"," device to Asterisk Box $du ++++\n"); route(RELAY); exit; } } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } ds_mark_dst("P"); $var(x) = $au; $var(x) = $(var(x){s.substr,0,3}); if(!ds_select_dst($(var(x){s.int}), "1")) { $var(rip) = $sel(cfg_get.asterisk.bindip); }else{ $var(rip1) = $du; $var(rip) = $(var(rip1){s.replace,:5060,}) ; $var(rip) = $(var(rip){s.replace,sip:,}) ; xlog("L_INFO","$au <------- Loadbalancer REGISTER to Asterisk $var(rip) ------->"); } if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 2

newbie question - routing sip-calls
by thomas peterseil 12 Dec '17

12 Dec '17

hello, sorry for this very basic question, but i am a new kamailio user. i installed kamailio and calls between users work fine. now i would like to make it possible to route outbound sip calls. so when a user dials for example 21 the call should be routed to wbdemo(a)zipdx.com thank you so much for your help! best regards, thomas peterseil

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Re: [SR-Users] Unable to enable TLS on Kamailio
by otron2016＠gmail.com 12 Dec '17

12 Dec '17

Broken is in the eyes of the beholder: well designed cryptographic code wants to ensure that information (keys, cleartext) doesn't leak via unsanitized memory (there are many ways, both within and beyond calling programs); the easy and more foolproof way to do that for the cryptography programmer is often to use a memory manager that takes care of that, such as jemalloc (with appropriate configuration parameters). If you make security representations (and the certificate is reasonably construed to make a security representation) you shouldn't bypass this unless you verify that you prevent all possible information leaks. From armslength, you might just try to use jemalloc as kamailio's mm library, but even there it would be necessary to be really careful about kamailio freeing sensitive memory immediately after use--everywhere that happens. That's why it's probably easier to just let a properly implemented crypto library do what it's designed to do. Sent from Samsung Mobile -------- Original message -------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: 12/12/2017 2:26 AM (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>,Tomi Hakkarainen <tpaivaa(a)gmail.com> Subject: Re: [SR-Users] Unable to enable TLS on Kamailio Hello, there were some broken versions of openssl that didn't allow anymore to set custom memory manager. The only option is to upgrade libssl to a version that doesn't expose the issue. If you search on kamailio issues tracker on gihub.com, there should be one closed about this topic. Cheers, Daniel On 11.12.17 22:20, Tomi Hakkarainen wrote: Hi, I have problem to enable TLS on just installed Kamailio server openSUSE 42.3 (x86_64) VERSION = 42.3 CODENAME = Malachite version: kamailio 5.0.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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 18:06:25 Dec 3 2017 with gcc 4.8.5 I get this on debug log: 0(11336) DEBUG: <core> [core/cfg.y:1642]: yyparse(): loading modules under /usr/lib64/kamailio/modules/ loading modules under config path: /usr/lib64/kamailio/modules/ 0(11336) DEBUG: <core> [core/cfg.y:1623]: yyparse(): loading module tls.so 0(11336) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/tls.so> 0(11336) DEBUG: <core> [core/mem/q_malloc.c:189]: qm_malloc_init(): qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 0(11336) DEBUG: <core> [core/mem/q_malloc.c:191]: qm_malloc_init(): qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=235152 0(11336) DEBUG: <core> [core/mem/q_malloc.c:193]: qm_malloc_init(): qm_malloc_init(0x7f6e001cb000, 67108864), start=0x7f6e001cb000 0(11336) DEBUG: <core> [core/mem/q_malloc.c:202]: qm_malloc_init(): qm_malloc_init: size= 67108864, init_overhead=235256 0(11336) ERROR: tls [tls_init.c:595]: tls_pre_init(): Unable to set the memory allocation functions 0(11336) ERROR: tls [tls_init.c:597]: tls_pre_init(): libssl current mem functions - m: 0x7f6e055b33d0 r: 0x7f6e055b3a30 f: 0x7f6e055b39a0 0(11336) ERROR: tls [tls_init.c:599]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(11336) ERROR: <core> [core/sr_module.c:607]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(11336) CRITICAL: <core> [core/cfg.y:3411]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 150, column 12-19: failed to load module for resolving have compiled openssl from 1.0.2j-fips to openssl version OpenSSL 1.0.2n 7 Dec 2017 Is this information enough to see what we are missing Will provide more info if needed. Any help and suggestions are appreciated. Regards, T _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Dispatcher flow regarding weight and priority.
by Joel Serrano 12 Dec '17

12 Dec '17

Hi all, I have a doubt regarding how dispatcher selects a node... >From the docs: *[...]* *alg* - the algorithm used to select the destination address. The parameter can be an integer or a variable holding an interger. - “4” - round-robin (next destination). - “8” - select destination sorted by priority attribute value (serial forking ordered by priority). - “9” - use weight based load distribution. You have to set the attribute 'weight' per each address in destination set. *[...]* I normally use "4", so dispatcher round-robins through the destinations of a setid, but I have now the following doubt: If a trunk has 3 nodes, say 1.1.1.1, 2.2.2.2, 3.3.3.3 Can I do: Send all calls to 1.1.1.1, if it goes down, then do a 80%-20% between 2.2.2.2 and 3.3.3.3. if the trunk has a correct DNS SRV setup with weight and priority it's easy to achieve, but in this case there is no SRV or nothing, just 3 IPs. So can this logic be achieved with dispatcher directly? Can do some combination of alg "8" and alg "9"? Or would the correct approach need two separate setid groups and handle that logic in failure route? Thanks, Joel.

2 2

pua.publish using RPC
by Alex Balashov 12 Dec '17

12 Dec '17

Hello, Is pua.publish strictly an MI function, or is it possible nowadays to call it via the RPC channel, and specifically, using jsonrpc_exec()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio does not add port number to addresses in Via, Record-Route headers
by George Diamantopoulos 12 Dec '17

12 Dec '17

Hello again, Indeed this issue does not manifest at all. I'm awfully sorry for the false alarm, and on release day no less! The problem was there was a lingering DNAT rule in iptables, which would translate port 5066 to port 5060. The deployment script injected this as it was carried over from our legacy platform. Of course, I kept banging my head against the wall here because sngrep wouldn't show the DNAT's effect as it captures traffic from the NIC directly: it would show a REGISTER arriving on 5066, but the dport was masqueraded before being handed over to kamailio. Similarly for the outgoing INVITE. NAT is wrong in so many ways... :-) BR, George On 11 December 2017 at 18:17, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > I did a quick test and all looks fine, ports are set in via and > record-route, in my config I have: > > record_route(); > > $fs="udp:127.0.0.1:5080"; > $du = "sip:127.0.0.1:9"; > t_relay(); > exit; > > Then sending an OPTIONS resulted in the trace shown below. > > Cheers, > Daniel > > U 2017/12/11 17:14:47.108430 127.0.0.1:56729 -> 127.0.0.1:5060 > OPTIONS sip:test@127.0.0.1 SIP/2.0. > Via: SIP/2.0/UDP 192.168.178.84:62516;branch=z9hG4bK.3aaddf68;rport;alias. > From: sip:sipsak@192.168.178.84:62516;tag=16d1c24. > To: sip:test@127.0.0.1. > Call-ID: 23927844(a)192.168.178.84. > CSeq: 1 OPTIONS. > Contact: sip:sipsak@192.168.178.84:62516. > Content-Length: 0. > Max-Forwards: 70. > User-Agent: sipsak 0.9.7pre. > Accept: text/plain. > . > > > U 2017/12/11 17:14:51.010251 127.0.0.1:5080 -> 127.0.0.1:9 > OPTIONS sip:test@127.0.0.1 SIP/2.0. > Record-Route: <sip:127.0.0.1:5080;r2=on;lr>. > Record-Route: <sip:127.0.0.1;r2=on;lr>. > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK61bd. > b2882fea15c488761489f8ef588efbe1.0. > Via: SIP/2.0/UDP 192.168.178.84:62516;received=127.0.0.1;branch=z9hG4bK. > 3aaddf68;rport=56729;alias. > From: sip:sipsak@192.168.178.84:62516;tag=16d1c24. > To: sip:test@127.0.0.1. > Call-ID: 23927844(a)192.168.178.84. > CSeq: 1 OPTIONS. > Contact: sip:sipsak@192.168.178.84:62516. > Content-Length: 0. > Max-Forwards: 69. > User-Agent: sipsak 0.9.7pre. > Accept: text/plain. > . > > On 11.12.17 16:37, George Diamantopoulos wrote: > > Hello all, > > I have the following issue in my configuration, tested with 5.2.0-rc1 so > far: > > At some point, I set the $fs pseudovariable to force a request to be > relayed through a specific socket. Although this is honoured by kamailio > (i.e. the request does indeed leave the kamailio host from the respective > socket), the port number is not added to the Via and RR headers. As a > result, all replies to the request, as well as all subsequent requests from > the other SIP UA are relayed to the default port, 5060. Here's an example: > > SIP UAC to kamailio: > INVITE 192.168.1.1:5060 ---> 192.168.1.254:5060 > Kamailio to UAS ($fs is set): > INVITE 2.2.2.2:5066 ---> 3.3.3.3:5060 > Topmost Via in request relayed by kamailio is: > SIP/2.0/UDP 2.2.2.2;branch=aaaaaaaaaaaaaa <- port 5066 is not added > Topmost RR in request relayed by kamailio is: > <sip:2.2.2.2;r2=on;lr;did=bbbbbbb;nat=yes> <- port 5066 is not added > RESULT: Reply from UAS is sent to 2.2.2.2:5060 > > Is this behaviour valid? Am I missing anything? Kamailio is configured to > listen on both sockets on IP 2.2.2.2, namely: a) udp:2.2.2.2:5060 and b) > 2.2.2.2:5066. Thanks. > > BR, > George > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com > >

2 1

Change Calling Number
by Duarte Rocha 11 Dec '17

11 Dec '17

Hey I'm trying to do some normalization on my Calling Number after Kamailio chooses the Gateway in which it will deliver the request. So, i'm trying to do the operation in the DISPATCH route. However, in this route, msg_apply_changes() doesn't work and returns (-1). I've tried to do this on a Branch route but Kamailio doesn't allow me to use msg_apply_changes() there and otherwise changes won't be persistent. Is there a way to modify $fU at this stage? Cheers

2 1

Kamailio v5.1.0 Released - new major version is out
by Daniel-Constantin Mierla 11 Dec '17

11 Dec '17

Kamailio v5.1.0 is out – it comes with 9 new modules and a significant set of improvements touching more than 90 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to export most of the config functions to the KEMI framework, allowing to write SIP routing logic in other scripting languages executed by the embedded interpreters for Lua, JavaScript, Python or Squirrel, with the option to reload the routing script without restarting Kamailio. Enjoy Kamailio v5.1.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2018 - http://www.kamailioworld.com

1 0

Kamailio does not add port number to addresses in Via, Record-Route headers
by George Diamantopoulos 11 Dec '17

11 Dec '17

Hello all, I have the following issue in my configuration, tested with 5.2.0-rc1 so far: At some point, I set the $fs pseudovariable to force a request to be relayed through a specific socket. Although this is honoured by kamailio (i.e. the request does indeed leave the kamailio host from the respective socket), the port number is not added to the Via and RR headers. As a result, all replies to the request, as well as all subsequent requests from the other SIP UA are relayed to the default port, 5060. Here's an example: SIP UAC to kamailio: INVITE 192.168.1.1:5060 ---> 192.168.1.254:5060 Kamailio to UAS ($fs is set): INVITE 2.2.2.2:5066 ---> 3.3.3.3:5060 Topmost Via in request relayed by kamailio is: SIP/2.0/UDP 2.2.2.2;branch=aaaaaaaaaaaaaa <- port 5066 is not added Topmost RR in request relayed by kamailio is: <sip:2.2.2.2;r2=on;lr;did=bbbbbbb;nat=yes> <- port 5066 is not added RESULT: Reply from UAS is sent to 2.2.2.2:5060 Is this behaviour valid? Am I missing anything? Kamailio is configured to listen on both sockets on IP 2.2.2.2, namely: a) udp:2.2.2.2:5060 and b) 2.2.2.2:5066. Thanks. BR, George

2 1

Re: [SR-Users] Failed to find command http_client_query
by Alexandru Covalschi 11 Dec '17

11 Dec '17

Hello, Wanted to ask if you found something ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493 > 24 нояб. 2017 г., в 22:19, Alexandru Covalschi <568691(a)gmail.com> написал(а): > > Hello, sharing whole config except vars.cfg file, where values like API_PROTO, DBURL etc. are defined. > > <kamailio_504.cfg> > > > ________________________________ > Alexandru Covalschi > VoIP Engineer and System Administrator > tel: +373 673 98 493 > >> 24 нояб. 2017 г., в 13:21, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >> >> Hello, >> >> can you provide here all the http_client_query(...) you have in your config and list in which route block are they used (like request_route, failure_route, ...)? You can replace IP addresses, usernames, passwords with other strings, but if you use any kamailio variables (e.g. $rU), let them in place. >> Cheers, >> Daniel >> >> On 23.11.17 20:54, Alexandru Covalschi wrote: >>> sip:172.21.0.14:5060 <sip:172.21.0.14:5060> is the local address of the server, it seems like a bug in http_client, as I don't have such http_query call anywhere :/ >>> just re-checked >>> Also the fact that it works perfectly on 4.4.6 tells that it's not a configuration issue >>> >>> ________________________________ >>> Alexandru Covalschi >>> VoIP Engineer and System Administrator >>> tel: +373 673 98 493 >>> >>>> 23 нояб. 2017 г., в 16:59, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>> >>>> There was a mistake in the exports structure, the name of the function with 4 parameter was http_query instead of http_client_query. I pushed a patch to fix it. >>>> Then, related to the issue with cannot resolve the host, the debug messages show an invalid http url being used: >>>> >>>> Nov 21 04:18:20 ast1 /usr/sbin/kamailio[20804]: DEBUG: http_client [functions.c:146]: curL_query_url(): ****** ##### CURL URL [sip:172.21.0.14:5060 <sip:172.21.0.14:5060>] >>>> >>>> The it fails to do the dns stuff with it... >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 22.11.17 22:38, Alexandru Covalschi wrote: >>>>> Just wanted to ask if you need my help to test/check/etc. >>>>> >>>>> ________________________________ >>>>> Alexandru Covalschi >>>>> VoIP Engineer and System Administrator >>>>> tel: +373 673 98 493 >>>>> >>>>>> 21 нояб. 2017 г., в 11:58, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>> >>>>>> Kamailio is actually starting, please take a look at that debug=4 log file: https://filebin.net/r54y3jkmzdyk9zak/syslog_http_client_query_failed.txt <https://filebin.net/r54y3jkmzdyk9zak/syslog_http_client_query_failed.txt> >>>>>> at line 1042 http_client module is successfully loading, but at line 12785 it fails to load http_client_query >>>>>> >>>>>> Also please check line 50385 in https://filebin.net/352vff41fwx7uowg/syslog_http_query_curl_6.txt <https://filebin.net/352vff41fwx7uowg/syslog_http_query_curl_6.txt> >>>>>> I'm using http_query here with 5.0.4 http_client, you can see that http_client makes a request, but afterwards it returns curl error 6, but the weirdest thing is - at line 19887 I request token from the very same IP address, only url is changed, and it actually works fine. >>>>>> ________________________________ >>>>>> Alexandru Covalschi >>>>>> VoIP Engineer and System Administrator >>>>>> tel: +373 673 98 493 >>>>>> >>>>>>> 21 нояб. 2017 г., в 10:44, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>>>>> >>>>>>> But then kamailio is not starting, or? >>>>>>> >>>>>>> Can you provide all the error message as they are printed in syslog file? File, line and other details are part of the logs message, allowing to identify the location in the source code where that happens ... >>>>>>> >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 21.11.17 09:38, Alexandru Covalschi wrote: >>>>>>>> It is still one of the issues >>>>>>>> >>>>>>>> ________________________________ >>>>>>>> Alexandru Covalschi >>>>>>>> VoIP Engineer and System Administrator >>>>>>>> tel: +373 673 98 493 >>>>>>>> >>>>>>>>> 21 нояб. 2017 г., в 9:40, Daniel-Constantin Mierla <miconda(a)gmail.com <mailto:miconda@gmail.com>> написал(а): >>>>>>>>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> in your email in this thread, you said: >>>>>>>>> >>>>>>>>> > When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". >>>>>>>>> >>>>>>>>> Is it still the case, or that was sorted out? >>>>>>>>> >>>>>>>>> Cheers, >>>>>>>>> Daniel >>>>>>>>> >>>>>>>>> On 20.11.17 22:43, Alexandru Covalschi wrote: >>>>>>>>>> Works ok on 4.4.6. So I suppose it's a weird 5.0.4 + my environment bug. I'm rolling back to 4.4.6 right now, will keep one 5.0.4 machine for testing (on debian 9) >>>>>>>>>> Let me know if I can provide any dumps/cores etc. >>>>>>>>>> >>>>>>>>>> ________________________________ >>>>>>>>>> Alexandru Covalschi >>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>> >>>>>>>>>>> 20 нояб. 2017 г., в 23:29, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>> >>>>>>>>>>> So the weirdest thing is Kamailio actually sends the request, but it returns curl error 6 in the answer variable. I have a plenty of clonned servers and same behaviour is on all of them, also just now tried on a fresh debian 9 - same result. I'm inside a virtual machine (vmware), but not sure it matters... >>>>>>>>>>> >>>>>>>>>>> ________________________________ >>>>>>>>>>> Alexandru Covalschi >>>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>>> >>>>>>>>>>>> 20 нояб. 2017 г., в 22:37, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>>> >>>>>>>>>>>> I often get WARNING: http_client [functions.c:281]: curL_query_url(): Couldn't resolve host, while I don't even use hostnames. >>>>>>>>>>>> After I changed IP address to hostname (which I've added into /etc/hosts) nothing changed. It seems like a weird bug, but I've no idea - is it CURL or kamailio, but when I execute curl from shell it is working fine. I have google dns in resolve.conf and nslookup is also working ok. >>>>>>>>>>>> Can anyone help me with that? >>>>>>>>>>>> >>>>>>>>>>>> ________________________________ >>>>>>>>>>>> Alexandru Covalschi >>>>>>>>>>>> VoIP Engineer and System Administrator >>>>>>>>>>>> tel: +373 673 98 493 >>>>>>>>>>>> >>>>>>>>>>>>> 20 нояб. 2017 г., в 22:06, Alexandru Covalschi <568691(a)gmail.com <mailto:568691@gmail.com>> написал(а): >>>>>>>>>>>>> >>>>>>>>>>>>> t >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Daniel-Constantin Mierla >>>>>>>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>>>>>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>>>>>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla >>>>>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>>>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>>>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla >>>> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >>>> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >>>> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/> >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >> Kamailio Advanced Training - www.asipto.com <http://www.asipto.com/> >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com <http://www.kamailioworld.com/>

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sr-users December 2017