- sr-users - kamailio.org

Recommendation
by Carsten Bock 05 Aug '10

05 Aug '10

Hi everybody, a former colleague and friend of mine is searching for a new job. He is currently working as NOC-Support-Engeneer and has good knowledge of the SIP-protocol and about running/supporting Kamailio (and supporting systems, e.g. databases). He would prefer to work in northern Germany; but that is not a must but more like a "nice-to-have". You can find his profile on Xing: https://www.xing.com/profile/Marcel_Weinberg3 I can provide further information/contact information if someone is interested. Kind regards, Carsten

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help with configure kamailio authenticate with freeradius server.
by thanh.truong＠htklabs.com 05 Aug '10

05 Aug '10

Hi all I try to configure kamailio to register sip client with freeradius via kamailio. My configure as: ###################kamailio.cfg############# #!KAMAILIO # # $Id$ # # Kamailio (OpenSER) SIP Server v3.0 - basic configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <users(a)lists.kamailio.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 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 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 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no port=5060 /* 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 ####### 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 module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/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" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_radius.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 # ----------------- setting module-specific parameters --------------- # ----- 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", 0) # ----- 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", "") # ----- 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://openser:openserrw@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://openser:openserrw@localhost/kamailio") #modparam("auth_db", "load_credentials", "") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("auth_radius", "auth_extra", "Acct-Session-Id=$ci") modparam("auth_radius", "use_ruri_flag", 22) #!endif # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #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://openser:openserrw@localhost/openser") 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") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ 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; } # NAT detection route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... 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() ) { # non 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.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # 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(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { 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(); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); 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 setflag(2); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag("6"); } if (isflagset(5) || isbflagset("6")) { 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_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence server route route[PRESENCE] { #!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 (!radius_www_authorize("192.168.1.70")) { www_challenge("192.168.1.70", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!radius_proxy_authorize("192.168.1.70")) { proxy_challenge("192.168.1.70", "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 } } #!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(5); } #!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; } # 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; } # 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(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[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { 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(); ##} } ############################################## I run command : kamctl start But : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed And I run : tail -f /var/log/message to show log: The log is: Aug 4 15:05:07 appliance kamailio: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/auth_radius.so>: libradiusclient-ng.so.2: cannot open shared object file: No such file or directory Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 12-27: failed to load module Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 234, column 92: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 235, column 43: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 236, column 60: Can't set module parameter Aug 4 15:05:07 appliance kamailio: ERROR: <core> [modparam.c:154]: set_mod_param_regex: No module matching <auth_radius> found Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 237, column 44: Can't set module parameter Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 479, column 50: unknown command, missing loadmodule? Aug 4 15:05:07 appliance kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 494, column 53: unknown command, missing loadmodule? ############### I try to add : /usr/local/lib/* to /etc/ld.so.conf And run : ldconfig -v But the result is the same error . Please help me if someone know what is wrong. Any suggest is welcome. Thanks and Regards Thanh.Truong

2 1

help with sip client register with freeradius via kamailio.
by thanh.truong＠htklabs.com 05 Aug '10

05 Aug '10

Hi all. Please help me with my problem. I trying to set up kamailio and freeradius. I want sip client authenticate with freeradius server, but for a long time I can make it work. I have referenced many reference document but it cannot work. My configure file as my attachment . Please suggest if you have ideal. Kamailio -3.0.1 freeradius-1.1.3-1.5.el5_4.src.rpm radiusclient-ng-0.5.6 please help me as soon as possible. Thanks and regards.

2 1

DBI Error on startup
by "Nicolas Rüger" 05 Aug '10

05 Aug '10

Hello, I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. I included the following line in a PERL-script (that I want to be started from the routing-logic in kamailio.cfg) for interaction of PERL with a MySQL Database: use DBI; The problem is that I get the follwing error at kamailio-startup: lenny:/opt/kamailio-3.0.0/sbin# 0(17385) ERROR: perl [openserxs.xs:1027]: perl error: Can't load '/usr/lib/perl5/auto/DBI/DBI.so' for module DBI: /usr/lib/perl5/auto/DBI/DBI.so: undefined symbol: PL_memory_wrap at /usr/lib/perl/5.10/DynaLoader.pm line 196. at /usr/lib/perl5/DBI.pm line 263 The PERL-script itself compiles without problems and other scripts work fine as well. What can I do??? Thanks for the help in advance. Regards, Nicolas -- GMX DSL: Internet-, Telefon- und Handy-Flat ab 19,99 EUR/mtl. Bis zu 150 EUR Startguthaben inklusive! http://portal.gmx.net/de/go/dsl

2 2

[OT] Problem with <sr-users@lists.sip-router.org> subscription
by Iñaki Baz Castillo 02 Aug '10

02 Aug '10

Hi, each N weeks I receive the following mail from sr-users-request(a)lists.sip-router.org: --------------------- Your membership in the mailing list sr-users has been disabled due to excessive bounces The last bounce received from you was dated 26-Jul-2010. You will not get any more messages from this list until you re-enable your membership. You will receive 4 more reminders like this before your membership in the list is deleted. -------------------- Not sure what could be the problem as I receive mail in ibc(a)aliax.net with no problem. Any tip please? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

INumWare: Inum Users Conference
by Meftah Tayeb 31 Jul '10

31 Jul '10

hello folk you're invited to join the inum ware conference this conference is for inum users that frequantly use inum (+883 country code) (http://www.inum.net) you can join it by dialing +883510001288888 from any provider that provide INum termination (see http://www.inum.net) thank you! -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

1 0

Loops using uac_replace_from
by Ján ONDREJ (SAL) 30 Jul '10

30 Jul '10

Hello, after some failures with uac_auth (uac_auth can't increase CSeq number, which is required by my provider) my provider maked me an testing access without autentification, but I still can't make calls with kamailio or openser. I can make calls with opensips, but I am still not chosen, which sip software to use in future (still using openser). This is a part of my configuration: route[6] { uac_replace_from("sip:556807505@as.xxxxxx.sk"); rewritehostport("122.33.44.200"); route(5); } When trying to sniff packets and diff them, I am still getting this from my kamailio server: U kamailio_ip:5060 -> client_ip:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL). From: "Ondrej Jan" <sip:xxxxxx@xxxx>;tag=59427D6F-81E63720. To: <sip:xxxxx@xxxxx;user=phone>;tag=cc70c0fe52d7af0accfb78a94d592620.f69c. CSeq: 1 INVITE. Call-ID: 37cadcfb-f93b8e1d-431cdf6@xxxxxxxxxxx. Server: kamailio (3.0.2 (i386/linux)). Content-Length: 0. Logs are below. Any idea, what happening and why a default opensips configuration works and kamailio/openser don't? SAL [root@kamailio ~]# kamailio -DddE loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ Listening on udp: 123.123.242.20:5060 Aliases: udp: kamailio.xxxx.sk:5060 *: 123.123.242.20:* WARNING: no fork mode 0(2364) INFO: <core> [tcp_main.c:4150]: init_tcp: using epoll_lt as the io watch method (auto detected) 0(2364) INFO: usrloc [hslot.c:53]: locks array size 512 0(2364) INFO: <core> [udp_server.c:166]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2364) INFO: <core> [udp_server.c:217]: INFO: udp_init: SO_RCVBUF is finally 262142 4(2368) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) ERROR: <core> [action.c:1269]: WARNING: too many recursive routing table lookups (101) giving up! 0(2364) BUG: tm [t_lookup.c:1556]: tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0xb5a15490: 17 (hex 11) ^C 4(2368) INFO: <core> [main.c:788]: INFO: signal 2 received 3(2367) INFO: <core> [main.c:788]: INFO: signal 2 received 1(2365) INFO: <core> [main.c:788]: INFO: signal 2 received 2(2366) INFO: <core> [main.c:788]: INFO: signal 2 received 0(2364) ALERT: <core> [main.c:719]: child process 2365 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2366 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2367 exited normally, status=0 0(2364) ALERT: <core> [main.c:719]: child process 2368 exited normally, status=0 0(2364) INFO: <core> [main.c:734]: INFO: dont_fork turned on, living on 0(2364) NOTICE: <core> [main.c:679]: Thank you for flying kamailio

3 2

[Kamailio-users] Weight based routing
by Chandrakant Solanki 30 Jul '10

30 Jul '10

Hello, I have setup 3 SIP server. It should be Kamailio, Asterisk or Any SIP Server. I have setup 3 server say Server A, Server B and Server C and each has weight like 50, 30, 20 in percentage. And I taken 10 calls and try to forward call based on weight. Is this possible in kamailio's module.. and if yes then which module will help me? Any Idea.. -- Regards, Chandrakant Solanki

4 6

Modifying replies in failure route
by Alex Balashov 29 Jul '10

29 Jul '10

I know failure routes operate relative to the request that failed, rather than the final reply received per se. Nevertheless, is there any way provided by Kamailio/SR at this time to modify a final reply (say, append a header, or something else theoretically permissible) prior to its being passed to the originating UAC? -- 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 4

Re: [SR-Users] urgent :geographic number problem
by Alex Balashov 29 Jul '10

29 Jul '10

These (very poorly formatted!) requests and replies all relate to OPTIONS pings and have nothing to do with calls, failed or otherwise. On 07/29/2010 07:48 AM, hala alramli wrote: > some thing strang ,befor every thing work fine but now the most of the > geografic dont receive calls from outbound (landline) but recive calls > from our system it seem that it deal with it as local number just,when i > snif the backet by using ngrep it give me this: > 2010/07/29 12:58:28.111091 77.93.***.***:5060 -> 173.33.***.***:60005 > OPTIONS sip:173.33.***.***:60005 SIP/2.0..Via: SIP/2.0/UDP > 77.93.***.***:5060 ;branch=0..From: > sip:pinger@nogafone.com;tag=b4a8a831..To: sip:173.33.***.***: > :60005..Call-ID: f8d76aa2-5f768331-3b621@173.33.***.***:..CSeq > <mailto:f8d76aa2-5f768331-3b621@77.93.249.153..CSeq>: 1 > OPTIONS..Content-Length: 0.... > U 2010/07/29 12:58:28.111114 77.93.***.***:5060 -> 173.33.***.***::60004 > OPTIONS sip:173.33.***.***::60004 SIP/2.0..Via: SIP/2.0/UDP > 77.93.249.153:5060;branch=0..From: > sip:pinger@nogafone.com;tag=c4a8a831..To: sip:173.33.***.***: > :60004..Call-ID: f8d76aa2-6f768331-3b621@77.93.***.***:..CSeq > <mailto:f8d76aa2-6f768331-3b621@77.93.249.153..CSeq>: 1 > OPTIONS..Content-Length: 0.... > U 2010/07/29 12:58:28.247854 173.33.***.***:60004 -> 77.93.249.153:5060 > SIP/2.0 404 Not Found..To: > sip:173.33.***.***:60004;tag=c8b17aeb9f7688a1i0..From: > sip:pinger@nogafone.com;tag=c4a8a831..Call-ID: f8d76aa2-6f768331-3b621@ > 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP > 77.93.***.***::5060;branch=0..Server: > Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... > U 2010/07/29 12:58:28.255958 173.33.***.***:60005 -> 77.93.249.153:5060 > SIP/2.0 404 Not Found..To: > sip:173.33.***.***:60005;tag=c8b17aeb9f7688a1i0..From: > sip:pinger@nogafone.com;tag=b4a8a831..Call-ID: f8d76aa2-5f768331-3b621@ > 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP > 77.93.***.***:5060;branch=0..Server: > Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... > what is strang is that the replay is 404 not found while > 173.33.***.***:60005 is an IP address for an online adapter i will be > thankfull for any hint > regards > hala > > --- On *Wed, 7/28/10, Alex Balashov /<abalashov(a)evaristesys.com>/* wrote: > > > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] urgent :geographic number problem > To: "hala alramli" <doreme202002(a)yahoo.com> > Cc: "SR-Users" <sr-users(a)lists.sip-router.org> > Date: Wednesday, July 28, 2010, 5:52 PM > > On 07/28/2010 10:48 AM, hala alramli wrote: > > > our geographic number cant receive calls from outbound (from normal > > landlines number) but it receive from our system numbers just > ?where is > > the problem,is it from our server or from the company that provide us > > with this geographic number? > > It is impossible to know the answer to this question without digging > deeper into the underlying cause of the superficial effect you are > describing. The best way is to take a packet capture and see if the > requests from the provider are actually arriving; then you will know > at least whether they are reaching the server, and from there can > see why they may not be getting handled properly at the application > level. > > If the number worked before, and spontaneously stopped working, and > you did not change any configuration, it is most likely the provider. > > -- 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/ > > -- 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

Fw: Re: urgent :geographic number problem
by hala alramli 29 Jul '10

29 Jul '10

--- On Thu, 7/29/10, hala alramli <doreme202002(a)yahoo.com> wrote: From: hala alramli <doreme202002(a)yahoo.com> Subject: Re: [SR-Users] urgent :geographic number problem To: "Alex Balashov" <abalashov(a)evaristesys.com> Date: Thursday, July 29, 2010, 2:48 PM some thing strang ,befor every thing work fine but now the most of the geografic dont receive calls from outbound (landline) but recive calls from our system it seem that it deal with it as local number just,when i snif the backet by using ngrep it give me this: 2010/07/29 12:58:28.111091 77.93.***.***:5060 -> 173.33.***.***:60005 OPTIONS sip:173.33.***.***:60005 SIP/2.0..Via: SIP/2.0/UDP 77.93.***.***:5060 ;branch=0..From: sip:pinger@nogafone.com;tag=b4a8a831..To: sip:173.33.***.***: :60005..Call-ID: f8d76aa2-5f768331-3b621@173.33.***.***:..CSeq: 1 OPTIONS..Content-Length: 0.... U 2010/07/29 12:58:28.111114 77.93.***.***:5060 -> 173.33.***.***::60004 OPTIONS sip:173.33.***.***::60004 SIP/2.0..Via: SIP/2.0/UDP 77.93.249.153:5060;branch=0..From: sip:pinger@nogafone.com;tag=c4a8a831..To: sip:173.33.***.***: :60004..Call-ID: f8d76aa2-6f768331-3b621@77.93.***.***:..CSeq: 1 OPTIONS..Content-Length: 0.... U 2010/07/29 12:58:28.247854 173.33.***.***:60004 -> 77.93.249.153:5060 SIP/2.0 404 Not Found..To: sip:173.33.***.***:60004;tag=c8b17aeb9f7688a1i0..From: sip:pinger@nogafone.com;tag=c4a8a831..Call-ID: f8d76aa2-6f768331-3b621@ 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP 77.93.***.***::5060;branch=0..Server: Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... U 2010/07/29 12:58:28.255958 173.33.***.***:60005 -> 77.93.249.153:5060 SIP/2.0 404 Not Found..To: sip:173.33.***.***:60005;tag=c8b17aeb9f7688a1i0..From: sip:pinger@nogafone.com;tag=b4a8a831..Call-ID: f8d76aa2-5f768331-3b621@ 77.93.***.***:..CSeq: 1 OPTIONS..Via: SIP/2.0/UDP 77.93.***.***:5060;branch=0..Server: Linksys/PAP2T-3.1.15(LS)..Content-Length: 0.... what is strang is that the replay is 404 not found while 173.33.***.***:60005 is an IP address for an online adapter i will be thankfull for any hint regards hala --- On Wed, 7/28/10, Alex Balashov <abalashov(a)evaristesys.com> wrote: From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] urgent :geographic number problem To: "hala alramli" <doreme202002(a)yahoo.com> Cc: "SR-Users" <sr-users(a)lists.sip-router.org> Date: Wednesday, July 28, 2010, 5:52 PM On 07/28/2010 10:48 AM, hala alramli wrote: > our geographic number cant receive calls from outbound (from normal > landlines number) but it receive from our system numbers just ?where is > the problem,is it from our server or from the company that provide us > with this geographic number? It is impossible to know the answer to this question without digging deeper into the underlying cause of the superficial effect you are describing. The best way is to take a packet capture and see if the requests from the provider are actually arriving; then you will know at least whether they are reaching the server, and from there can see why they may not be getting handled properly at the application level. If the number worked before, and spontaneously stopped working, and you did not change any configuration, it is most likely the provider. -- 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

userblacklist Module - Documentation fault?
by "Nicolas Rüger" 29 Jul '10

29 Jul '10

Hello, in the documentation of the Module "Userblacklist" seems not really clear to me. Please clarify my doubts. Refering to: http://www.kamailio.org/docs/modules/3.0.x/modules_k/userblacklist.html in "4.1. check_user_blacklist" is written: "[...]If a match is found and it is not set to whitelist, false is returned. Otherwise, true is returned.[...]" Question: 1.) Is "not whitelisted" equal to "blacklisted"? 2.) Why is "false" returned if the blacklist_check is positive? 3.) Is "true" even returned if there's no match? I'd have expected the following: if (user_is_blacklisted) then return TRUE; else return FALSE; Can you please clarify the behavior? Thank you. Regards, Nicolas Example that seems clear to me: =============================== In "4.2. check_user_whitelist" is written: "[...]If a match is found and it is set to whitelist, true is returned. Otherwise, false is returned.[...]" cause it means to me: if (user_is_whitelisted) then return TRUE; else return FALSE; -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 2

Meeting in Chicago, Cluecon 2010, next week
by Daniel-Constantin Mierla 29 Jul '10

29 Jul '10

Hello, I will give a presentation about latest Kamailio - SIP Router news and usage together with FreeSwitch Media Server at Cluecon 2010 (http://www.cluecon.com), next week in Chicago. If you are around and want to meet during the event or out of it, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

Kamailio Transformation Assignment to an AVP
by Brandon Armstead 29 Jul '10

29 Jul '10

Hello, I have the following transformation on the SDP Body: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}) However when I assign this to an AVP, i.e. $avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s}); I receive back a NULL result / transformation "regex does not match" is what I receive with high verbose syslog on kamailio. HOWEVER. When I do something like this: xlog("L_INFO", "Payloads Available: $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})"); It executes properly -- transformation regex matches and the payloads are displayed as expected. If anyone can provide any insight as to what I may be doing wrong it would be greatly appreciated. P.S. Kamailio SVN Revision # 2:5906M (1.5.2-notls). Also I CC'ed devel list (as I do believe this may be a bug). Thanks! Sincerely, Brandon Armstead

3 5

Re: [SR-Users] how to set up iptel.org sip outbound
by Daniel-Constantin Mierla 29 Jul '10

29 Jul '10

Hello, please keep the mailing list cc-ed, private emails related to mailing lists discussions will not be answered. On 7/29/10 1:46 AM, j w wrote: > Hi Daniel, > > Thanks for your reply. Regarding set up outbound from ATA. I have > check with the Linksys tech support. They checked and said the > setting that they can do is all done, I have check with the outbound > provider. > > Can your outbound service allow me to dial out with a landline phone > via ATA. Would you please give me some setting information for your > outbound. Then it is not my service, I pointed you to a mailing list that is more appropriate for this subject. What I can give you as hints, try to put the ata in a network hub and sniff the sip traffic (with ngrep, wireshark, tcdump, etc...) to see if there is some attempt to send registrations/calls to sip server. Also, check maybe the device has some troubleshooting tool embedded (i.e., dedicated web page for logs). Cheers, Daniel > > Much obliged. > > Joseph > > ------------------------------------------------------------------------ > Date: Wed, 28 Jul 2010 10:54:24 +0200 > From: miconda(a)gmail.com > To: w7269826(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] how to set up iptel.org sip outbound > > Hello, > > On 7/19/10 8:35 PM, j w wrote: > > Hi, > > I registered iptel.org for an account and set up my Linksys > SPA3102 ATA gateway. The inbound is working but outbound cannot > dial out. > > First, the mailing list for iptel.org service is: > http://lists.iptel.org/mailman/listinfo/services > > You should get better answers there. > > Can any one help me: > 1. how to set up the sip outbound in order for me to dial out via ATA. > > > This has to do with ATA configuration, not much related to the server > side. You should have a user manual for that device. > > 2. how to get Canada phone number. > > > I don't think iptel.org is offering this, if you need for incoming, > you probably can get it from somewhere else and forward it to you > iptel.org accoung. > > Thanks a lot. > > I am not sure how to retrieve the post/mailing list, would you > please forward me your help reply to my > e-mail:w7269826@hotmail.com <mailto:e-mail:w7269826@hotmail.com>. > > > This is a public mailing list based on subscriptions. You have to > subscribe via web at: > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > Enter your email address, you will get a confirmation email with > further instructions. Once you follow them, you will receive all > emails sent to mailing list. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

urgent :geographic number problem
by hala alramli 28 Jul '10

28 Jul '10

our geographic number cant receive calls from outbound (from normal landlines number) but it receive from our system numbers just ?where is the problem,is it from our server or from the company that provide us with this geographic number? please help me regards

2 1

urgent :geographic number problem
by hala alramli 28 Jul '10

28 Jul '10

our geographic number cant receive calls from outbound (from normal landlines number) but it receive from our system numbers just ?where is the problem,is it from our server or from the company that provide us with this geographic number? please help me regards

1 0

multiple instances on different ports
by Robert R 28 Jul '10

28 Jul '10

Hi, I am trying to run multiple different instances of openser on the same server using different ports. Regardless of receiving INVITE on any of the ports (5061, 5062, ...) I want openser always to proxy the message to port 5060. invite:5061 ---> openser --> invite:5060 invite:5062 ---> openser --> invite:5060 By adding two listen ports as follow, openser sends the traffic to the same outgoing port as the incoming port: listen=udp:209.16.123.160:5061 listen=udp:209.16.123.160:5062 Is there a way to hard code the outgoing port? or I should overwrite the port? Thanks, R

4 5

Missing fields in ACC
by David 28 Jul '10

28 Jul '10

Hello, I am doing db logging, and my ACCs are being added to the table, but it is missing the leg_src and leg_dst columns. I set the leg_src and leg_dst columns, I do this : modparam("acc", "multi_leg_info", "leg_src=$avp(s:caller);leg_dst=$avp(s:dialed)") In my routing script on the initial INVITE I do this : $avp(s:dialed) = $tu; $avp(s:caller) = $fu; xlog("L_WARNING", "Call Tracked - leg_dst=$avp(s:dialed) leg_src=$avp(s:caller)\n"); The xlog shows the leg_src and leg_dst, but they do not appear in the leg_dst and leg_src columns of my acc table. I have no warnings or errors ( other than "ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted" ), but that's not worrysome. I do not use t_newtran(), t_check_trans() is called before I set the values, but this does not create a transaction. After the above code is called, I call t_relay(); Any idea what might cause the ACC record to be added without the multi_leg_info ? BTW, the avpops table in the database is a ENGINE=BLACKHOLE. As far as I can tell, the ACC is the same as my Kamailio 1.5.3 setup, but it would seem in migrating to 3.0, something broke, any ideas ? Thanks, David

2 1

no signal in phone line ,server slow
by hala alramly 28 Jul '10

28 Jul '10

hello, we have a problem with our kamailio server sometimes it being very slow ,and i cant hear the signal in the phone line for a while then it fix him self ,when i return to syslog every thing normal, what is the reasons ? regards, hala _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969

2 1

how to set up iptel.org sip outbound
by j w 28 Jul '10

28 Jul '10

Hi, I registered iptel.org for an account and set up my Linksys SPA3102 ATA gateway. The inbound is working but outbound cannot dial out. Can any one help me: 1. how to set up the sip outbound in order for me to dial out via ATA. 2. how to get Canada phone number. Thanks a lot. I am not sure how to retrieve the post/mailing list, would you please forward me your help reply to my e-mail:w7269826@hotmail.com. Much obliged. Joseph _________________________________________________________________ MSN Dating: Find someone special. Start now. http://go.microsoft.com/?linkid=9734384

2 1

[Serusers] Error while running icscf.sh
by Abdul Khadar 28 Jul '10

28 Jul '10

Hi, Can you please help me in resolving this issue while setting OpenIMSCore setup. Error : [root@hss OpenIMSCore]# sh icscf.sh * 0(18935) ERROR: load_module: could not open module </opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so>: /opt/OpenIMSCore/ser_ims/modules/mysql/mysql.so: undefined symbol: log * 0(18935) parse error (30,13-14): failed to load module ERROR: bad config file (1 errors) 0(18935) INFO:cdp:cdp_exit(): CDiameterPeer stoping ... icscf.sh: line 5: 18935 Segmentation fault /opt/OpenIMSCore/ser_ims/ser -f /opt/OpenIMSCore/icscf.cfg -D -D please help me to resolve this problem. -- thanks & regards, Abdul Khadar

4 4

Kamailio 3.0.2 mysql create table domain definition
by Stagg Shelton 28 Jul '10

28 Jul '10

Hello, The script in the source scripts/mysql/my_create.sql uses the the following definition CREATE TABLE domain ( did VARCHAR(64) NOT NULL, domain VARCHAR(128) NOT NULL, flags INT UNSIGNED NOT NULL DEFAULT '0', UNIQUE KEY domain_idx (domain), KEY did_idx (did) ); The script in the source utils/kamctl/mysql/domain-create.sql uses the following definition CREATE TABLE domain ( id INT(10) UNSIGNED AUTO_INCREMENT PRIMARY KEY NOT NULL, domain VARCHAR(64) DEFAULT '' NOT NULL, last_modified DATETIME DEFAULT '1900-01-01 00:00:01' NOT NULL, CONSTRAINT domain_idx UNIQUE (domain) ) ENGINE=MyISAM; Which table definition is the correct definition to be used. Thanks Stagg Shelton

3 2

Doubs about using append_branch ...
by Raúl Alexis Betancor Santana 26 Jul '10

26 Jul '10

Hi all I'm trying to do a "followme" from the proxy side, so I receive a request to some SIP URI and that URI have configured it's followme function that in this case means ... progress a call to the original URI + a call to a new URI, just a parallel branching, but I'm no luck ... :-( if(is_avp_set($avp(s:followme)) { append_branch($avp(s:followme)); } That doesn't run, because append_branch does not accept a PV as param, so ... ... $var(s:tmp_ru) = $ru; append_branch(); $ru = $avp(s:followme); ... But now ... only one of the branches progress ..., the new created branch (after the append_branch call) ... must go to a PSTN number ... but no way, because I must "mark" the branch in some way that I could send it directly to the go_to_pstn route block ... any clue or example about how to work with the new branch ONLY ? ... because branch 1 works ok, I need to do changes only on branch 2 Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

3 16

Re: [SR-Users] Doubs about using append_branch ...
by Alex Balashov 25 Jul '10

25 Jul '10

It sounds like what you need is to do branch-specific logic in a branch route: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#branch_route Not all functions can be used in a branch route, but those that can be operate only on that branch. On 07/25/2010 07:07 PM, Raúl Alexis Betancor Santana wrote: > > Hi all > > I'm trying to do a "followme" from the proxy side, so I receive a request to > some SIP URI and that URI have configured it's followme function that in this > case means ... progress a call to the original URI + a call to a new URI, > just a parallel branching, but I'm no luck ... :-( > > if(is_avp_set($avp(s:followme)) > { > append_branch($avp(s:followme)); > } > > That doesn't run, because append_branch does not accept a PV as param, so ... > > ... > $var(s:tmp_ru) = $ru; > append_branch(); > $ru = $avp(s:followme); > ... > > But now ... only one of the branches progress ..., the new created branch > (after the append_branch call) ... must go to a PSTN number ... but no way, > because I must "mark" the branch in some way that I could send it directly to > the go_to_pstn route block ... > > any clue or example about how to work with the new branch ONLY ? ... because > branch 1 works ok, I need to do changes only on branch 2 > > Best regards -- 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

Kamailio 3.0 cann't access the RTPProxy
by JinKevin 24 Jul '10

24 Jul '10

Hi All, I have the kamailio configed as below on Solaris 10 (Sparc): # ----- 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") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif and rtpproxy is started as AppSer01:ipunity@/opt/ipunity$ ps -ef |grep rtp ipunity 4687 1 0 12:58:20 ? 0:00 /usr/local/bin/rtpproxy -l 210.xx.xx.xx -s udp:localhost:7722 AppSer01:ipunity@/opt/ipunity$ netstat -an |grep 7722 127.0.0.1.7722 Idle 127.0.0.1.41509 127.0.0.1.7722 Connected 127.0.0.1.41510 127.0.0.1.7722 Connected 127.0.0.1.41511 127.0.0.1.7722 Connected 127.0.0.1.41512 127.0.0.1.7722 Connected 127.0.0.1.41513 127.0.0.1.7722 Connected 127.0.0.1.41514 127.0.0.1.7722 Connected 127.0.0.1.41515 127.0.0.1.7722 Connected 127.0.0.1.41516 127.0.0.1.7722 Connected 127.0.0.1.41517 127.0.0.1.7722 Connected 127.0.0.1.41518 127.0.0.1.7722 Connected 127.0.0.1.41519 127.0.0.1.7722 Connected However, kamailio is not able to connect the rtpporxy, Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 788202 local0.error] ERROR: nathelper [nathelper.c:2457]: can't send command to a RTP proxy Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 844113 local0.error] ERROR: nathelper [nathelper.c:2492]: proxy <udp:127.0.0.1:7722> does not respond, disable it Jul 24 12:58:59 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22867]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:3144]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Jul 24 12:59:04 AppSer01 /usr/local/kamailio-3.0/sbin/kamailio[22870]: [ID 582370 local0.error] ERROR: nathelper [nathelper.c:2627]: no available proxies Any helps are appreciated. Thanks, Kevin _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

1 0

Regex Transformations
by Brandon Armstead 22 Jul '10

22 Jul '10

Hello, I am trying to match a multi-line psuedo variable, i.e. $rb However I am wishing to pull out the payload values i.e. "0 18 101" "18 101" etc.. etc... I am having trouble matching this. Any help would be appreciated. One part that is giving me trouble is that it seems xlog prints out \r\n as \015\012 I am not able to successfully just pull out the "payloads" I have tried many different variations, however here is one of my latest: xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9 ]+)\015\012(.*)$/\3/s})"); Any help / input is greatly appreciated, thank you ahead of time! Happy 4th of July (for those who celebrate) Sincerely, Brandon Armstead

2 9

Sniffer SIP
by Fernando Pessoa 22 Jul '10

22 Jul '10

Friend you know tell me if it is possible to make the packets sniffer sip with SER or SEMS? thanks, -- Atenciosamente, Fernando Pessoa ITIntegration Diretor 85-34865449 85-99222598

3 2

Outbound calls
by Alan 22 Jul '10

22 Jul '10

I'm trying to set up my account so outbound calls will go through a proxy: free.sipout.com:161 I've set it up under OTHER in my iptel.org account so I access pbxes.com and put free.sipout.com:161 into a trunk and set it as an outbound route. I then dial +001 area code and number but it doesn't work. What am I doing wrong? Thanks, Alan

2 1

Perl Code Execution via kamailio.cfg fails
by "Nicolas Rüger" 22 Jul '10

22 Jul '10

Hello, I try to execute some PERL-code within kamailio v3 for some research due to my thesis. I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. I added the follwoing lines to "kamailio.cfg": loadmodule "perl.so" modparam("perl", "filename", "/opt/kamailio-3.0.0/PERL/myperl.pl") modparam("perl", "modpath", "/opt/kamailio-3.0.0/lib/kamailio/perl") I added a subroutine called "myprint" to the file "myperl.pl" for testing. I tried to exec the subroutine by adding the following line to "kamailio.cfg" just before "Routing Logic" (...even tried within "route{..}") as described in the perl_module description : perl_exec_simple("myperl"); Unfortunately it seems the code isn't executed, as it should write some lines in a log-file. What can I do? Thank you!! Cheers Nicolas additional information: ======================= Kamailio is starting without problems it just shows the follwing warning: "0(28679) WARNING: <core> [sr_module.c:445]: /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will notbe supported in newer versions; consider using mod_register() instead 0(28679) DEBUG: <core> [mem/f_malloc.c:265]: fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048" -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

3 13

Re: [SR-Users] PDT ran out of pkg memory using pdt_list [SOLVED}
by JR Richardson 21 Jul '10

21 Jul '10

>> try to play with mysql module timeout parameter in case the records are not >> loaded fast enough: >> >> http://kamailio.org/docs/modules/stable/modules/db_mysql.html#id2928400 >> >> Regards, >> Ramona > > Ok, I figured out that sr was taking about 9-10 seconds to load 300K > records so I increased: > # ----- db_mysql params ----- > modparam("db_mysql", "timeout_interval", 20) > > And found that the table would load, so I started increasing the > record count by 60K, then the fault condition occurred again at around > 400K records, then went away by itself. I got all the way to 760K > records without a problem then hit another wall at 800K records. > > I think there is something flaky going on, I can not reproduce it > reliable, sometimes the table loads just fine and other times it does > not load at all. > > I started monitoring mysqld. When I restart sr or issue pdt_reload > and msqld has a load, the table seems to load fine, regardless of the > table size or "timeout_interval" setting. When the problem is > occurring, myslqd has zero usage. When the issue was occurring, I > restarted the mysql process and the table loaded into sr. I'm now > suspecting I have a buggy mysqld so I'm going to try to update that > and see if I can reproduce the problem. So for now I can't isolate > the issue, but I'm still working on it. > > I do see a difference with the "timeout_interval" during sr restarts, > the longer the time the better it does. Also there is a time > difference when loading the table, sometimes it takes 30 seconds, > sometimes 90 seconds. I think maybe there is a combination of issues > going on here. I need to rebuild a fresh install and test a bit more > methodically to find a root cause. Also I want to grab a newer > version of mysql-server. > > I'll update with my findings. # ----- db_mysql params ----- modparam("db_mysql", "timeout_interval", 20) Time interval (in seconds) after that an connection attempt, read or write request is aborted. The value counts three times, as several retries are done from the driver before it gives up. "The values counts 3 times" this was my issue, as I was adjusting the value between 10, 20, 30, 60, 120, and having inconsistent table load times from mysqld. After a reboot, mysql started performing better, more consistent with load times. My table with 800K records was taking between 30-38 seconds to load. With timeout_interval set to 10 (30 second timeout), I was right on the edge of the table loading or not loading, sometimes it would and other times it would not. So giving a bit of cushion on the timeout_interval seems to resolve the problem. Adjusting this value I could consistently reproduce the problem. So for large database tables, this timeout value should be set to about 1 second per 20K records. As far as the other issue goes with using pdt_list and pkg memory size, compileing mi using system mem solved this issue as well. Also pdt_list has the option to search for prefixes which works in my case without the need to increase pkg memory or system memory: sip-router2:~# kamctl fifo pdt_list . 99999 PDT:: SDOMAIN=* PREFIX=999990 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999991 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999992 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999993 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999994 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999995 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999996 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999997 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999998 DOMAIN=2 PDT:: SDOMAIN=* PREFIX=999999 DOMAIN=2 Using the list prefix search option will pipe out about 10,000 records, but errors out with "no more pkg mem" past that. Thanks for all your input on this thread to help me isolate the issue. JR -- JR Richardson Engineering for the Masses

1 0

freezing for major release 3.1
by Daniel-Constantin Mierla 21 Jul '10

21 Jul '10

Hello, as discussed during our devel meeting, next major release 3.1 is scheduled for this autumn. Therefore the freezing should happen soon. Would be 2 weeks from now enough for everyone to push planned new features to repo? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

4 9

Dialplan module examples
by Raúl Alexis Betancor Santana 20 Jul '10

20 Jul '10

Anyone known about examples for the dialplan table ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

2 2

kamailio server load balancing
by RAJNIKANT VANZA 20 Jul '10

20 Jul '10

Hi Guys, I have 2 kamailio servers. I need to load balancing for this 2 kamailio servers. Scenario: ------------ Each kamailio server call limit is 5000 approx. kamailio server 1 kamailio server 2 If kamailio server 1 reach call limit (5000) then call go on kamailio server 2. What is the best way for achieve this? Thanks in advance. -- Best Regards, Rajnikant Vanza

2 1

our kamailio server very slow , and there is no signal in the line
by hala alramli 17 Jul '10

17 Jul '10

Hello, some time ,there is no signal in phone line for a while ,also the server become very slow then it fix him self ,i don't know how . ,could any body give me possible reasons ,and how i can fix that, i checked syslog file every thing normal as i can see ,please help me. regards, hala

2 1

Simple Least Cost Routing NPA-NXX?
by JR Richardson 16 Jul '10

16 Jul '10

Hi All, I'm in the lab with sip router 3.0 mocking up some trunk-group prefix routing with PDT and load balancing with Dispatcher. This is working great and as expected. I would like to incorporate some sort of LCR matching on NPA-NXX but I'm having some difficulty. The PDT module automatically strips the prefix so for trunk group routing it works great but really can't be used for matching NPA-NXX. The Carrierroute module is pretty nice and works as expected but the issue I'm having, it appears that I have to have a NPA-NXX entry for each domain gateway I want to send calls to. For instance if I have 3 gateways in domain 1, then I would need the same NPA-NXX record entry in the database for each gateway. The database entries would be [NPA-NXX] x [Number of Gateways] per domain. So having 100K routes would mean having 300K records. Am I looking at this correctly? This is really what it seems like to me but logically I would think this should not be the case? I really like the Dispatcher polling functions to actively know what gateways are on-line and off-line plus the different distribution algorithms to choose from. So it makes sense to me to implement a pseudo PDT/Distpacher scenario as an LCR but with PDT stripping the prefix, that can not be done. Is there another mechanism whereby I can dip the database looking for the NPA-NXX, extract the domain group and send that to the Dispatcher function, without stripping the prefix? Or can a PDT module parameter be added to not strip the prefix. But I'm not sure this will work, can the PDT module hold in memory a few hundred thousand records? Thanks. JR -- JR Richardson Engineering for the Masses

2 3

Load Balancing REDIRECT
by Maw Zin Tun - Real Telekom 16 Jul '10

16 Jul '10

Hi, I started using Kamailio 3.0 as redirect server about 2 months ago, basically I'm quite new to it. Now I need to change current setup to redundant active-active system but I'm struggling where to find documentation that can lead me to achiving it. Can anyone pls advise me where to start looking ? Many thanks, Maw Tun

4 5

No error line printed ?
by Raúl Alexis Betancor Santana 15 Jul '10

15 Jul '10

Hi, I'm trying to test the new 3.0.2 release ... but with no luck ... sbc02:/etc/kamailio# kamailio -ddddddddd -u kamailio -g kamailio -c -E 0(1036) DEBUG: <core> [main.c:1800]: read 343643572 from /dev/urandom 0(1036) DEBUG: <core> [main.c:1806]: seeding PRNG with 1622837431 0(1036) DEBUG: <core> [main.c:1810]: test random numbers 396346077 2140170049 1800197589 0(1036) : <core> [cfg.lex:1390]: included file name must be between quotes If i'm not wrong ... I should get the line/caracter at witch the parser get the error ... but that's not true ... I'm missing something really stupid ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

2 12

Dispatcher: ds_select_domain(set, alg) with a pseudo variable.
by Geoffrey Mina 15 Jul '10

15 Jul '10

How difficult would it be to make ds_select_domain(set, alg) accept a pseudo variable for the "set" argument? I would like to map the $rU to a specific dispatcher set via a database table. I suppose i could just hard code three or four sets with a conditional if() statement, but I would prefer to make it a little more dyanimc. Correct me if I am wrong in my assumption that ds_select_domain() doesn't accept pseudo variables. Thanks.

3 4

CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7f1106077a60
by Iñaki Baz Castillo 15 Jul '10

15 Jul '10

Hi, what does it mean this error message?: CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7f1106077a60 Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 14 Jul '10

14 Jul '10

Hi Daniel, I have followed your instruction (I'm using kamailio 1.5) and before the t_relay("0x05") statement I've setted: $du="sip:" + $rd + ";transport=tls"; but I get the following mesasge: Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790] Request leaving server, *D_-URI='sip:garr.it;transport=tls'_* - M=INVITE RURI=sip:xxx@garr.it F=sip:yyyyy@sip.mydomqin.it:5066 T=sip:123456789@sip.mydomain.it:5066 Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: CRITICAL:core:mk_proxy: could not resolve hostname: "garr.it" Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:uri2proxy: bad host name in URI <sip:garr.it;transport=tls> Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:t_forward_nonack: failure to add branches Therefore I would like to use information about TLS connection (tls port?) got from NAPTR record. Doesn't it possible with kamailio 1.5 ? Many thanks, Daniel Il 08/07/2010 18.12, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:10:35 +0200 > Mittente: Daniel-Constantin Mierla <miconda(a)gmail.com> > A: matteo.campana(a)klarya.it > CC: sr-users(a)lists.sip-router.org > > > > Hello, > > On 7/8/10 5:59 PM, Matteo Campana wrote: > > > > Hi all, > > I'm using kamailio 1.5 with TLS module. > > I need to make ENUM query and get NAPTR record. > > > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > > > > How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? > > > > I've try to set : > > > > dns_tls_pref=1 > > dns_udp_pref=2 > > dns_tcp_pref=3 > > > > in the general section of kamailio.cfg, but I get a parse error. > > > > > these parameters were introduced in kamailio with version 3.0. > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > better implementation. That will make the life easier to migrate to > upcoming 3.1 that will bring asynchronous TLS. > > No matter what you have in R-URI, you can force TLS via setting outbound > proxy address to be a TLS uri: > > $du = "sip:__ip_or_host__;transport=tls"; > t_relay(); > > The IP or host you can take from R-URI without any problem via PV $rd. > Other option is to use function from tm - t_relay_to_tls(): > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > >

2 1

dialog module: in-early-dialog requests / race conditions
by Timo Reimann 13 Jul '10

13 Jul '10

Hello, I have two issues related to dialog module state keeping/logging that caught my attention today: 1.) The module considers requests routed in early (but not yet confirmed) dialogs to be bogus, as can be seen from a few lines of state machine code in dlg_hash.c: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules_… In the logs, it shows up something like this: CRITICAL:dialog:log_next_state_dlg: bogus event 8 in state 2 for dlg 0x7f9f237ffff8 [2901:1286709577] with clid '<Call-ID>' and tags '<caller tag>' '<callee tag, always empty>' I believe that such in-early-dialog requests should not trigger bogus-event log messages as they look fine from a standard's point of view. 2.) Various race conditions may generate log messages such as following when DID mode is enabled: WARNING:dialog:dlg_onroute: unable to find dialog for <REQUEST TYPE> with route param 'faf.f69e7b4' [4015:79161711] Observed REQUEST TYPEs at our site were "BYE" when BYE requests were transmitted by both caller and callee at roughly the same time due to user behavior; and "PRACK" when a PRACK was received by the UAS after a 486 (Busy Here)/ACK pair of messages was already exchanged between the UAS and the proxy. In both cases, the associated dialog had already been destroyed when another request was received. My suggestion is to lower the log level for this kind of message because it can simply happen too often but doesn't affect proper dialog handling. Feedback? Cheers, --Timo

3 8

Several BYEs conveying Kamailio
by Maciej Bylica 11 Jul '10

11 Jul '10

Dear ALL, I have configured Kamailio 1.5 from svn with radius (radiusclient-ng) support for acc. All seem to work okay, except one thing. >From time to time I am receiving following radius error (debug from radius server on different ip) Fri Jul 9 12:22:58 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:08 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:18 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null A few lines from radiusclient.conf # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 After closer look at this, it turned out that there is one START request and four STOPs with the same CallID (Acct-Session-Id). START and first STOP are treated as a pair, the next ones are odd. Taking about sip i do have following scenario: MGW(1)---Kamailio(2)---SIP_Provider(SER based)(3) The call flow that generate aforemtioned errors looks like below (debug on kamailio) 1 - (2) is receiving INVITE from (3) 2 - (2) is sending Giving a Try to (3) 3 - (2) is sending INVITE to (1) 4 - (2) is receiving Trying from (1) 5 - (2) is receiving Ringing from (1) 6 - (2) is receiving OK from (1) 7 - (2) is sending OK to (3) 8 - (2) is receiving ACK from (3) 9 - (2) is sending ACK to (1) ---- call---- 10 - (2) is receiving BYE from (1) (tearing down from (1) side, STOP radius request is generated) 11 - (2) is sending BYE to (3) 12 - (2) is receiving BYE from (3) (second STOP is generated, radius_retries 3, radius_timeout 10) 13 - (2) is sending BYE to (1) 14 - (2) is receiving OK from (1) 15 - (2) is sending OK to (3) 16 - (2) is receiving OK to (3) 17 - (2) is sending OK to (1) For sure there is sth wrong with SIP_Provider but how Kamilio should behave is such situation. Should second BYE (p.12) with the same callid as in p.10 be continued with p.13? Thanks in advance, Maciej.

2 3

Use AVP as regex in avp_subst
by Uriel Rozenbaum 10 Jul '10

10 Jul '10

Hey guys, I'm using Kamailio 1.5.3-notls and need to apply some re operation to an AVP. The problem I have is that my re expression is held in an AVP, so the parser is not recognizing it :( Would it be very hard to change that behavior? Any ideas/tips to implement? This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", "$avp(s:carrierAniRegex)"); ... } returns this error ERROR:avpops:fixup_subst: avpops: bad subst re $avp(s:carrierAniRegex) This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", $avp(s:carrierAniRegex)); ... } returns this error CRITICAL:core:yyerror: parse error in config file, line 751, column 51-74: syntax error CRITICAL:core:yyerror: parse error in config file, line 751, column 74-75: bad arguments Do you think some other way I can achieve this? Be aware I'm using SQLops to get realtime the value of the expression... Thanks in advance! Uriel

3 6

Kamailio, spiral with TM
by David 09 Jul '10

09 Jul '10

Hey, I have setup two Kamailio servers, my INVITEs will go through server A, which forwards to server B which forwards it back to server A. This is intential, the trouble is when the packet comes back through server B, nothing has changed so TM sees it as a retransmission. The difference between the first time through and the second time is the port 5061 instead of 5060. How can I tell tm that these are not the same transactions ? Would it be easier to have a seperate instance of Kamailio for 5061 ? Thanks, David

4 7

How to force snd_port?
by Victor Pascual Avila 09 Jul '10

09 Jul '10

Hi Folks, How could I force my sip-router to send from a given port number? Many thanks in advance, -- Victor Pascual Ávila

4 8

Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi all, I'm using kamailio 1.5 with TLS module. I need to make ENUM query and get NAPTR record. >From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? I've try to set : dns_tls_pref=1 dns_udp_pref=2 dns_tcp_pref=3 in the general section of kamailio.cfg, but I get a parse error. Regards, Daniel

4 4

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi klaus, Suppose I can't access to NAPTR settings. I need to manage SIP URI, so , If I right understand, the only way to use TLS protocol in kamailio 1.5 is to append ";transport=tls" in R-URI before relay. In other words I need to rewrite R-URI: $ru = $ru + ";transport=tls" ; # and the t_relay t_relay() ; something like that? Regards, Daniel Il 08/07/2010 18.45, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:44:27 +0200 > Mittente: Klaus Darilion <klaus.mailinglists(a)pernau.at> > A: Daniel-Constantin Mierla <miconda(a)gmail.com> > CC: matteo.campana(a)klarya.it, sr-users(a)lists.sip-router.org > > > > Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla: > > Hello, > > > > On 7/8/10 5:59 PM, Matteo Campana wrote: > >> > >> Hi all, > >> I'm using kamailio 1.5 with TLS module. > >> I need to make ENUM query and get NAPTR record. > >> > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > >> > >> How can I tell Kamailio to use TLS protocol ( instead of udp) after > >> NAPTR lookup ? > >> > >> I've try to set : > >> > >> dns_tls_pref=1 > >> dns_udp_pref=2 > >> dns_tcp_pref=3 > >> > >> in the general section of kamailio.cfg, but I get a parse error. > >> > > these parameters were introduced in kamailio with version 3.0. > > > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > > better implementation. That will make the life easier to migrate to > > upcoming 3.1 that will bring asynchronous TLS. > > > > No matter what you have in R-URI, you can force TLS via setting outbound > > proxy address to be a TLS uri: > > > > $du = "sip:__ip_or_host__;transport=tls"; > > t_relay(); > > IIRC we do have NAPTR support in Kamailio 1.5 - don't we? > > Then I think it should work when putting a domain into $du and makeing > sure that there is no transport parameter, no port, and NAPTR TLS record > has highest priority. > > regards > klaus > > > > > The IP or host you can take from R-URI without any problem via PV $rd. > > Other option is to use function from tm - t_relay_to_tls(): > > > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > > > Cheers, > > Daniel > > > >

2 1

Differentiation between ACK INVITE and ACK CANCEL
by inge 08 Jul '10

08 Jul '10

Hello all, I need to work on particular ACK within the configuration of SER 0.9.9. Firstly, I was thinking to make a difference using the CSeq field, but it seems that the CSeq on ACK contains the ACK himself. Probably because ACK is a request and not a response. Would it be possible to proceed in another way? Regards, Adrien .L

4 9

Kamailio integration - Registration issue
by Lucas Alvarez 08 Jul '10

08 Jul '10

Hi, I new with kamailio, I've been able to integrate kamailio 3.02 with asterisk 1.6. The only issue I'm having is if I have to restart asterisk( for some config update) I loose all the sip registration in asterisk, is there any way of fixing this?

9 13

Issue building sip-router (kamailio flavour)
by Santiago Gimeno 08 Jul '10

08 Jul '10

Hi, When I build sip-router (from master) with kamailio flavour using: make config mode=debug FLAVOUR=kamailio; make all; make install I'm getting this error when trying to start kamailio: ERROR: <core> [sr_module.c:396]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/tmx.so>: /usr/local/lib/kamailio/modules_k/tmx.so: undefined symbol: _tm_table On the other hand, if I use: make FLAVOUR=kamailio cfg; make all; make install kamailio starts correctly. I've tried also: make mode=debug FLAVOUR=kamailio cfg with no luck either. What's the problem? How should I compile kamailio with debugging so it starts correctly? Thank you in advance. Regards, Santi

3 5

Re: [SR-Users] no Contact in REFER generated with dlg_bridge
by Anton Roman 08 Jul '10

08 Jul '10

Hi, regarding the missing Contact header in the REFER message, it can be solved by including some lines in /modules_k/dialog/dlg_ transfer.c. In this file, the Contact of the initial INVITE generated with dlg_bridge command can be easily changed as well. The problem now is only in the processing of the ACK generated by the TM module. When this ACK leaves the Kamailio server the branch value of the topmost Via header is '0'. Below you can find the debug output where the ACK is generated and routed by Kamailio. Any idea about where the error can be? thanks in advance, regards Anton 11(5056) DEBUG: tm [t_msgbuilder.c:791]: building ACK for out-of-dialog INVITE (using RS in RR set). 11(5056) DEBUG: tm [t_msgbuilder.c:967]: ACK RURI: `sip:235@10.1.3.15:5061', NH: `sip:10.1.2.122;lr=on'. 11(5056) DEBUG: tm [t_reply.c:1059]: ->>>>>>>>> T_code=180, new_code=200 11(5056) DEBUG: tm [t_reply.c:1794]: DEBUG: local_reply: branch=0, save=0, winner=0 11(5056) DEBUG: tm [t_reply.c:356]: DEBUG: update_totag_set: new totag 11(5056) DEBUG: tm [t_reply.c:1831]: DEBUG: local transaction completed 11(5056) DEBUG: tm [t_hooks.c:288]: DBG: trans=0x7f9767eb1268, callback type 256, id 0 entered 11(5056) DEBUG: dialog [dlg_transfer.c:207]: completed with status 200 11(5056) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/msg_parser.c:612]: SIP Request: 9(5052) DEBUG: <core> [parser/msg_parser.c:614]: method: <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:616]: uri: < sip:235@10.1.3.15:5061> 9(5052) DEBUG: <core> [parser/msg_parser.c:618]: version: <SIP/2.0> 9(5052) DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bKf7bc.00856364.0>; state=16 9(5052) DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 9(5052) DEBUG: <core> [parser/msg_parser.c:500]: parse_headers: Via found, flags=2 9(5052) DEBUG: <core> [parser/msg_parser.c:502]: parse_headers: this is the first via 9(5052) DEBUG: <core> [receive.c:137]: After parse_msg... 9(5052) DEBUG: <core> [receive.c:177]: preparing to run routing scripts... 9(5052) DEBUG: sl [sl_funcs.c:335]: to late to be a local ACK! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=94cf927f9c6eed24i1 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: <core> [parser/msg_parser.c:174]: DEBUG: get_hdr_field: <To> [43]; uri=[sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:176]: DEBUG: to body [ sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>] 9(5052) DEBUG: <core> [parser/msg_parser.c:154]: get_hdr_field: cseq <CSeq>: <10> <ACK> 9(5052) DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_body : content_length=0 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 9(5052) DEBUG: <core> [parser/parse_to.c:179]: DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd 9(5052) DEBUG: <core> [parser/parse_to.c:808]: end of header reached, state=29 9(5052) DEBUG: sanity [mod_sanity.c:220]: all sanity checks passed 9(5052) ERROR: <script>: kamailio!!!: ACK 9(5052) DEBUG: siputils [checks.c:68]: totag found 9(5052) DEBUG: rr [loose.c:85]: is_preloaded: No 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==9 && [10.1.3.15] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 9==10 && [10.1.3.15] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5061 9(5052) DEBUG: <core> [forward.c:356]: check_self: host != me 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.1] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==9 && [10.1.2.122] == [127.0.0.2] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: <core> [socket_info.c:485]: grep_sock_info - checking if host==us: 10==10 && [10.1.2.122] == [10.1.2.122] 9(5052) DEBUG: <core> [socket_info.c:488]: grep_sock_info - checking if port 5060 matches port 5060 9(5052) DEBUG: rr [loose.c:781]: Topmost route URI: 'sip:10.1.2.122;lr=on' is me 9(5052) DEBUG: <core> [parser/msg_parser.c:90]: found end of header 9(5052) DEBUG: rr [loose.c:257]: No next Route HF found 9(5052) DEBUG: rr [loose.c:800]: No next URI found 9(5052) DEBUG: rr [rr_cb.c:97]: callback id 0 entered with <lr=on> 9(5052) DEBUG: dialog [dlg_handlers.c:797]: Route param 'did' not found 9(5052) DEBUG: tm [t_lookup.c:1387]: DEBUG: t_newtran: msg id=10 , global msg id=9 , T on entrance=0xffffffffffffffff 9(5052) DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=52095, isACK=1 9(5052) DEBUG: tm [t_lookup.c:720]: DEBUG: t_lookup_request: e2e proxy ACK found 9(5052) DEBUG: tm [t_reply.c:276]: DEBUG: totag for e2e ACK found: 0 9(5052) DEBUG: tm [t_funcs.c:315]: SER: forwarding ACK statelessly 9(5052) DEBUG: <core> [msg_translator.c:200]: check_via_address(10.1.2.122, 10.1.2.122, 0) 9(5052) DEBUG: <core> [forward.c:509]: Sending: ACK sip:235@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.2.122;branch=0 Via: SIP/2.0/UDP 10.1.2.122;branch=z9hG4bKf7bc.00856364.0 From: sip:click2dial@172.22.5.17 <sip%3Aclick2dial(a)172.22.5.17> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-3bcd Call-ID: 2bef8d5474ff8cd4 To: sip:235@10.1.2.122 <sip%3A235(a)10.1.2.122>;tag=94cf927f9c6eed24i1 CSeq: 10 ACK User-Agent: kamailio (3.0.1 (x86_64/linux)) Content-Length: 0

2 2

LCR, problem with routing (prefix, from_uri, grp_id)
by Maciej Bylica 07 Jul '10

07 Jul '10

Hi ALL, I have a problem to configure LCR module to work properly (kamailio rel. 1.5) The task in my case is to handle: 1) All calls coming with CLD number prefix 482223344 from my MGW (77.77.77.77) towards kamilio and direct them to sip world (88.88.88.88) 2) Calls with CLD starting with 48 coming from SIP world towards kamailio and direct them to my MGWs. Additionaly i need to route it on the basis of CLI number. So MGWs <---> Kamailio <---> SIPProxies (88.88.88.88) In db: +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 48 | 1 | 1 | | 2 | 48 | 49 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ gw, +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | id | gw_name | grp_id | ip_addr | hostname | port | uri_scheme | transport | strip | tag | weight | ping | flags | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ | 1 | CLI48 | 1 | 77.77.77.70 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 2 | CLI48backup | 1 | 77.77.77.71 | NULL | 5060 | NULL | NULL | NULL | NULL | 150 | 0 | 0 | | 3 | CLI49 | 2 | 77.77.77.75 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 4 | CLI49backup | 2 | 77.77.77.76 | NULL | 5060 | NULL | NULL | NULL | NULL | 15 | 0 | 0 | | 5 | SIPWORLD | 3 | 88.88.88.88 | NULL | 5060 | NULL | NULL | NULL | NULL | 250 | 0 | 0 | +----+--------------+--------+----------------+----------+------+------------+-----------+-------+------+--------+------+-------+ Unfortunately: - call is originated on 77.77.77.77 with CLD 48222334455 number and kamailio forward this call to grp_id=3 SIPWORLD which is okay. The problem is that if the call fail, kamailio will try to use 77.77.77.70 and 77.77.77.71 from grp_id=1 which is wrong. I have no idea how to provide a kind of huntstop in grp_id=3. I've been trying with +----+-----------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+-----------+----------------+--------+----------+ | 1 | 48 | 88.88.88.88 | 1 | 1 | | 2 | 48 | 88.88.88.88 | 2 | 1 | | 3 | 482223344 | 77.77.77.77 | 3 | 1 | +----+-----------+----------------+--------+----------+ but in this case i am unable to provide kamailio with CLI number routing. Here is a part of my config file... if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if (!next_gw()) { sl_send_reply("503", "Unable to find a gateway"); exit; } route(1); exit; and failover for route(1)... failure_route[11] { # In case of failure, send it to an alternative route: if (t_check_status("408|404|5[0-9][0-9]")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); exit; } else { t_on_failure("11"); t_relay(); } exit; } } Anybody could help my to get out of that? Thx, Maciej.

3 19

CANCEL before INVITE
by David 07 Jul '10

07 Jul '10

Hey, I am using a Cisco WIP310 wifi phone. Seeing as wifi is very battery demanding, the phone goes into a standby mode. When it's in the standby mode, it takes a few seconds to come back on. So I send an INVITE to the phone, statefully using TM, I send out a CANCEL before the phone returns the "180 ringing" message. Somehow the device is answering the CANCEL before the INVITE, so the result is that it responds to the CANCEL with "481 Call Leg/Transaction Does Not Exist.", after that it responds to the INVITE with a "180 Ringing", the phone than rings indefinitely because the CANCEL is not sent out again as the transaction is completed from the 481 SIP message. I had a look at the CANCEL, there is no Route: header or tag on the To, so it looks like it is part of a new dialog ( I believe that's what rfc3261 says. As far as I can tell, my Kamailio is working properly. It is retransmitting the messages at the proper intervals, and it is passing along the messages as it receives them. The trouble is that the device answers the initial CANCEL before it answers any of the retransmitted INVITEs. Is there something that I can do in Kamailio to resolve this issue ? Is there an option that I can set that will cause Kamailio to relay the CANCEL only to devices that have already returned a 100 Trying or 180 Trying ? What information do you need to know about my config? What parts of the SIP trace do you need ? Thanks, David

6 19

Summary of Devel Meeting, Berlin 2010
by Daniel-Constantin Mierla 06 Jul '10

06 Jul '10

Hello, fyi: summary of the devel meeting in Berlin, June 8, 2010, was uploaded to: http://sip-router.org/wiki/meetings/berlin-2010-minutes Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

2 1

TimeStamp header
by mustafa rifaee 06 Jul '10

06 Jul '10

Hello all; I need some Clarification about timestamp header in RFC 3261; i need to use it to estimate the RTT value; how it is work in this scenario?: Invite UA --------------------> Proxy1 | | Invite | | ------------------------------ --------->Proxy2 | 100 trying | | | <---------------------------| | invite | | | | | | --------------------------------------->UA2 | | <-------100 trying------------------- | | | | | <--------180 Ringing-------------- | | | <---------------180 Ringing------- | | | | | | |<---180 Ringing----------| | | UA send invite to proxy1 with timestamp=10 ; then the Proxy1 send the invite message to proxy2 and reply with provisional response (100 trying) and with timestamp=10 0.5; When UA receive 100trying; UA will adapt its retransmition timer T1 to the calculated value from timestamp; then UA adapt T2 Timer value; is This true? Here Can Proxy1 send invite message with timestamp header?; for example timestamp=8 and when proxy2 recive invite request, it send it to UA and he send 100 trying to proxy1 with timestamp=8 0.4; Please Help me; Thanks; Mustafa AL Rifaee

1 0

About setflag()'s parameter
by zhou tianjun 06 Jul '10

06 Jul '10

Hi all, I'm in doubt of the means of setflag()'s parameter. I have looked for it everywhere including internet and the ser source code, but no any docs explained it in detail. can anybody tell me about it ? -- With best regards by sunone

5 16

$rU rewrite problem
by Dmitri Korotkov 05 Jul '10

05 Jul '10

Hello, I have problems with rewriting $rU. Kamailio 3.0.1 on debian squeeze x86_64 In config file I have: $avp(s:dnumber) = $(ru{re.subst,/^sip:\*99\*(.*)@.*/\1/}); $rU=$avp(s:dnumber); And log file shows following warnings: Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core> [rvalue.c:987]: automatic string to int conversion for "sip:1234567890@voice.serv.net" failed Jul 5 17:22:30 voice /usr/sbin/kamailio[23641]: WARNING: <core> [rvalue.c:1839]: rval expression conversion to int failed (630,22-630,23) Could you please show any samples how to change $rU with regexp matched part of $rU ? BR, Dmitri

2 2

Cancel rtp proxy changes to SDP
by David 05 Jul '10

05 Jul '10

3 2

How SER verdict a session is over when missed BYEs
by zhou tianjun 05 Jul '10

05 Jul '10

Hi all, I'm using SER, SEMS and RTPPROXY, they all work fine. When caller or callee want to stop the dialog, one of them may send BYE to proxy, but the sip proxy doesn't receive it. so dows the other's BYE. In this case, how the proxy verdict the dialog is over and log it to mysql ? -- With best regards by zhoutj

2 1

About building ACD
by Trung Anh 04 Jul '10

04 Jul '10

Hi everyone, I have a problem about Agents status. Now, i want to develop ACD system but i don't know how to collect agent status. Please help me. Thanks in advance! -- Best Regards, Mr House Cricket

3 4

voicemail
by mattias jonsson 03 Jul '10

03 Jul '10

Have the server voicemail kameleo

2 1

Missing ACK
by Ole Kaas 03 Jul '10

03 Jul '10

Hello, See transmission below. For some reason the ACK package is lost/dropped and the remote server retransmits the 200 OK status. But Kamailio does not retransmit the ACK on receiving the retransmitted OK's. Eventually the remote server gives up and pulls the plug after ~20 seconds. Is this a bug or should I look closer at my routing script? The script is for the ACK handling pretty much the same at the vanilla script delivered with the source. I'm still on version 3.0.0. /Ole -> Request: INVITE sip:phonenum@destination, with session description <- Status: 100 Trying <- Status: 180 Ringing <- Status: 183 Session Progress, with session description <- Status: 200 OK, with session description -> Request: ACK sip:phonenum@destination <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description <- Status: 200 OK, with session description -> Request: BYE sip:phonenum@destination -> Request: BYE sip:phonenum@destination <- Status: 481 Call leg/transaction does not exist

6 17

Re: [SR-Users] change ptime / RTP packet size
by Carsten Bock 02 Jul '10

02 Jul '10

In order to change the attribute in the SDP, you need manually to alter the SDP (at least for 1.5.x). Tha NAT-Helper module will not do this for you. Please see the Textops-Module on how to do this. Carsten 2010/7/2 Laurent Schweizer <laurent.schweizer(a)peoplefone.com> > hello, > > > > I use kamailio 1.5.4 with rtpproxy 1.2.1 > > > > I try to change the ptime (RTP packet size) with the Nathelper module . > > > > to do it I call the function rtpproxy_offer with the option "z20" but the > "a=ptime:30" in SDP is not changed > > > > any idea ? > > > > Regards > > > > Laurent > > > > Log of RTPproxy: > > > > > > Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:handle_command: received command > "19072_4 UZ20c0,8,101 2ab4d3ad6ff9676a 217.162.26.26 3000 112a4c5d45;1" > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session > 2ab4d3ad6ff9676a, tag 112a4c5d45;1 requested, type strong > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session on a > port 54858 created, tag 112a4c5d45;1 > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: pre-filling > caller's address with 217.162.26.26:3000 > > Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: RTP packets from > caller will be resized to 20 milliseconds > > Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:doreply: sending reply "19072_4 > 54858 15.43.10.128 " > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:process_rtp: session timeout > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTP stats: 0 in > from callee, 0 in from caller, 0 relayed, 0 dropped > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTCP stats: 0 in > from callee, 0 in from caller, 0 relayed, 0 dropped > > Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: session on ports > 54858/0 is cleaned up > > _______________________________________________ > 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 > > -- Carsten Bock Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Home +49 40 34927217 Fax +49 40 34927218 mailto:carsten@bock.info

4 5

Re: [SR-Users] sr-users Digest, Vol 62, Issue 10
by marco＠bungalski.de 02 Jul '10

02 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

3 2

Re: [SR-Users] XCAP SERVER MODULE
by Daniel-Constantin Mierla 02 Jul '10

02 Jul '10

Hello, please address the questions to public mailing lists (e.g., sr-users(a)lists.sip-router.org), emails sent to private address only may not be answered at all or replied after long time. There are many people on ML that can answer faster and better. On 6/17/10 5:11 PM, michael ayoola wrote: > Hello Daniel, > > Thanks for your work on XCAP Server. I am quite new to kamailio. I > have succeeded in installing kamailio server and I am trying to follow > your documentation on XCAP server. However I do not know how to load > the db module. Terminology of 'db module' refers to a database connection module, like db_mysql, db_postgres, etc. Depending where you store the data, in a mysql, postgres or other type of sql server, you have to load the appropriate module. > I noticed that the sl module is already loaded as indicated in the > kamailio.cfg file (i.e loadmodule "sl.so"). There seem to many db > related module. Which do I use? Please, it will be nice if you can help. > Note that xcap_server module is not completed yet, should work with very basic xcap operations over http/1.0 or http/2.0 non-chunked body for PUT operations. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

permissions module address_reload not working
by JR Richardson 02 Jul '10

02 Jul '10

Hi All, I'm running kamailio 3.0.2. # ----- permissions params ----- modparam("permissions", "db_url", "mysql://openserro:openserro@localhost/openser") #modparam("permissions", "db_mode", 1) snip in route block: if (!allow_source_address_group()) { sl_send_reply("403", "Forbidden"); exit; } >From command line: sip-router2:/etc/kamailio# kamctl fifo address_reload 400 Address table reload failed debug output: during initial kamailio startup: 0(10655) DEBUG: permissions [address.c:122]: Number of rows in address table: 4 0(10655) DEBUG: permissions [address.c:164]: Tuple <1, 10.10.12.0, 24, 0> inserted into subnet table 0(10655) DEBUG: permissions [address.c:164]: Tuple <1, 10.10.14.0, 24, 0> inserted into subnet table 0(10655) DEBUG: permissions [address.c:148]: Tuple <1, 10.10.15.97, 32> inserted into address hash table 0(10655) DEBUG: permissions [address.c:148]: Tuple <1, 10.10.15.98, 32> inserted into address hash table 0(10655) DEBUG: <core> [db_res.c:81]: freeing 4 columns 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x827c158 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x827c220 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x827c168 0(10655) DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x827c230 0(10655) DEBUG: <core> [db_res.c:94]: freeing result names at 0x827c250 0(10655) DEBUG: <core> [db_res.c:99]: freeing result types at 0x827c1b8 0(10655) DEBUG: <core> [db_res.c:54]: freeing 4 rows 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c710 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c6b8 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c660 0(10655) DEBUG: <core> [db_row.c:97]: freeing row values at 0x827c598 0(10655) DEBUG: <core> [db_res.c:62]: freeing rows at 0x827ca10 0(10655) DEBUG: <core> [db_res.c:136]: freeing result set at 0x827c1f8 0(10655) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(10655) DEBUG: permissions [address.c:181]: address table reloaded successfully. result when issuing "kamctl fifo address_reload" 3(10389) DEBUG: mi_fifo [fifo_fnc.c:489]: entered consume 3(10389) DEBUG: mi_fifo [fifo_fnc.c:489]: **** done consume 3(10389) DEBUG: mi_fifo [fifo_fnc.c:503]: done parsing the mi tree 3(10389) ERROR: permissions [address.c:82]: db_handle already exists The address table loads fine during startup and if I reload kamailio it will load any new addresses or subnets so I know there is no syntax error in the database. I can reload trusted table and pdt table without an error but address table is not working. Any ideas on how I can trouble shoot this further? Thanks. JR -- JR Richardson Engineering for the Masses

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Re: [SR-Users] change ptime / RTP packet size
by Alex Balashov 02 Jul '10

02 Jul '10

On 07/02/2010 04:15 AM, Laurent Schweizer wrote: > hello, > > I use kamailio 1.5.4 with rtpproxy 1.2.1 > > I try to change the ptime (RTP packet size) with the Nathelper module . > > to do it I call the function rtpproxy_offer with the option "z20" but > the "a=ptime:30" in SDP is not changed Try use force_rtp_proxy() instead. Last time I tried to use it, rtpproxy_offer()/answer() were still broken, although admittedly that was a while ago. -- 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

Re: [SR-Users] sr-users Digest, Vol 62, Issue 9
by marco＠bungalski.de 02 Jul '10

02 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

change ptime / RTP packet size
by Laurent Schweizer 02 Jul '10

02 Jul '10

hello, I use kamailio 1.5.4 with rtpproxy 1.2.1 I try to change the ptime (RTP packet size) with the Nathelper module . to do it I call the function rtpproxy_offer with the option "z20" but the "a=ptime:30" in SDP is not changed any idea ? Regards Laurent Log of RTPproxy: Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:handle_command: received command "19072_4 UZ20c0,8,101 2ab4d3ad6ff9676a 217.162.26.26 3000 112a4c5d45;1" Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session 2ab4d3ad6ff9676a, tag 112a4c5d45;1 requested, type strong Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: new session on a port 54858 created, tag 112a4c5d45;1 Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: pre-filling caller's address with 217.162.26.26:3000 Jul 2 09:40:49 dns2 rtpproxy[19050]: INFO:handle_command: RTP packets from caller will be resized to 20 milliseconds Jul 2 09:40:49 dns2 rtpproxy[19050]: DBUG:doreply: sending reply "19072_4 54858 15.43.10.128 " Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:process_rtp: session timeout Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Jul 2 09:41:51 dns2 rtpproxy[19050]: INFO:remove_session: session on ports 54858/0 is cleaned up

1 0

UAC, parsing auth header failed
by Ján ONDREJ (SAL) 02 Jul '10

02 Jul '10

Hello, our voip provider requires autentication to voip services. Trying to use UAC module, I get "401 Unauthorized" packet, but looks like all ser based programs have problems to parse this packet. Can somebody make a patch to fix this? Here is message from logs: Jun 17 15:01:42 kamailio /usr/sbin/kamailio[1291]: ERROR: uac [auth_hdr.c:267]: parse error in <DIGEST realm="BroadWorks",qop="auth",algorithm=MD5,nonce="BroadWorksXgajlo4shTqsrkgdBW"> around 0 And ngrep sniffed packet: SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 100.100.242.20;branch=z9hG4bK119f.8a41a626.0. Via: SIP/2.0/UDP 100.100.16.77;branch=z9hG4bK119f.7cc235e.0. Via: SIP/2.0/UDP 100.100.10.10;branch=z9hG4bK8ec5c892A88F71C9. From: "Ondrej Jan" <sip:556807505@as.vvn.t-com.sk>;tag=89D71D23-32F3D9BC. To: <sip:0248213335@as.vvn.t-com.sk;user=phone>;tag=1621075049-1276779704129. CSeq: 2 INVITE. Call-ID: 41c7437-58259a9d-d157be06(a)100.100.10.10. WWW-Authenticate: DIGEST realm="BroadWorks",qop="auth",algorithm=MD5,nonce="BroadWorksXgajlo4shTqsrkgdBW". Content-Length: 0. I tryed to look at sources, but this "C" code is too complicated for me. :-( SAL

6 11

One way audio on calls from pstn
by Dmitri Korotkov 02 Jul '10

02 Jul '10

Hello, I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy. Using following scenario: [kamailio]<-sip trunk ->[asterisk gw] <->sip trunk <-> [PSTN provider] All kamailio sip subscribers are behind nat in different networks. 1. OK. Local kamailio users can call one to other even they are on different networks behind nat. 2. OK. Outgoing calls from kamailio users to PSTN work also very well. 3. Not OK. Incoming from PSTN side calls have only one way audio. I tcpdump'ed kamailio box and found, that pstn provider sends RTP packets to kamailio IP in case of answered call. I guess that rtpproxy is not active in case of pstn call. Is it true ? I am using more less default kamailio config Could you please suggest solution ? BR, Dmitri

2 6

Re: [SR-Users] sr-users Digest, Vol 62, Issue 8
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 7
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 6
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 19.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] sr-users Digest, Vol 62, Issue 5
by marco＠bungalski.de 01 Jul '10

01 Jul '10

Sehr geehrte Damen und Herren, leider kann ich Ihre e-mail momentan nicht beantworten / bearbeiten, da ich mich bis einschliesslich 12.07.2010 im Urlaub befinde. In dringenden Fällen wenden Sie sich bitte telefonisch an meine Mitarbeiter unter der Rufnummer 04231 / 776 9999. Mit freundlichen Grüßen Marco Bungalski

1 0

Re: [SR-Users] error loading perl module
by Daniel-Constantin Mierla 01 Jul '10

01 Jul '10

Hello, keep cc-ing the mailing list for any message, so next people that come and scan mailing list archive for the same issue discover the solution and whether was good or not. Cheers, Daniel On 6/30/10 11:40 PM, "Nicolas Rüger" wrote: > Hello Daniel, > > thanks for the fast answer. I followed your suggestion and now (after fixing a few minor bugs) it works! :) > > Many many thanks... > > Nicolas > > > > > > -------- Original-Nachricht -------- > >> Datum: Wed, 30 Jun 2010 21:44:36 +0200 >> Von: Daniel-Constantin Mierla<miconda(a)gmail.com> >> An: "Nicolas Rüger"<NicolasRueger(a)gmx.de> >> CC: sr-users(a)lists.sip-router.org >> Betreff: Re: [SR-Users] error loading perl module >> > >> Hello, >> >> On 6/30/10 9:24 PM, "Nicolas Rüger" wrote: >> >>> Hello, >>> >>> I try to use the PERL module in kamailio v3 for some research due to my >>> >> thesis. >> >>> I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. >>> >>> >>> 1.) The Problem >>> >>> Starting Kamailio via "./kamctl start" drops the general error JUST when >>> >> I try to load the PERL module: >> >>> ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start >>> >> failed >> >>> >>> >>> >>> 2.) The search for errors >>> >>> I got the following errors (and warnings) in DEBUG mode by starting >>> >> "kamailio -d" for debugging: >> >>> [...] >>> >>> 0(10776) DEBUG:<core> [sr_module.c:382]: load_module: trying to >>> >> load</opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so> >> >>> 0(10776) WARNING:<core> [sr_module.c:445]: >>> >> /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will >> not be supported in newer versions; consider using mod_register() instead >> 0(10776) >> >>> [...] >>> >>> 0(10778) ERROR: perl [perl.c:330]: insufficient module parameters. >>> >> Module not loaded. >> >>> 0(10778) ERROR:<core> [sr_module.c:874]: init_mod(): Error while >>> >> initializing module perl >> >>> >>> >> seems you don't have the parameter 'filename' set: >> >> http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2928387 >> >> Cheers, >> Daniel >> >> >> >>> >>> >>> 3.) What I did before... >>> >>> 3.1) Compiled Kamailio >>> >>> I compiled Kamailio 3.0.0 using the following commands, where $1 is the >>> >> target: >> >>> make prefix=$1/ include_modules="db_mysql tls perl" cfg >>> >>> make Q=1 all >>> >>> make install >>> >>> #make prefix=$1/ TLS=1 >>> #make prefix=$1/ TLS=1 modules >>> #make prefix=$1/ TLS=1 install >>> >>> #make prefix=$1/ include_modules="db_mysql" modules >>> #make prefix=$1/ include_modules="db_mysql" install >>> >>> >>> 3.2) Set the environment variables as described in >>> >>> http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2920768 >>> >>> You need to set the environment variables PERLLDOPTS, PERLCCOPTS and >>> >> TYPEMAP to values similar to the output of >> >>> PERLLDOPTS: perl -MExtUtils::Embed -e ldopts >>> PERLCCOPTS: perl -MExtUtils::Embed -e ccopts >>> TYPEMAP: echo "`perl -MConfig -e 'print >>> >> $Config{installprivlib}'`/ExtUtils/typemap" >> >>> 3.3) Trying to load the perl-module via kamailio.cfg >>> >>> loadmodule "perl.so" >>> >>> >>> >>> >>> Can you PLEASE help me to solve the problem! >>> >>> Thank you very much... >>> >>> Cheers Nicolas >>> >>> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com/ >> > -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

using avpops for querying an external db on kamailio 3.0.2
by yaron nahcum 01 Jul '10

01 Jul '10

Hi, I have recently installed a kamailio server with release 3.0.2. I would like to use the kamilio as a redirect server that would recieve REGISTER messages from users and query a proprietary database that holds users information - if the users are registered to the service it would redirect the user to the registrar. I need it for security reasons which I can't fully explain. Now, I am using the avpops module. I have been able to make it work using the following configuration: modparam("avpops","db_url","mysql://user:pass@localhost/mysql") modparam("avpops","avp_table","proptable") modparam("avpops","db_scheme","scheme0:username_col=sipUsername;value_col=sipDomainName;table=proptable") ---------- if (is_method("REGISTER")) { if (avp_db_load("$fu/username","$avp(s:CONSTANT_DOMAIN)/$scheme0")) { append_branch("sip:registrarIP"); sl_send_reply("302","Test Redirect"); } else { sl_send_reply("404","Test Not found"); } -------------- The thing is that I don't need to match the string CENTREX against the domain- it is something I added because I couldn't find another way to do it. I would like to match only the username. I tried wildcards or leaving it empty but it didn't work. I tried not to use the scheme and instead use the table it self but again it didn't work. Is it possible? Do you know a better to do it? Thanks, Yaron.

5 5

error loading perl module
by "Nicolas Rüger" 30 Jun '10

30 Jun '10

Hello, I try to use the PERL module in kamailio v3 for some research due to my thesis. I'm using debian lenny, kamailio_3.0.0 and perl_5.10.0. 1.) The Problem Starting Kamailio via "./kamctl start" drops the general error JUST when I try to load the PERL module: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed 2.) The search for errors I got the following errors (and warnings) in DEBUG mode by starting "kamailio -d" for debugging: [...] 0(10776) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so> 0(10776) WARNING: <core> [sr_module.c:445]: /opt/kamailio-3.0.0/lib/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead 0(10776) [...] 0(10778) ERROR: perl [perl.c:330]: insufficient module parameters. Module not loaded. 0(10778) ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module perl 3.) What I did before... 3.1) Compiled Kamailio I compiled Kamailio 3.0.0 using the following commands, where $1 is the target: make prefix=$1/ include_modules="db_mysql tls perl" cfg make Q=1 all make install #make prefix=$1/ TLS=1 #make prefix=$1/ TLS=1 modules #make prefix=$1/ TLS=1 install #make prefix=$1/ include_modules="db_mysql" modules #make prefix=$1/ include_modules="db_mysql" install 3.2) Set the environment variables as described in http://kamailio.org/docs/modules/stable/modules_k/perl.html#id2920768 You need to set the environment variables PERLLDOPTS, PERLCCOPTS and TYPEMAP to values similar to the output of PERLLDOPTS: perl -MExtUtils::Embed -e ldopts PERLCCOPTS: perl -MExtUtils::Embed -e ccopts TYPEMAP: echo "`perl -MConfig -e 'print $Config{installprivlib}'`/ExtUtils/typemap" 3.3) Trying to load the perl-module via kamailio.cfg loadmodule "perl.so" Can you PLEASE help me to solve the problem! Thank you very much... Cheers Nicolas -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

2 1

force_send_socket - multiple times per dialog, subsequent times fail
by Geoffrey Mina 30 Jun '10

30 Jun '10

I am having an issue with 1.5.4 where force_send_socket() isn't behaving as I would expect. I have an LCR scenario where depending on which gateway I am sending to, I send from a different socket. It appears that calling force_send_socket from failure_route has no affect on the routing. In my initial route block I call force_send_socket(X.X.X.180:5060). In my failure_route I am calling force_send_socket(X.X.X.179:5060), yet after calling the subsequent time, my sip trace and tcpdump are clearly showing kamailio is still sending from X.X.X.180:5060. I have added a lot of debug lines, so I am 100% sure the second call to force_send_socket is happening. Am I misusing this module in some way?

5 8

[nathelper] clarification about rtpproxy_retr parameter
by Iñaki Baz Castillo 30 Jun '10

30 Jun '10

Hi, according to the doc: 1.5.9. rtpproxy_retr (integer) How many times nathelper should retry to send and receive after timeout was generated. Default value is “5”. So imagine this configuration: modparam("nathelper", "rtpproxy_tout", 2) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_retr", 10) Let's imagine that the RtpProxy servers crashes and is down for 1 day. Does this configuration mean that the rtpproxy server will NEVER be contacted again by nathelper module since 60x10 seconds < 1 day? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Dispatcher module problem or feature?
by JR Richardson 29 Jun '10

29 Jun '10

Hi All, I'm testing dispatcher functions with kamailio 3.0 and using database to load destination address. The problem I am seeing is the 1'st destination address in a group is not being used, the dispatcher always starts at the second database entry. For instance, if I have 2 destination addresses listed in the database, sip:10.10.14.101:5060 and sip:10.10.14.102:5060, only sip:10.10.14.102:5060 will receive calls. But if I list .101 in the database twice, then I will get proper distribution. So here is my setup and some debug traffic: sipp><sr-router><dispatcher round robin group 1><10.10.14.101, 10.10.14.102, 10.10.14.101 This will send calls to both .101 and 102 evenly. sip-router2:~# kamctl fifo ds_list SET:: 1 URI:: sip:10.10.14.101:5060 flag=A URI:: sip:10.10.14.102:5060 flag=A URI:: sip:10.10.14.101:5060 flag=A first call to dispatcher: 0(4780) DEBUG: dispatcher [dispatch.c:1111]: set [1] 0(4780) DEBUG: dispatcher [dispatch.c:1185]: alg hash [0] 0(4780) DEBUG: dispatcher [dispatch.c:1223]: selected [4-1/0] <sip:10.10.14.101:5060> 0(4780) DEBUG: dispatcher [dispatch.c:1257]: using entry [1/1] 0(4780) INFO: <script>: ds_dispatcher sip:10.10.14.101:5060 1 3 second call to dispatcher: 0(4780) DEBUG: dispatcher [dispatch.c:1111]: set [1] 0(4780) DEBUG: dispatcher [dispatch.c:1185]: alg hash [1] 0(4780) DEBUG: dispatcher [dispatch.c:1223]: selected [4-1/1] <sip:10.10.14.102:5060> 0(4780) DEBUG: dispatcher [dispatch.c:1245]: using entry [1/0] 0(4780) INFO: <script>: ds_dispatcher sip:10.10.14.102:5060 1 3 third call to dispatcher goes back to .101, then .102, ect. sipp><sr-router><dispatcher round robin group 2><10.10.14.103 and 10.10.14.104 This will send calls only to .104 sip-router2:~# kamctl fifo ds_list SET_NO:: 2 SET:: 2 URI:: sip:10.10.14.104:5060 flag=A URI:: sip:10.10.14.103:5060 flag=A first call to dispatcher: 0(4780) DEBUG: dispatcher [dispatch.c:1111]: set [2] 0(4780) DEBUG: dispatcher [dispatch.c:1185]: alg hash [0] 0(4780) DEBUG: dispatcher [dispatch.c:1223]: selected [4-2/0] <sip:10.10.14.104:5060> 0(4780) INFO: <script>: ds_dispatcher sip:10.10.14.104:5060 2 2 second call to dispatcher: 0(4780) DEBUG: dispatcher [dispatch.c:1111]: set [2] 0(4780) DEBUG: dispatcher [dispatch.c:1185]: alg hash [1] 0(4780) DEBUG: dispatcher [dispatch.c:1223]: selected [4-2/0] <sip:10.10.14.104:5060> 0(4780) INFO: <script>: ds_dispatcher sip:10.10.14.104:5060 2 2 So it appears with only 2 entries loaded in the database, there is a missing "DEBUG: dispatcher [dispatch.c:1245]: using entry [1/0]" This is the only difference I can see between the call executions between the two groups. But I don't really know what that means. I can reproduce these symptoms on kamailio 3.0, 1.5.x and older versions. The same thing happens when the destinations are loaded via config file dispatcher.lst. I am using debian Lenny with siremis web interface. mysql> select * from dispatcher\G; *************************** 1. row *************************** id: 1 setid: 1 destination: sip:10.10.14.101:5060 flags: 0 priority: 0 description: lab1 *************************** 2. row *************************** id: 2 setid: 1 destination: sip:10.10.14.102:5060 flags: 0 priority: 0 description: lab2 *************************** 3. row *************************** id: 3 setid: 2 destination: sip:10.10.14.103:5060 flags: 0 priority: 0 description: lab3 *************************** 4. row *************************** id: 4 setid: 2 destination: sip:10.10.14.104:5060 flags: 0 priority: 0 description: lab4 *************************** 5. row *************************** id: 5 setid: 1 destination: sip:10.10.14.101:5060 flags: 0 priority: 0 description: lab1 5 rows in set (0.00 sec) Is this a bug? Thanks. JR -- JR Richardson Engineering for the Masses

2 6

Re: [SR-Users] using avpops for querying an external db on kamailio 3.0.2
by Daniel-Constantin Mierla 29 Jun '10

29 Jun '10

Hello, On 6/29/10 6:39 PM, yaron nahcum wrote: > Hi Daniel, > Thanks a lot for the quick response. I will look at it. fyi: keep cc-ing the mailing list for any discussion that started there - message sent privately are usually ignored. Daniel > Yaron. > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* yaron nahcum <yaron_nachum(a)yahoo.com> > *Cc:* sr-users(a)lists.sip-router.org > *Sent:* Tue, June 29, 2010 6:57:53 PM > *Subject:* Re: [SR-Users] using avpops for querying an external db on > kamailio 3.0.2 > > Hello, > > look at sqlops module, it is more suitable for custom sql queries: > http://kamailio.org/docs/modules/stable/modules_k/sqlops.html > > Cheers, > Daniel > > > On 6/29/10 5:47 PM, yaron nahcum wrote: >> Hi, >> I have recently installed a kamailio server with release 3.0.2. >> I would like to use the kamilio as a redirect server that would >> recieve REGISTER messages from users and query a proprietary database >> that holds users information - if the users are registered to the >> service it would redirect the user to the registrar. I need it for >> security reasons which I can't fully explain. >> Now, I am using the avpops module. I have been able to make it work >> using the following configuration: >> modparam("avpops","db_url","mysql://user:pass@localhost/mysql") >> modparam("avpops","avp_table","proptable") >> modparam("avpops","db_scheme","scheme0:username_col=sipUsername;value_col=sipDomainName;table=proptable") >> ---------- >> >> if (is_method("REGISTER")) >> { >> if >> (avp_db_load("$fu/username","$avp(s:CONSTANT_DOMAIN)/$scheme0")) >> { >> append_branch("sip:registrarIP"); >> sl_send_reply("302","Test Redirect"); >> } >> else >> { >> sl_send_reply("404","Test Not found"); >> } >> -------------- >> The thing is that I don't need to match the string CENTREX against >> the domain- it is something I added because I couldn't find another >> way to do it. I would like to match only the username. I tried >> wildcards or leaving it empty but it didn't work. I tried not to use >> the scheme and instead use the table it self but again it didn't work. >> Is it possible? Do you know a better to do it? >> Thanks, >> Yaron. >> >> >> _______________________________________________ >> 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/

1 0

Re: [SR-Users] Permissions module question?
by JR Richardson 29 Jun '10

29 Jun '10

> 2010/6/28 JR Richardson <jmr.richardson(a)gmail.com>: > > Kamailio 3.0 permissions module, I see that the address database table > > is cached by default, really the only option. ?But there is no MI > > Function to reload the database into cache, you must restart kamailio > > to update the cache for the address table. ?Is there any reason the > > reload MI function is not there for the address table? > > It is: > http://kamailio.org/docs/modules/stable/modules_k/permissions.html#id29613 > 70 Yes, I see that now, I was looking for subnet_reload, didn't realize that the address_ and subnet_ entries were in the same table. Thanks. JR

1 0

WARNING: timer: add_timeout: 0 expire timer added in pua
by David 29 Jun '10

29 Jun '10

Hello, Using Kamailio 3.0.1 : My syslog is being flooded with this : WARNING: <core> [timer_funcs.h:111]: WARNING: timer: add_timeout: 0 expire timer added Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9322.13398.2 - len= 25 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.9b5f4c1c67b65f80(a)192.168.70.127 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 154 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:42]: ^Ipres_uri= sip:development1.106@dev.mydomain.tld len= 40 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9317.13425.1 - len= 25 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.2497e17925460cf05484d1b21909aca1(a)192.168.58.253 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 158 Jun 25 12:23:55 kamailio-dev /usr/sbin/kamailio[11259]: WARNING: <core> [timer_funcs.h:111]: WARNING: timer: add_timeout: 0 expire timer added Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:42]: ^Ipres_uri= sip:development1.110@dev.mydomain.tld len= 40 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9322.13398.2 - len= 25 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.9b5f4c1c67b65f80(a)192.168.70.127 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 153 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:42]: ^Ipres_uri= sip:development1.106@dev.mydomain.tld len= 40 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9317.13425.1 - len= 25 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.2497e17925460cf05484d1b21909aca1(a)192.168.58.253 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 157 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: WARNING: <core> [timer_funcs.h:111]: WARNING: timer: add_timeout: 0 expire timer added Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:42]: ^Ipres_uri= sip:development1.110@dev.mydomain.tld len= 40 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9322.13398.2 - len= 25 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.9b5f4c1c67b65f80(a)192.168.70.127 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 153 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:42]: ^Ipres_uri= sip:development1.106@dev.mydomain.tld len= 40 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:54]: ^Ietag= a.1276278619.9317.13425.1 - len= 25 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:56]: ^Iid= DIALOG_PUBLISH.2497e17925460cf05484d1b21909aca1(a)192.168.58.253 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: DEBUG: pua [hash.c:58]: ^Iexpires= 157 Jun 25 12:23:56 kamailio-dev /usr/sbin/kamailio[11259]: WARNING: <core> [timer_funcs.h:111]: WARNING: timer: add_timeout: 0 expire timer added Why is this timer being set to 0? What is this timer ? How do I set it at a more reasonable value ? Because this is at 0 it is basically consuming all my server resources cause the timer is completed instantly, so the message repeats hundreds of times per second. Thank, David

2 1

Permissions module question?
by JR Richardson 29 Jun '10

29 Jun '10

Hi All, Kamailio 3.0 permissions module, I see that the address database table is cached by default, really the only option. But there is no MI Function to reload the database into cache, you must restart kamailio to update the cache for the address table. Is there any reason the reload MI function is not there for the address table? It is present for the trusted table and works fine. I like the trusted db option to use cache or no cache, I'm wondering if this option can be extended to the address table as well? Also I like the address table ability to define subnets, can this be extended to the trusted table? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

New functionality LCR + force_send_socket() - sanity check
by Geoffrey Mina 28 Jun '10

28 Jun '10

Hey everyone, Just wanted to bounce some things off the list to make sure I'm not heading down the wrong path here. I'll quickly explain my problem and then explain my plan to deal with it. Please let me know if anyone sees an issue here. Problem: I need to send FROM two different IP addresses based on the LCR group returned from next_gw();. For example, if the gateway returned is in group_id=1 from the LCR I need to send FROM 1.1.1.1:5060, if it's from group_id=2, I need to send FROM 1.1.1.2:5060. I need to do this on REQUEST_ROUTE and FAILURE_ROUTE. Here is what I am thinking will work: modparam("pv","shvset","gw_one=s:1.1.1.1:5060") modparam("pv","shvset","gw_two=s:1.1.1.2:5060") route{ load_gws(); next_gw(); if(to_gw("1")){ force_send_socket($shv(gw_one)); }else if(to_gw("2")){ force_send_socket($shv(gw_two)); } } Sorry for not just trying this out on my own. I currently don't have access to a dev machine, and I need to make changes to the live script... so I would prefer not to have a ton of restarts while trying to work the bugs out. Thanks.

4 17

job offer: VoIP developer, team Karlsruhe
by Henning Westerholt 28 Jun '10

28 Jun '10

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional VoIP developer position open in our team in Karlsruhe, Germany at the moment. For more informations (requirements, how to apply..) please refer to the opening on the job web page. Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. http://www.united-internet.de/xml/ui/JobsJobDetail;?__lf=JobsApplication200… Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2010-linuxtag/presentation.pdf -- Henning Westerholt - Technical Leader VoIP backend development 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 5

Check for existence of URI param
by Alex Balashov 28 Jun '10

28 Jun '10

Greetings, I have a Contact URI with params as in the following form returned to me in a 302 Moved Temporarily reply: <sip:a=foo;b;c=bar> I am able to check the existence of parameters 'a' and 'b', which have values assigned to them, in the following way: failure_route[...] { ... $var(dest) = $T_rpl($ct); $var(dest) = $(var(dest){nameaddr.uri}); $var(dest) = $(var(dest){uri.user}); ... if(! $(var(dest){param.value,a})) { .... } ... } However, what I cannot seem to figure out is a reliable test condition for the existence of a parameter like 'b', which does not have a value. I have tried: if(! $(var(dest){param.value,b})) which clearly cannot work as the parameter has no _value_. Also have tried: if(!defined $(var(dest){param.value,b})) and also tests of strempty(), !defined, etc. for $var(dest){uri.param,b}) (prior to extraction of param list using {uri.user} above). Ideas appreciated, 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/

3 5

how to evaluate a string value?
by Juha Heinanen 26 Jun '10

26 Jun '10

http_query function takes an url as the first argument, for example, http_query("http://localhost/test.php?uri=$(ru{s.escape.param})", "$var(result)"); what i would like to do is this: http_query("$sel(cfg_get.local.http_query_url)", "$var(result)"); that is, instead of giving url argument as a fixed string, i would like to take it from cfg variable. the problem is that if i give local.http_query_url string value "http://localhost/test.php?uri=$(ru{s.escape.param})", pseudo variables and transformations in the value are not evaluated. is there any current means to get them evaluated? something like http_query(eval("$sel(cfg_get.local.http_query_url)"), "$var(result)"); ???? -- juha

2 1

Pipes in modparam
by David 25 Jun '10

25 Jun '10

Hey, Is this format still valid in Kamailio 3.0 ? modparam("module1|module2|module3", "db_url", "mysql://me:secret@server/database") This line : modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", "db_url", "mysql://kamailiodev:kamailiosecret@192.168.1.10/kamailiodev") gives me this error : -e 0(9334) ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <db_url> not found in module <pua_dialoginfo> 0(9334) : <core> [cfg.y:3333]: parse error in config file /etc/kamailio/kamailio.cfg, line 399, column 153: Can't set module parameter ERROR: bad config file (1 errors) 0(9334) ERROR: dialog [dlg_db_handler.c:177]: invalid database handle Funny thing is I have not set the module parameter for pua_dialoginfo, looks like kamailio is getting confused with my pipes, if I name each parameter one at a time, it works fine. Any thoughts on this? David

2 3

[OT] sip-router world cup 2010 edition
by Henning Westerholt 25 Jun '10

25 Jun '10

Hi, the sip-router world cup 2010 edition: http://www.vuvuzela-time.co.uk/sip-router.org SCNR ;-) Henning

3 4

Question about rate limiting a subscriber
by Stagg Shelton 24 Jun '10

24 Jun '10

I have a customer who has just started operating a dialer. Over the past couple of days this dialer has been creating some minor issues with our US48 termination. We are currently running Server:: Kamailio (1.5.1-notls (x86_64/linux)). I would like to limit on a subscriber by subscriber basis the amount of outgoing calls per second that they can do. I am reading through the ratelimit module documentation, but haven't been able to determine if I can use it to limit one particular subscriber outgoing activity. I have also looked at the pike module, and think that I may be able to use it in order to limit calls per second by IP. Does anyone have any recommendations, suggestions, or examples that can be used to control the calls per second based on the kamailio subscriber name. Stagg Shelton VocalCloud

3 6

memcached compile problems
by Ricardo Martinez 24 Jun '10

24 Jun '10

Hello list. I'm using the GIT versión of Kamailio 3.0. When I add memcached in the compilation I got the next error : make[1]: Entering directory `/usr/src/kamailio/modules_k/memcached' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m32 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=athlon64 -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.4"' -D__CPU_i386 -D__OS_linux -DSER_VER=3000002 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -DOPENSER_MOD_INTERFACE -I/usr/local/include -I/include -DMOD_NAME='"memcached"' -c mcd_var.c -o mcd_var.o -MMD -MP In file included from mcd_var.c:29: memcached.h:27:22: warning: memcache.h: No such file or directory What could be the problem. I have installed memcached , memcached-devel , libmemcached and libmemcached-devel Thanks Ricardo.-

4 4

(no subject)
by rajesh volluru 24 Jun '10

24 Jun '10

http://kmkhzr.http://tabletsdrugs.com/

1 0

new developer: Timo Reimann
by Henning Westerholt 23 Jun '10

23 Jun '10

Hi all, i'm happy to announce a new developer for the sip-router project: Timo Reimann. Timo is an experienced developer in our offices in Karlsruhe and since over one year part of the team that develops our VoIP backend services. His past work involved extensions for the dialog module and also a new lawful interception solution that we use in-house. He contributed (with some other developers) to the new dialog design in the last months and will continue to work on this in the future. Best regards, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

4 3

Daniel-Constantin Mierla to Speak at ClueCon Open Source Telephony Conference
by Michael Collins 22 Jun '10

22 Jun '10

The ClueCon team would like to let everyone in the Kamailo community know that Daniel-Constantin Mierla is one of many speakers at this year's ClueCon event. ClueCon MMX will be held in downtown Chicago, August 3-5, at the Trump Hotel and Tower. Daniel's talk is entitled, "Blending FreeSWITCH and Kamailio to Build Large Unified Communication Platforms." It will be one of many highlights for those attending this year. Speakers, sponsors, and attendees will be coming from around the world to attend. ClueCon is a great place to meet fellow open source telephony developers, users, and businesspersons. Among the speakers this year are: Anthony Minessale, author of the FreeSWITCH project Michael Picher, author of "Building Enterprise Ready Telephony Systems with sipXecs 4.0" (Packt) Matt Florell, author of the ViciDial Call Center Suite Moises Silva, software engineer, Sangoma Technologies Darren Schreiber, founder of the 2600Hz project and former lead developer of FreePBX V3 ClueCon attendees will enjoy not only the formal presentations by these and other speakers but they will have ample time after the sessions to interact directly with them. Attendees also have time to speak directly with other delegates as well as representatives of ClueCon sponsors - all in a relaxed environment without the traditional gauntlet of vendor booths. A highlight that everyone looks forward to each year is the collection of giveaways. This year ClueCon will give away two iPads and a Macbook Pro! Additionally, the sponsors will be giving away lots of great prizes. All paid attendees are eligible to win. To learn more about ClueCon visit http://www.cluecon.com or call 877-742-CLUE (2583). We hope to see many Kamailio community members at ClueCon this year! The ClueCon Team http://www.clueccon.com 877.742.CLUE

1 0

SIPv6 UA Can't receive a 420 (Bad Extension) response from openser register server after sending Registrar with Non-supported option-tag in Require filed
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, SIPv6 UA Can't receive a 420 (Bad Extension) response from openser register server after sending Registrar with Non-supported option-tag in Require filed. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "19 SIP Registrar- REGISTER request with two Contact header values and update failure.pcap" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive 420 (Bad Extension) ? -------------------------------------- Regards! Hu xiaoqiang

3 4

SIPv6 UA Can't receive a 500 (Server Error) response from openser register server after sending REGISTER request with two Contact header field values
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, SIPv6 UA Can't receive a 500 (Server Error) response from openser register server after sending REGISTER request with two Contact header field values. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "19 SIP Registrar- REGISTER request with two Contact header values and update failure.pcap" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive 404 (Not Found)? -------------------------------------- Regards! Hu xiaoqiang

3 4

Can't receive a 200 0k response from openser register server when URI in a To header field in a REGISTER request contains escaped characters.
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, Can't receive a 200 0k response from openser register server when URI in a To header field in a REGISTER request contains escaped characters. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "14 SIP Registrar- SIP Register-Sequence Number is not Incremented.pcap" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive the 200 OK? -------------------------------------- Regards! Hu xiaoqiang

3 4

Does the portsip sdk is compatibles with Kamailio?
by list mail 22 Jun '10

22 Jun '10

Hi all, I'm working for a small ITSP, we have been setup an Kamailio server, now we want to develop a dialer for our user, we have been tested the portsip sdk(www.portsip.com), it's working fine with Kamailioso far, but we not doing long time test, does anyone knows this product is working good with Kamailio? Or have any other dialer SDK provider to recommend? Thanks

2 1

GSoC2010: welcome Marius Bucur
by Daniel-Constantin Mierla 22 Jun '10

22 Jun '10

Hello, GSoC2010 selected projects were announced several days ago. I want to welcome Marius Bucur, students at Computer Science Faculty in Bucharest. He is going to implement our proposed project about presence support for audio conference mixers: http://www.sip-communicator.org/index.php/GSOC2010/Kamailio4575 It will be tight related to implementation in SEMS conference mixer: http://www.sip-communicator.org/index.php/GSOC2010/SEMSAudioLevels Shortly, is about getting presence updates about participants in a conference room, who is speaking, audio levels of speakers, etc. SIP Communicator is a softphone that has support for this features, see a presentation about it at: http://www.kamailio.org/events/2010-FOSDEM/SIP.Communicator-FOSDEM-2010.pdf http://www.sip-communicator.org/ Please provide Marius hints and guidelines as fast as you can for him to get quickly into application architecture. Along with me, Marius Zbihlei will officially help in mentoring, if you have questions about GSoC project administration, etc., contact one of us. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

3 2

install SER on UBUNTU 7.04
by Josef Tupag 22 Jun '10

22 Jun '10

Hi everyone. im posting this to help anyone pretending to install SER on UBUNTU 7.04. Steps: 1. download the source from http://ftp.iptel.org/pub/ser/LATEST-IS-0.9.6/src/ 2. upgrade your build utils by downloading the following packages needed packages for compiling all the modules: Debian: - libmysqlclient-dev for libmysqlclient - libpq-dev for libpq - libexpat1-dev for libexpat - libxml2-dev for libxml2 - libradiusclient-ng-dev for libradiusclient (you can download the package from http://apt.sip-router.org/debian/dis...1-0.5_i386.deb<http://apt.sip-router.org/debian/dists/unstable/main/binary-i386/libradiusc…>). NOTE: you can get up-to-date ser packages or libradiusclient packages from http://apt.sip-router.org: <http://apt.sip-router.org/> add to your /etc/apt/sources.list the following lines: deb http://apt.sip-router.org/debian testing main contrib non-free deb http://apt.sip-router.org/debian unstable main contrib non-free and then: apt-get update; apt-get install libradiusclient-ng-dev (or, if you want to use the pre-built modules: apt-get install ser ser-cpl-module ser-jabber-module ser-mysq-module ser-pa-module ser-postgres-module ser-radius-modules ) Compile example (all the modules and ser in a tar.gz): make bin include_modules="mysql jabber cpl-c auth_radius group_radius uri_radius postgres pa" Since my system was quite updated i have just issued " apt-get install libpq-dev " on shell the as root on ser extracted folder issued make install. everythings were successfully compiled and installed. i recommend u to do the following on section 2 of this link http://ftp.iptel.org/pub/ser/0.9.6/doc/INSTALL 3. then follow the installation of the sip User agent (client ) on http://www.en.voipforo.com/Telephone...figuration.php<http://www.en.voipforo.com/Telephones/sjphone-configuration.php> enjoy uself Josef Tupag humidifiers <http://thebesthumidifiers.com>

1 0

Re: [SR-Users] SIPv6 UA Can't receive a 500 (Server Error) response from openser register server after sending REGISTER request with two Contact header field values
by Iñaki Baz Castillo 22 Jun '10

22 Jun '10

Please keep the thread in the maillist. 2010/6/22 Xiaoqiang Hu <xhu(a)redhat.com>: > REFERENCE] > [RFC3261-10-47, 48, 49] > 10.3 Processing REGISTER Requests > 7. The binding updates MUST be commited (that is, made visible to > the proxy or redirect server) if and only if all binding > updates and additions succeed. If any one of them fails (for > example, because the back-end database commit failed), the > request MUST fail with a 500 (Server Error) response and all > tentative binding updates MUST be removed. > -------------------------------------- Do you suggest that the REGISTER contains "invalid" Contact URI's? Not at all. URI's in the Contact DON'T require to have same username as the AoR being registered (the URI in To header). Again, the REGISTER is 100% valid. PD: If you reference some text in a RFC please append also what you interpret from such text ;) -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

SIPv6 UA Can't receive a 200 ok response from openser register server after sending Forwarded REGISTER request
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, SIPv6 UA Can't receive a 200 ok response from openser register server after sending Forwarded REGISTER request. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "22 SIP Registrar- Forwarded REGISTER request.pcap" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive 200 OK? -------------------------------------- Regards! Hu xiaoqiang

2 3

Re: [SR-Users] Can't receive a 200 0k response from openser register server when URI in a To header field in a REGISTER request contains escaped characters.
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

sorry,here it is:) NUT UA11 UA12 UA13 UA14 PX2 PX3 PX4 DNS No time | | | | | | | | | No time | | | | | | | | | [0001: 0.00| ] |--->| | | | | | | | NS [0002: 0.00| ] |<---| | | | | | | | NA [0003: 3.00|U ] REGIS |<---| | | | | | | | REGISTER sip:ss.under.test.com:5060 [0004: 3.00|U ] 401 |--->| | | | | | | | 401 Unauthorized (REGISTER) [0005: 3.07|U ] REGIS |<---| | | | | | | | REGISTER sip:ss.under.test.com:5060 [0006: 3.07|U ] 403 |--->| | | | | | | | 403 Forbidden auth ID (REGISTER) 0003: <UA11 REGISTER sip:ss.under.test.com:5060 SIP/2.0 Via: SIP/2.0/UDP node.under.test.com:5060;branch=z9hG4bKPUA9531271 Max-Forwards: 70 From: UA11 <sip:U%4111@under.test.com>;tag=11171 To: UA11 <sip:U%4111@under.test.com> Call-ID: 11171(a)sun.example.com CSeq: 1 REGISTER Contact: <sip:U%4111@node.under.test.com;transport=udp> Expires: 3600 Content-Length: 0 0004: >UA11 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP node.under.test.com:5060;branch=z9hG4bKPUA9531271;received=3FFE:501:FFFF:1:0:0:0:1 From: UA11 <sip:U%4111@under.test.com>;tag=11171 To: UA11 <sip:U%4111@under.test.com>;tag=0c0d2af89845c72ddfdc7de2133116df.46dd Call-ID: 11171(a)sun.example.com CSeq: 1 REGISTER WWW-Authenticate: Digest realm="under.test.com", nonce="4c147cfa07dd3f3a059e319e43cfc80d0b8b78db", qop="auth" Server: OpenSER (1.3.4-tls (i386/linux)) Content-Length: 0 0005: <UA11 REGISTER sip:ss.under.test.com:5060 SIP/2.0 Via: SIP/2.0/UDP node.under.test.com:5060;branch=z9hG4bKPUA9531272 Max-Forwards: 70 Authorization: Digest username="UA11", realm="under.test.com", qop=auth, nonce="4c147cfa07dd3f3a059e319e43cfc80d0b8b78db", opaque="", nc=00000001, cnonce="6f54a149", uri="sip:ss.under.test.com:5060", response="17f29661e1187c5da753955ab11707ec" From: UA11 <sip:U%4111@under.test.com>;tag=11171 To: UA11 <sip:U%4111@under.test.com> Call-ID: 11171(a)sun.example.com CSeq: 2 REGISTER Contact: <sip:U%4111@node.under.test.com;transport=udp> Expires: 3600 Content-Length: 0 0006: >UA11 SIP/2.0 403 Forbidden auth ID Via: SIP/2.0/UDP node.under.test.com:5060;branch=z9hG4bKPUA9531272;received=3FFE:501:FFFF:1:0:0:0:1 From: UA11 <sip:U%4111@under.test.com>;tag=11171 To: UA11 <sip:U%4111@under.test.com>;tag=0c0d2af89845c72ddfdc7de2133116df.fffe Call-ID: 11171(a)sun.example.com CSeq: 2 REGISTER Server: OpenSER (1.3.4-tls (i386/linux)) Content-Length: 0 -------------------------------------- Regards! Hu xiaoqiang Quality Assurance Engineer IRC Account:xhu Channel:#apac #apac-qa ----- Original Message ----- From: "Iñaki Baz Castillo" <ibc(a)aliax.net> To: "Xiaoqiang Hu" <xhu(a)redhat.com> Cc: sr-users(a)lists.sip-router.org Sent: Tuesday, June 22, 2010 6:09:10 AM GMT -05:00 US/Canada Eastern Subject: Re: [SR-Users] Can't receive a 200 0k response from openser register server when URI in a To header field in a REGISTER request contains escaped characters. 2010/6/22 Xiaoqiang Hu <xhu(a)redhat.com>: > Test sequence are as follows: Hi, I mean a real SIP trace in which I could inspect the REGISTER headers and so. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 1

SIPv6 UA Can't receive 404 (Not Found) response from openser register server after sending REGISTER request with an invalid AoR
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

SIPv6 UA Can't receive 404 (Not Found) response from openser register server after sending REGISTER request with an invalid AoR. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "17 SIP Registrar- REGISTER request with an invalid AoR.pcap" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive 404 (Not Found)? -------------------------------------- Regards! Hu xiaoqiang

2 1

Fwd: A Record-Route header field exists in 200 OK
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, A Record-Route header field exists in 200 OK from openser register server after sending a REGISTER request with a Record-Route header field. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "12 SIP Registrar- Record-Route header field in REGISTER request" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I remove the Record-Route field from 200 OK? -------------------------------------- Regards! Hu xiaoqiang

3 3

Re: [SR-Users] Can't login to Siremis.
by Henk Oegema 22 Jun '10

22 Jun '10

Vicente. Thanks for solving my problem. :-) You pointed me in the right direction. The mistake I made was this: ================== When reading the website Create user access table: * create a new table in Kamailio (OpenSER) database name ob_users: CREATE TABLE `ob_users` ( `SYSID` int(11) NOT NULL auto_increment, `USERID` varchar(15) NOT NULL, `PASSWORD` varchar(15) default NULL, PRIMARY KEY (`SYSID`) ); I created a new database ob_users with a table ob_users, in stead of inserting the table into the database openser. Thanks, Henk On Tue, 2010-06-22 at 08:54 +0200, Vicente Ferrando Berenguer wrote: > Have you created the ob_users table in openser database and added the > openser user and password you are using? Is MySQL accepting the > connection? > > > > Quite a few things can fail but DB connection seems to be the most > probable cause. Try with another browser also. Safari is not working > for me. And you can also try to clean the browser cache and cookies. > > > > El 21/06/2010, a las 16:42, Henk Oegema escribió: > > > > > Vicente > > > > > > I've tried with Server="localhost', Server="192.168.1.14" and > > Server="127.0.0.1, but I still can't login. <face-sad.png> > > > > <?xml version="1.0" standalone="no"?> > > <Config> > > <DataSource> > > <Database Name="Default" Driver="Pdo_Mysql" > > Server="localhost" DBName="openser" User="openser" > > Password="openserrw" /> > > <Database Name="Siremis" Driver="Pdo_Mysql" > > Server="localhost" DBName="openser" User="openser" > > Password="openserrw" /> > > </DataSource> > > </Config> > > ============================================================== > > <?xml version="1.0" standalone="no"?> > > <Config> > > <DataSource> > > <Database Name="Default" Driver="Pdo_Mysql" > > Server="192.168.1.14" DBName="openser" User="openser" > > Password="openserrw" /> > > <Database Name="Siremis" Driver="Pdo_Mysql" > > Server="192.168.1.14" DBName="openser" User="openser" > > Password="openserrw" /> > > </DataSource> > > </Config> > > ============================================================= > > <?xml version="1.0" standalone="no"?> > > <Config> > > <DataSource> > > <Database Name="Default" Driver="Pdo_Mysql" > > Server="127.0.0.1" DBName="openser" User="openser" > > Password="openserrw" /> > > <Database Name="Siremis" Driver="Pdo_Mysql" > > Server="127.0.0.1" DBName="openser" User="openser" > > Password="openserrw" /> > > </DataSource> > > </Config> > > > > > > On Mon, 2010-06-21 at 09:41 +0200, Vicente wrote: > > > > > Hi, > > > > > > Have you tried this? > > > On 05/27/2010 10:15 AM, Elena-Ramona Modroiu wrote: > > > > Note as well that in some OSes, connecting to openser database > > > from > > > > 127.0.0.1 is not same as localhost (which is used by kamctl to > > > create > > > > openser user and database), so try with localhost instead of > > > 127.0.0.1 > > > > in siremis/metadata/Config.xml, in case you still get problems. > > > > > > The problems you report seems to be related with the database > > > connection settings. > > > > > > Regards. > > > > > > El dom, 20-06-2010 a las 18:10 +0200, Henk Oegema escribió: > > > > > > > I'm trying to install Siremis 1.0.0 to work with Kamailio 3.0 > > > > and I'm having some problems which hopefully somebody can help > > > > with. > > > > Kamailio, the MySQL server and Siremis are running on the same > > > > server. The server is running Debian 5.0.4 > > > > > > > > I have followed the Siremis installation instructions at the > > > > following link: http://siremis.asipto.com/install/ > > > > > > > > I reach a point were I can get to the Siremis web interface > > > > login page but when I enter the user name and password nothing > > > > happens. > > > > > > > > > > > > /var/log/apache2/access.log : > > > > ============================= > > > > 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/ > > > > HTTP/1.1" 302 271 "-" "Mozilla/5.0 (X11; U; Linux i686; en-US; > > > > rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" > > > > 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] > > > > "GET /siremis/bin/ HTTP/1.1" 200 1083 "-" "Mozilla/5.0 (X11; U; > > > > Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 > > > > (lucid) Firefox/3.6.3" > > > > 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] > > > > "GET /siremis/js/sound.js HTTP/1.1" 404 270 > > > > "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux > > > > i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) > > > > Firefox/3.6.3" > > > > 192.168.1.101 - - [19/Jun/2010:15:53:23 +0200] > > > > "GET /siremis/js/sound.js HTTP/1.1" 404 270 > > > > "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux > > > > i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) > > > > Firefox/3.6.3" > > > > 192.168.1.101 - - [19/Jun/2010:15:53:38 +0200] > > > > "POST /siremis/bin/controller.php HTTP/1.1" 200 20 > > > > "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux > > > > i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) > > > > Firefox/3.6.3" > > > > > > > > /var/log/apache2/error.log : > > > > ============================ > > > > [Sat Jun 19 15:53:22 2010] [error] [client 192.168.1.101] File > > > > does not exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, > > > > referer: http://192.168.1.14/siremis/bin/ > > > > [Sat Jun 19 15:53:23 2010] [error] [client 192.168.1.101] File > > > > does not exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, > > > > referer: http://192.168.1.14/siremis/bin/ > > > > > > > > > > > > >

2 1

Can't receive a 500 (Server Internal Error) response from openser register server when sending a REGISTER with the equal CSeq header field value to the last value of CSeq or lower than the number and “*” in a Contact header field.
by Xiaoqiang Hu 22 Jun '10

22 Jun '10

Hi all, Can't receive a 500 (Server Internal Error) response from openser register server when sending a REGISTER with the equal CSeq header field value to the last value of CSeq or lower than the number and “*” in a Contact header field. The details can be as follows: 1 Test topology: NUT(REG && PX) UA11 UA12 DNS | | | | IP : 3ffe:501:ffff:50::50 3ffe:501:ffff:1::1 3ffe:501:ffff:2::2 3ffe:501:ffff:4::1 aor-uri : ss.under.test.com UA11(a)under.test.com UA12(a)under.test.com contact-uri : UA11(a)node.under.test.com UA12(a)node11.under.test.com 2 tcpdump file can be seen in "12 SIP Registrar- Record-Route header field in REGISTER request" attachment 3 openser configuration file can be seen in "openser.cfg" attachment How can I modify the configuration file if I want to receive the 500? -------------------------------------- Regards! Hu xiaoqiang

1 0

Can't login to Siremis.
by Henk Oegema 21 Jun '10

21 Jun '10

I'm trying to install Siremis 1.0.0 to work with Kamailio 3.0 and I'm having some problems which hopefully somebody can help with. Kamailio, the MySQL server and Siremis are running on the same server. The server is running Debian 5.0.4 I have followed the Siremis installation instructions at the following link: http://siremis.asipto.com/install/ I reach a point were I can get to the Siremis web interface login page but when I enter the user name and password nothing happens. /var/log/apache2/access.log : ============================= 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/ HTTP/1.1" 302 271 "-" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/bin/ HTTP/1.1" 200 1083 "-" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" 192.168.1.101 - - [19/Jun/2010:15:53:22 +0200] "GET /siremis/js/sound.js HTTP/1.1" 404 270 "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" 192.168.1.101 - - [19/Jun/2010:15:53:23 +0200] "GET /siremis/js/sound.js HTTP/1.1" 404 270 "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" 192.168.1.101 - - [19/Jun/2010:15:53:38 +0200] "POST /siremis/bin/controller.php HTTP/1.1" 200 20 "http://192.168.1.14/siremis/bin/" "Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.9.2.3) Gecko/20100423 Ubuntu/10.04 (lucid) Firefox/3.6.3" /var/log/apache2/error.log : ============================ [Sat Jun 19 15:53:22 2010] [error] [client 192.168.1.101] File does not exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, referer: http://192.168.1.14/siremis/bin/ [Sat Jun 19 15:53:23 2010] [error] [client 192.168.1.101] File does not exist: /var/www/siremis-1.0.0/siremis-web/js/sound.js, referer: http://192.168.1.14/siremis/bin/

2 2

How do deal with a lost ACK and TM
by Geoffrey Mina 21 Jun '10

21 Jun '10

I am not 100% sure I fully understand my issue, but I think I'm on the right track. I have a situation where Asterisk will drop calls a few seconds after they are set up. What I believe is happening: a=asterisk k=kamailio k --> INVITE --> a a --> 100 TRYING --> k a --> 183 RINGING --> k a --> 200 OK --> k k --> ACK -->a (this packet is never received by asterisk) a --> 200 OK (retransmit) -->k a --> 200 OK (retransmit) -->k a --> 200 OK (retransmit) -->k a --> no response to critical packet - terminating call It doesn't appear that Kamailio is retransmitting the ACK. I would think that would be part of the TM module, but perhaps I am not using it properly. Is this possible or is there something else going on? Thanks.

3 2

Kamailio on Solaris(OpenSolaris)
by Daniel Cardoso Alves 21 Jun '10

21 Jun '10

Hi people, I am trying to install kamailio in a server with OpenSolaris (SunOS sun-test 5.11 snv_83 i86pc i386 i86pc), to make some tests, performance, capacity, etc. I tried to compile on Solaris but, it dont works perfectly. :) On the Internet and in the Manuals, I saw that there is a port from kamailio to solaris, but I dont find where to get this. Docs: INSTALL Supported architectures: - Linux/i386, Linux/amd64, Linux/armv4l, - FreeBSD/i386, OpenBSD/i386, NetBSD/sparc64 - Solaris/sparc64, Solaris/i386 So I have some questions: Is there a port to Solaris? Can I to compile Kamailio on Solaris (OpenSolaris)? Are there any manuals to install kamailio on Sun Machines? By the way, Can anybody tell me if the performance and capacity of kamailio on the solaris is better than linux? Thanks...

3 2

TM clarifications
by David 21 Jun '10

21 Jun '10

Hello, I am using Kamailio 1.5.4. I read RFC3261 section 17 and the TM doc, I had a few questions : Does t_relay() manage both the client and server transactions for the packet being forwarded ? Meaning it would absorb retransmissions it receives and would retransmit the relayed message when needed ? Do I need to call t_release() when I call t_relay () ? Where would I use t_checktrans(), t_release() and t_newtran() ? My understanding is that if I want to receive a packet and process it locally without forwarding it, I should use t_newtran(). This would be good for things such as PUBLISH, PRESENCE and REGISTER ( even though it's not a good idea ). So when do I release the transaction ? In the reply route ? In the same run as I call t_newtran() ? Does t_checktrans() absorb retransmissions that were started with t_relay and t_newtran? If you have any material that I could read that would help me better understand the TM module, please say so. Thanks, David

5 15

Paid Service
by Omar 21 Jun '10

21 Jun '10

2 1

No voice issue with NAT
by JinKevin 20 Jun '10

20 Jun '10

Hi, My kamailio and application server are with Public IPs so the system uses nathelper without RTPProxy. It works with the eyebeam behind the ADSL router, but I cannot hear any voice ( ethereal capture on the PC shows no RTP from the application server) when try it in a Hotel or with a 3G data card. Below is the SIP messages for the issue, is there anyway can fix the issuse? ============Start=========================== UAC ----> NAT/Firewall(FW) INVITE sip:0216666@210.x.y.15 SIP/2.0 Via: SIP/2.0/UDP 172.16.114.1:51828;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:2001@172.16.114.1:12461> To: "videoConf"<sip:0216666@210.x.y.15> From: "2001"<sip:2001@210.x.y.15>;tag=db06e72d Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 230 v=0 o=- 7 2 IN IP4 172.16.114.1 s=CounterPath eyeBeam 1.5 c=IN IP4 172.16.114.1 t=0 0 m=audio 27384 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv NAT/FW ---> Kamailio(KAM) INVITE sip:0216666@210.x.y.15 SIP/2.0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@113.108.Z.1:12302>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15> CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:2001@113.108.Z.1:13202> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 230 Content-Type: application/sdp v=0 o=- 7 2 IN IP4 172.16.114.1 s=CounterPath eyeBeam 1.5 c=IN IP4 113.108.Z.1 t=0 0 m=audio 14794 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv NAT/FW ---> Kamailio(KAM) INVITE sip:0216666@210.x.y.15 SIP/2.0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@113.108.Z.1:12302>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15> CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:2001@113.108.Z.1:13202> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 230 Content-Type: application/sdp v=0 o=- 7 2 IN IP4 172.16.114.1 s=CounterPath eyeBeam 1.5 c=IN IP4 113.108.Z.1 t=0 0 m=audio 14794 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.114.1:51828;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@210.x.y.15:5060>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15> CSeq: 1 INVITE Server: kamailio (3.0.2 (sparc64/solaris)) Content-Length: 0 KAM ---> AS INVITE sip:0216666@210.x.y.3:5060 SIP/2.0 Record-Route: <sip:210.x.y.15;lr=on;nat=yes> Via: SIP/2.0/UDP 210.x.y.15;branch=z9hG4bK316e.08d5.0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@113.108.Z.1:12302>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15> CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:2001@113.108.Z.1:12461> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 257 Content-Type: application/sdp v=0 o=- 7 2 IN IP4 113.108.Z.1 s=CounterPath eyeBeam 1.5 c=IN IP4 113.108.Z.1 t=0 0 m=audio 14794 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=oldmediaip:172.16.114.1 KAM ---> NAT/FW SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.114.1:51828;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Record-Route: <sip:210.x.y.15;lr;nat=yes> Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@210.x.y.15:5060>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 CSeq: 1 INVITE Allow: INVITE,ACK,BYE,CANCEL,PRACK,NOTIFY,INFO,OPTIONS,UPDATE Supported: 100rel,timer,precondition Contact: <sip:0216666@210.x.y.3:5060;transport=udp> Content-Length: 0 AS ---> KAM SIP/2.0 200 Ok Via: SIP/2.0/UDP 210.x.y.15;branch=z9hG4bK316e.08d5.0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Record-Route: <sip:210.x.y.15;lr=on;nat=yes> To: "videoConf" <sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 From: "2001" <sip:2001@113.108.Z.1:12302>;tag=db06e72d Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. CSeq: 1 INVITE Content-Length: 222 Content-Type: application/sdp Session-Expires: 1800;refresher=uas Require: timer Allow: INVITE,ACK,BYE,CANCEL,PRACK,NOTIFY,INFO,OPTIONS,UPDATE Supported: 100rel, timer,precondition Contact: <sip:0216666@210.x.y.3:5060;transport=udp> v=0 o=- 15638 1276800468 IN IP4 210.x.y.8 s=- c=IN IP4 210.x.y.11 t=0 0 m=audio 30654 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 KAM -----> NAT/FW SIP/2.0 200 Ok Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Record-Route: <sip:210.x.y.15;lr=on;nat=yes> To: "videoConf" <sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 From: "2001" <sip:2001@113.108.Z.1:12302>;tag=db06e72d Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. CSeq: 1 INVITE Content-Length: 222 Content-Type: application/sdp Session-Expires: 1800;refresher=uas Require: timer Allow: INVITE,ACK,BYE,CANCEL,PRACK,NOTIFY,INFO,OPTIONS,UPDATE Supported: 100rel, timer,precondition Contact: <sip:0216666@210.x.y.3:5060;transport=udp> v=0 o=- 15638 1276800468 IN IP4 210.x.y.8 s=- c=IN IP4 210.x.y.11 t=0 0 m=audio 30654 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 NAT/FW ----> UAC SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.114.1:51828;branch=z9hG4bK-d8754z-3870be05b936bf38-1---d8754z-;rport=12461 Record-Route: <sip:210.x.y.15;lr;nat=yes> Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@210.x.y.15:5060>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 CSeq: 1 INVITE Session-Expires: 1800;refresher=uas Require: timer Allow: INVITE,ACK,BYE,CANCEL,PRACK,NOTIFY,INFO,OPTIONS,UPDATE Supported: 100rel,timer,precondition Contact: <sip:0216666@210.x.y.3:5060;transport=udp> Content-Length: 222 Content-Type: application/sdp v=0 o=- 15638 1276800468 IN IP4 210.x.y.8 s=- c=IN IP4 210.x.y.11 t=0 0 m=audio 30654 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 UAC ---> NAT/FW ACK sip:0216666@210.x.y.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.114.1:51828;branch=z9hG4bK-d8754z-ff50226f4e20b91c-1---d8754z-;rport Max-Forwards: 70 Route: <sip:210.x.y.15;lr;nat=yes> Contact: <sip:2001@172.16.114.1:12461> To: "videoConf"<sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 From: "2001"<sip:2001@210.x.y.15:5060>;tag=db06e72d Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. CSeq: 1 ACK User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 NAT/FW ----> KAM ACK sip:0216666@210.x.y.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-ff50226f4e20b91c-1---d8754z-;rport Route: <sip:210.x.y.15;lr;nat=yes> Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@113.108.Z.1:12302>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2001@113.108.Z.1:13202> User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 KAM ----> AS ACK sip:0216666@210.x.y.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 210.x.y.15;branch=0 Via: SIP/2.0/UDP 113.108.Z.1:13945;branch=z9hG4bK-d8754z-ff50226f4e20b91c-1---d8754z-;rport=12461 Call-ID: NzYwMzhjOTc5NWJlY2Q5YzFiN2JlZGNjNDlkNzMyNmU. From: "2001"<sip:2001@113.108.Z.1:12302>;tag=db06e72d To: "videoConf"<sip:0216666@210.x.y.15>;tag=a72e08e3-09d3-4cd2-a0d2-d33113440310 CSeq: 1 ACK Max-Forwards: 69 Contact: <sip:2001@113.108.Z.1:12461> User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ==============End================ Thanks, Kevin _________________________________________________________________ 想知道明天天气如何？必应告诉你！ http://cn.bing.com/search?q=%E5%A4%A9%E6%B0%94%E9%A2%84%E6%8A%A5&form=MICHJ2

2 2

Tracking ITSP Trunk Utilization
by Geoffrey Mina 19 Jun '10

19 Jun '10

Greetings, I have a challenge which I can't imagine is unique to my situation. I have agreements with multiple ITSP/SIP providers for inbound and outbound use cases. I am trying to figure out a way to track and monitor - on a per trunk basis - statistics about open dialogs. In the end, I would like to have nice graphs that show my usage throughout the day... and most importantly, my max open dialogs so I can determine if I am coming close to my max sessions with any particular carrier. Does anyone know of a tool which could help me track this? It looks like Kamailio has some SNMP modules, perhaps SNMP + Cacti might be the common solution? Any input is greatly appreciated. Thanks.

3 2

t_on_failure to named failure route
by Steven C. Blair 18 Jun '10

18 Jun '10

I'm attempting to run an existing SER config file under a recent build of sip-router. I get a syntax error on the t_on_failure("noroute") call that exists in my config file. The specific error is "bad expression: type mismatch (str instead of int)". I presume this means sip-router does not support named failure route labels. Is that correct? Thanks,Steve --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax)

6 10

Functions to parse URI parameters
by Uriel Rozenbaum 18 Jun '10

18 Jun '10

Hey Guys, Is there any module/function to parse and assign to AVPs parameters from RPID? (in kamailio 1.5.3) I'm using $(hdr(Remote-Party-ID){tobody.params}) to get the parameters but maybe there's something already implemented to get privacy, screen, etc. I'll be using this with P-Asserted-Identity as well. Thanks in advance! Uriel

1 0

kamailio dialplan
by maybelater 18 Jun '10

18 Jun '10

Hi there, I am pretty new to all this (kamailio..) I am trying to set up the following: 1. Kamailio with web-gui -> done 2. Asterisk / elastix for connecting to sip-provider / PSTN-> done This works perfectly incoming / outgoing 3. Connecting elastix/asterix to kamailio -> done I already have extensions configured on kamailio. They can call each other perfectly. Now I do need a route-config or dialplan that routes ALL outgoing calls for all extensions to the Extension of the asterisk/elastix-machine. Is that possible and if so, how ? I do need to provision more than a few hundred sip-subscribers in the near future so connecting Them using just asterisk is not an option.. Thanks in advance, I really appreciate your help Regards lutz

2 3

'UDP' is not a valid transport
by Information 18 Jun '10

18 Jun '10

hi all, I am using Twinkle on fedora 11 as a client. it connected to the server fine but after some time it started giving on register as well as de-register. [Jun 18 15:15:14] ERROR[14964]: chan_sip.c:12215 register_verify: 'UDP' is not a valid transport for '70003'. we only use 'UNKNOWN'! ending call. [Jun 18 15:15:14] NOTICE[14964]: chan_sip.c:20561 handle_request_register: Registration from '"70003" <sip:70003@a.a.a.a>' failed for 'x.x.x.x' - Device not configured to use this transport type what could be the reason and the solution? regards Siddharth ***********************Confidentiality Notice*************************** The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Divinet or info(a)divinetaccess.com immediately and destroy all copies of this message and any attachments. Please print this message only if extremely necessary. Plant a tree today. Save the EARTH. ************************************************************************

2 2

Re: [SR-Users] UAC, parsing auth header failed
by Iñaki Baz Castillo 18 Jun '10

18 Jun '10

2010/6/18 Stefan Sayer <stefan.sayer(a)googlemail.com>: > but is that also true if the "Digest" appears as is in the ABNF? > > just curious Yes, there is no special reason for it. RFC 3261 - 25.1 ------------------------- Many SIP header field values consist of words separated by LWS or special characters. Unless otherwise stated, tokens are case- insensitive. ------------------------- As all we know the schema in a SIP/SIPS URI is case insensitive ("sip:" == "SIP:"): SIP-URI = "sip:" [ userinfo ] hostport uri-parameters [ headers ] However SIP methods are case sensitive ("INVITE" != "invite"): INVITEm = %x49.4E.56.49.54.45 ; INVITE in caps So *any* string appearing in SIP BNF (as "Digest", "MD5", "tag"....) are case insensitive. PS: /Ragel-SIP-Parser/sip_parser.sh Authorization 'Authorization: DIGEST username="Alice", realm="atlanta.com", nonce="84a4cc6f3082121f32b42a2187831a9e", response="7587245234b3434cc3412213e5f113a5"' OK ;) -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

uac_auth()
by Alex Balashov 16 Jun '10

16 Jun '10

Greetings, When I use uac_auth() in a failure route in this manner: if(t_check_status("407")) { uac_auth(); append_branch(); t_relay(); } or the reverse order vis-a-vis uac_auth() and append_branch(), I do not see any digest authentication headers being inserted into the retransmitted INVITE request. Also, CSeq is not being incremented. This is Kamailio 3.0.2. Please advise. -- 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 5

Re: [SR-Users] sipp to kamailio doesn't match ACK
by JR Richardson 16 Jun '10

16 Jun '10

> Date: Tue, 15 Jun 2010 16:57:37 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] sipp to kamailio doesn't match ACK > To: sr-users(a)lists.sip-router.org > Message-ID: <4C17E941.4020506(a)evaristesys.com> > Content-Type: text/plain; charset=UTF-8; format=flowed > > 1) AFAIK, the Record-Route/Route headers are used by TM for maintaining > transaction state, unless you are relying on the 'dialog' module's "fast > matching" cookies, which are identifiers that get placed as a URI > parameter into the Record-Route header so that it is not necessary to > match the message by more conventional means (Via/branch, Call-ID, tags, > etc.) > > 2) ACK is a separate transaction; every request and reply sequence is a > transaction. So, if you're trying to match it to an existing INVITE > transaction, for instance, that won't happen. > > t_check_trans() has special behaviour for ACKs, depending on whether > they're end-to-end ACKs or ACKs for negative replies to existing > transactions: > > http://www.kamailio.org/docs/modules/3.0.x/modules/tm.html#t_check_trans > > Are you sure that you're using it, or its newer variant, > t_lookup_request(), correctly? > > 3) You are, nevertheless, correct that the ACK should contain the RR > header: > > The UAC core MUST generate an ACK request for each 2xx received from > the transaction layer. The header fields of the ACK are constructed > in the same way as for any request sent within a dialog (see Section > 12) with the exception of the CSeq and the header fields related to > authentication. > > -- Alex > Thanks for the feedback Alex. JR

1 0

Can't build kamailio on fresh debian squeezy
by Dmitri Korotkov 16 Jun '10

16 Jun '10

Hello all, After debian squeeze upgrade (as well as in new squeeze installation) it is impossible to build kamailio. All dep. packages were installed and I successfully done this task before. Does anybody have same problem ? Is there any solution ? Part of the build log is shown below. BR, Dmitri devel-test:/usr/src/kamailio#make deb ~ ~ ~ /usr/bin/make every-module group_include="kpurple" Makefile.defs defs skipped make[2]: Entering directory `/usr/src/kamailio' Makefile.defs defs skipped make[3]: Entering directory `/usr/src/kamailio/modules_k/purple' make[3]: Leaving directory `/usr/src/kamailio/modules_k/purple' Makefile.defs defs skipped make[3]: Entering directory `/usr/src/kamailio/modules_k/purple' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.4"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=3000002 -DCFG_DIR='"/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/usr/lib/glib-2.0/include -I/usr/include/glib-2.0 -I/usr/include/libxml2 -DOPENSER_MOD_INTERFACE -DMOD_NAME='"purple"' -c clientaccount.c -o clientaccount.o -MMD -MP gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.4"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=3000002 -DCFG_DIR='"/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/usr/lib/glib-2.0/include -I/usr/include/glib-2.0 -I/usr/include/libxml2 -DOPENSER_MOD_INTERFACE -DMOD_NAME='"purple"' -c clientops.c -o clientops.o -MMD -MP gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -mtune=opteron -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.4.4"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=3000002 -DCFG_DIR='"/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/usr/lib/glib-2.0/include -I/usr/include/glib-2.0 -I/usr/include/libxml2 -DOPENSER_MOD_INTERFACE -DMOD_NAME='"purple"' -c clientpipe.c -o clientpipe.o -MMD -MP In file included from purple.h:23, from clientpipe.c:27: /usr/include/libpurple/status.h:93: error: expected specifier-qualifier-list before 'gpointer' make[3]: *** [clientpipe.o] Error 1 make[3]: Leaving directory `/usr/src/kamailio/modules_k/purple' make[2]: *** [modules_k] Error 1 make[2]: Leaving directory `/usr/src/kamailio' make[1]: *** [build-stamp] Error 2 make[1]: Leaving directory `/usr/src/kamailio' dpkg-buildpackage: error: debian/rules build gave error exit status 2 devel-test:/usr/src/kamailio#

3 2

Help need on NAT_Traversal
by JinKevin 16 Jun '10

16 Jun '10

Hi All, I'm building a video conference system with the kamailio and the video AS. The network is as below: UAC(behind NAT) ------ kamailio(public IP) ------ AS (public IP) I have setup the basic kamailio and the SIP signaling can reach the AS and the call can be established. Now the time to setup the NAT traveral since the UACs are behind the NAT. Since it's video scenario, I don't want the kamailio as the RTPproxy so try to use the NAT_Traversal module. >From the module doc, NAT_Traversal needs the Dialog module as well. Have no idea on how to load these two modules and the route config required although have readed the module docs. Wondering if someone can help on the config for this scenario? All call with $rU=="0216666" is forwarded to the video AS as the route[CONF]. Thanks in advance! Thanks, Kevin Below is the current config of the kamailio server: ==============CFG============= AppSer01:root@/usr/local/kamailio-3.0/etc/kamailio$ cat kamailio.cfg #!KAMAILIO #!define WITH_DEBUG #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # $Id$ # # Kamailio (OpenSER) SIP Server v3.0 - basic configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <users(a)lists.kamailio.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 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 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 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no port=5060 /* 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:210.13.124.15:5060 ####### 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 # #!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 module path mpath="/usr/local/kamailio-3.0/lib/kamailio/modules_k/:/usr/local/kamailio-3.0/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" #!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 # ----------------- setting module-specific parameters --------------- # ----- 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", 0) # ----- 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", "") # ----- 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://openser:openK2mrw23@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openK2mrw23@localhost/openser") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openK2mrw23@localhost/openser") modparam("auth_db", "load_credentials", "") #!endif # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openK2mrw23@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openK2mrw23@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://openser:openK2mrw23@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ 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; } # NAT detection route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... 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() ) { # non 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.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # 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(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { 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(); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); route(CONF); # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); 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 setflag(2); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag("6"); } if (isflagset(5) || isbflagset("6")) { 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; } # Presence server route route[PRESENCE] { #!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("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "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 } } #!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(5); } #!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; } # 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; } # add by KJ --start route[CONF] { if(is_method("INVITE") && $rU=="0216666") { rewritehostport("210.x.y.z:5060"); } route(RELAY); exit; } # add by KJ --end # Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } AppSer01:root@/usr/local/kamailio-3.0/etc/kamailio$ _________________________________________________________________ MSN十年回馈，每位用户可免费获得价值25元的卡巴斯基反病毒软件2010激活码，快来领取！ http://kaba.msn.com.cn/?k=1

2 2

Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by JR Richardson 15 Jun '10

15 Jun '10

On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > Hi All, > > I have been in the lab testing various load balancing scenarios with > kamailio 1.5 and 3.0 and distributing calls to asterisk servers. I am > running into an issue that I'm sure is related to sipp not kamailio or > asterisk. My testing is setup like this: > sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk servers > > My sipp scenario: > sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1 > -trace_err -m 1 -nr > > What I'm seeing in my ngrep sip traces is the ACK coming back from > sipp does not have the RR header or a proper ftag: > > Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97: > > U 10.10.12.22:5060 -> 10.10.14.97:5061 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0. > Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>. > From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001. > To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7. > Call-ID: 1-10002(a)10.10.14.97. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:65@10.10.14.102>. > Content-Type: application/sdp. > Content-Length: 184. > . > v=0. > o=root 23641 23641 IN IP4 10.10.14.102. > s=session. > c=IN IP4 10.10.14.102. > t=0 0. > m=audio 24270 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22: > > U 10.10.14.97:5061 -> 10.10.12.22:5060 > ACK sip:22265@10.10.12.22:5060 SIP/2.0. > Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5. > From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001. > To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7. > Call-ID: 1-10002(a)10.10.14.97. > CSeq: 1 ACK. > Contact: sip:sipp@10.10.14.97:5061. > Max-Forwards: 70. > Subject: Performance Test. > Content-Length: 0. > > And the ACK goes into the to-tag portion of the kamailio route script > but can not match the transaction so kamailio exits, the dialog drops > out of memory and when sipp sends a BYE, kamailio replys with 404 not > found. > > So if I'm diagnosing this correctly, sipp is not maintaining the > Record-Route: header in the responses back to kamailio and without > that info kamailio can not maintain transaction state and the call > fails. Is there any work around or possibly another SIP performance > testing suite that will handle RR headers? > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > After reassuring myself that my diagnosis was correct and shortly before I went blind tracing sip messages, I dug into sipp and found the ability to create a new scenario adding the record route header and next_url parameter. So now when I run the testing from sipp, the dialogs are properly processed statefully through kamailio and all calls complete as expected. I though I new SIP pretty well, but it's the problems I run into that convince me there is so much more to learn. Thanks. JR -- JR Richardson Engineering for the Masses

1 0

sipp to kamailio doesn't match ACK
by JR Richardson 15 Jun '10

15 Jun '10

Hi All, I have been in the lab testing various load balancing scenarios with kamailio 1.5 and 3.0 and distributing calls to asterisk servers. I am running into an issue that I'm sure is related to sipp not kamailio or asterisk. My testing is setup like this: sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk servers My sipp scenario: sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1 -trace_err -m 1 -nr What I'm seeing in my ngrep sip traces is the ACK coming back from sipp does not have the RR header or a proper ftag: Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97: U 10.10.12.22:5060 -> 10.10.14.97:5061 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0. Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>. From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001. To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7. Call-ID: 1-10002(a)10.10.14.97. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Contact: <sip:65@10.10.14.102>. Content-Type: application/sdp. Content-Length: 184. . v=0. o=root 23641 23641 IN IP4 10.10.14.102. s=session. c=IN IP4 10.10.14.102. t=0 0. m=audio 24270 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22: U 10.10.14.97:5061 -> 10.10.12.22:5060 ACK sip:22265@10.10.12.22:5060 SIP/2.0. Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5. From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001. To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7. Call-ID: 1-10002(a)10.10.14.97. CSeq: 1 ACK. Contact: sip:sipp@10.10.14.97:5061. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. And the ACK goes into the to-tag portion of the kamailio route script but can not match the transaction so kamailio exits, the dialog drops out of memory and when sipp sends a BYE, kamailio replys with 404 not found. So if I'm diagnosing this correctly, sipp is not maintaining the Record-Route: header in the responses back to kamailio and without that info kamailio can not maintain transaction state and the call fails. Is there any work around or possibly another SIP performance testing suite that will handle RR headers? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

auth_radius - Segmentation fault
by Dmitry Kirillov 14 Jun '10

14 Jun '10

Hi I have build SIP-Router and modules from sources from GIT repository (development branch). auth_radius module results in a "Segmentation fault: 11" when I try to register (radius_www_authorize). sercmd> core.version Server: ser (3.0.99-dev1 (i386/freebsd)) eabca5 Any help for solving the problem would be appreciated. Thanks. -- Dmitry Kirillov, System Engineer KDS-RIPE Network Operational Center, JSC Neotelecom Tomsk, Russia Telephone: +7(3822)710-008, ICQ: 22413725 Web: http://www.neotelecom.ru/ | mailto:noc@neotelecom.ru Looking Glass: http://lg.neotelecom.ru/ __________ Information from ESET NOD32 Antivirus, version of virus signature database 5192 (20100612) __________ The message was checked by ESET NOD32 Antivirus. http://www.esetnod32.ru/.ml

6 17

SER multidomain
by jessica Achkar 14 Jun '10

14 Jun '10

Hello, I need to configure my SER to support multidomain. I am working on solaris 10 and on the SER bundled version of SER. I have the domain module but i don't have the mediaproxy module. When i type serctl help the serctl add domain isn't included so what should i modify in my ser.cfg file. Thank you _________________________________________________________________ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969

2 1

Kphone
by Saz 14 Jun '10

14 Jun '10

hii Can anyone help me with Kphone-vic installation I will really appreciate for any documentation or manuals. Thanks in advance.

3 4

install: kamctlrc was not found anywhere!
by JinKevin 14 Jun '10

14 Jun '10

While running the[ gmake prefix="/usr/local/kamailio-3.0" install ] on solaris 10 sparc got the issue below, please help. Thanks, Kevin ============= Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/utils/sercmd' gmake[1]: `sercmd' is up to date. gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/utils/sercmd' Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/utils/kamctl' # kamctl config touch /usr/local/kamailio-3.0/etc/kamailio//kamctlrc.sample ginstall -m 644 kamctlrc \ /usr/local/kamailio-3.0/etc/kamailio//kamctlrc.sample find: cannot follow symbolic link /etc/init.d/ipunity: No such file or directory find: cannot follow symbolic link /etc/rc3.d/K02ipunity: No such file or directory find: cycle detected for /lib/32/ find: cycle detected for /lib/secure/32/ find: cycle detected for /usr/lib/locale/common/32/ find: cycle detected for /usr/lib/locale/common/LC_CTYPE/32/ find: cycle detected for /usr/lib/locale/common/LO_LTYPE/32/ find: cannot follow symbolic link /usr/lib/locale/es_MX.UTF-8/LC_MESSAGES: No such file or directory find: cannot follow symbolic link /usr/lib/locale/fr_CA.UTF-8/LC_MESSAGES: No such file or directory find: cycle detected for /usr/lib/secure/32/ find: cannot follow symbolic link /usr/lib/sparcv9/llib-ladm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-laio.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lbsm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lc.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ldl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lcmd.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lcontract.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lctf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lcurses: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lcurses.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ldevice.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ldevid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ldevinfo.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ldoor.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lefi.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lelf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lgen.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lintl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lkstat.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lmd5.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lnsl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lnvpair.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lpam.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lposix4.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lpthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lrt.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lresolv.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lrtld_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lscf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lsec.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lsecdb.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lsendfile.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lsocket.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lsysevent.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltermcap: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltermcap.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltermlib: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltermlib.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lthread_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltsnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-ltsol.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lumem.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-luuid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lxnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/sparcv9/llib-lm.ln: No such file or directory find: cycle detected for /usr/lib/32/ find: cannot follow symbolic link /usr/lib/64/llib-ladm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-laio.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lbsm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lc.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ldl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lcmd.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lcontract.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lctf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lcurses: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lcurses.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ldevice.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ldevid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ldevinfo.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ldoor.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lefi.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lelf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lgen.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lintl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lkstat.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lmd5.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lnsl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lnvpair.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lpam.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lposix4.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lpthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lrt.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lresolv.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lrtld_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lscf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lsec.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lsecdb.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lsendfile.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lsocket.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lsysevent.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltermcap: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltermcap.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltermlib: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltermlib.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lthread_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltsnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-ltsol.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lumem.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-luuid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lxnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/64/llib-lm.ln: No such file or directory find: cycle detected for /usr/lib/lwp/32/ find: cycle detected for /usr/lib/link_audit/32/ find: cannot follow symbolic link /usr/lib/llib-ladm: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ladm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-laio: No such file or directory find: cannot follow symbolic link /usr/lib/llib-laio.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lbsm: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lbsm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lc: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lc.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcmd: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcmd.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcontract: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcontract.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lctf: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lctf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcurses: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lcurses.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevice: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevice.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevid: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevinfo: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldevinfo.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldl: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldoor: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lefi: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ldoor.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lefi.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lelf: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lelf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lgen: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lgen.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lintl: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lintl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lkstat: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lkstat.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lmd5: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lmd5.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lnsl: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lnsl.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lnvpair: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lnvpair.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lpam: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lpam.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lposix4: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lposix4.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lpthread: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lpthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lresolv: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lresolv.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lrt: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lrt.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lrtld_db: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lrtld_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lscf: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lscf.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsec: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsec.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsecdb: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsecdb.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsendfile: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsendfile.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsocket: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsocket.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsysevent: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lsysevent.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltermcap: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltermcap.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltermlib: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltermlib.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lthread: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lthread.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lthread_db: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lthread_db.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltsnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-ltsol.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lumem: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lumem.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-luuid: No such file or directory find: cannot follow symbolic link /usr/lib/llib-luuid.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lxnet: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lxnet.ln: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lm: No such file or directory find: cannot follow symbolic link /usr/lib/llib-lm.ln: No such file or directory find: cannot follow symbolic link /usr/lib/libldap-2.4.so.2: No such file or directory install: kamctlrc was not found anywhere! gmake[1]: *** [install-cfg] Error 2 gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/utils/kamctl' ERROR: gmake -C utils/kamctl install-if-newer failed gmake: *** [install-utils] Error 1 AppSer01:root@/usr/local/src/kamailio-3.0.0/kamailio$ cd utils/ /usr/local/src/kamailio-3.0.0/kamailio/utils AppSer01:root@/usr/local/src/kamailio-3.0.0/kamailio/utils$ ls db_berkeley db_oracle fifo_relay kamctl kamunix pdbt pike_top profile route_graph sercmd sipgrep AppSer01:root@/usr/local/src/kamailio-3.0.0/kamailio/utils$ cd kamctl/ /usr/local/src/kamailio-3.0.0/kamailio/utils/kamctl AppSer01:root@/usr/local/src/kamailio-3.0.0/kamailio/utils/kamctl$ ls db_berkeley kamctl.base kamctl.mysql kamctlrc kamdbctl.dbtext Makefile dbtext kamctl.ctlbase kamctl.oracle kamdbctl kamdbctl.mysql mysql dbtextdb kamctl.db_berkeley kamctl.pgsql kamdbctl.8 kamdbctl.oracle oracle kamctl kamctl.dbtext kamctl.sqlbase kamdbctl.base kamdbctl.pgsql postgres kamctl.8 kamctl.fifo kamctl.unixsock kamdbctl.db_berkeley kamdbfunc.oracle _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

2 1

Fwd: [RAI] SIPit26 summary
by Klaus Darilion 14 Jun '10

14 Jun '10

FYI - the summary from SIPit regards Klaus -------- Original-Nachricht -------- Betreff: [RAI] SIPit26 summary Datum: Fri, 4 Jun 2010 13:50:50 -0500 Von: Robert Sparks <rjsparks(a)nostrum.com> An: rai(a)ietf.org SIPit 26 was hosted by Edvina and TANDBERG in Kista, Sweden the week of May 17-21, 2010. There were 67 attendees from 28 companies visiting from 15 countries. We had 42 distinct implementations. There was a huge upswing in the number of TLS and SRTP implementations at this event (and IPv6 support continues to grow steadily). There was also a large upswing in ICE and TURN implementations. The roles represented (some implementations act in more than one role) 29 endpoints 13 proxy/registrars/b2bua/sbcs 5 dedicated event servers I focused more on helping with the multiparty tests and specification discussions and did not review each survey response with its team as closely at this event as I have in the past few. As a result, please be cautious of a little more uncertainty in these values. Implementations using each transport for SIP messages: UDP 98% TCP 98% TLS 81% server-auth, 57% mutual-auth SCTP 21% DTLS 0% Just over half of the implementations present supported IPv6. There were three Identity implementations present For DNS we had support for: Full RFC3263 : 86% SRV only : 5% A records only : 2% no DNS support : 7% Support for various items in the endpoints 55% replaces 38% 3489stun 31% ice 24% gruu 21% sip/stun multiplexing 21% turn 17% diversion 17% history-info 17% path 17% ipsec 17% sigcomp 15% 5389stun 14% service-route 10% join Support for various items in the proxies: 38% path 23% service-route 15% sigcomp 31% ipsec 31% gruu 15% sip/stun multiplexing 31% diversion 0% history-info There were 2 MSRP and 2 XCAP implementations (one with xcap-diff support) present. The endpoints and B2BUAs implemented these methods: 100% INVITE, CANCEL, ACK, BYE 90% REGISTER 90% OPTIONS 84% REFER 79% NOTIFY 79% UPDATE 69% INFO 66% SUBSCRIBE 55% PRACK 52% MESSAGE 48% PUBLISH 79% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). 4 of the implementations required the other party to use symmetric-rtp. 72% of the UAs present both sent RTCP and paid attention to RTCP they received 55% of the endpoints present supported SRTP - all were using sdes. There were no dtls-srtp implementations at this SIPit. 45% of the endpoints supported multipart/mime. There was one implementation present with S/MIME support. There were 10 SIP Event Server implementations (not counting endpoints that support events only for refer). There were 13 SIP Event Client implementations (again, not counting endpoints that only support events for refer). These event packages were supported: Server Client 8 11 presence 4 7 presence.winfo 4 8 message-summary 5 6 dialog 0 3 reg 1 2 conference 1 1 xcap-diff 1 1 kpml 0 1 ua-profile 0 1 reg-gruu These packages were supported with the eventlist extension: Server Client 3 1 presence There were two Info-Packages implementations present. One was using a proprietary payload type. The other was reusing a SIP-Events package. Six of the proxies present still rely only on max-forwards. There were three implementations of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Reports contributed by Mark Thompson, Eoin McLeod, and Olle Johansson) * Forking Teams coalesced quickly into a working topology that was successful in having all forked targets ringing in parallel. With different UAs injecting the call, a number of different implementation issues arose, including: some UAs could not process sdp offers with multiple instances of the same media type in (e.g. 1x Audio, 2x Video, 2x Application). When stripped back to 1x Audio, 2x Video the situation improved, but even then some would only ring if there was at most one m-line of each type. some UAs could not handle messages greater than 2048 bytes (sdp payload and SIP header overhead). Rather than returning 513 Message Too Large, the UAs in question would silently discard the request. one UA implementation present elected to BYE early dialogs for forks that they do not wish to be answered having successfully established at least one confirmed dialog. The dialog-stateful outbound proxy in this test successfully survived the race condition by failing with a 481 Call/Transaction Does Not Exist response. one UA implementation incorrectly assumed that the To header could be used for request targeting rather than the Request-URI. This assumption led to them not being able to participate as the recipient of re-targeted call forks. one proxy implementation present, when configured with both IPv4 and IPv6 DNS servers to query, would stall request forwarding until all DNS requests had been resolved. Where one of the configured servers was unavailable or slow to respond, the entire call setup would suffer significant latency despite there being sufficient candidates available to route toward. The test procedure was then updated so that the multiple answer edge condition could be tested: the link between the top-most forking proxy and its downstream forks was disconnected whilst all signaled UAs were instructed to answer. Upon reconnecting the network, the forking proxy would receive multiple 200 OK responses to the initial request. most UA implementations placed as originators of the call were shown to successfully ACK-BYE the undesired alternative answers whilst maintaining the first responder's confirmed dialog. at least one UA implementation did not respond to subsequent 200 OK responses establishing confirmed dialogs, instead discarding them as if they were not received resulting in re-transmits and eventual timeout. * Loops/Spirals Implementations all behaved well during the basic loops and spirals tests using multiple protocols on IPv4 only. When we introduced IPv6 into some hops, several proxy and endpoint implementations would not recognize messages as valid when they failed to parse the IPv6 addresses in Vias or Record-Routes that should have just been opaque bits to them. We constructed a loop with 3 of the participating B2BUA implementations and discovered that none of them were decrementing Max-Forwards (some were resetting the value to 70) or performing any other type of loop-detection. * SRTP There were a much larger number of SRTP implementations present at this event, and many scenarios were exercised with success. Some of the notable problems encountered included a few implementations that were using SDES without TLS and continued arguments around how to implement sips/how to indicate one hop of SIP over TLS. Individal testing between the peers exercised wrapping and hold/resume. The next event's test will focus more closely on scenarios like those, and streams utilizing multiple SSRCs * STUN/TURN/ICE This test was very well attended. Endpoints successfully negotiatied sessions utilizing both server and peer reflexive candidates. Connectivity was established between UAs behind different NATs that were using the same private address ranges. We exercised connectivity through TURN chosen using ICE. When older SDP mangling elements were introduced, fewer setup attempts were successful - some implementations were not recognizing ICE mismatches. Several implementations stopped sending keepalives when streams were put on hold. Some implementations restarted ICE when coming off hold. * TLS As noted earlier, there was a large upswing in the number of implementations utilizing TLS present. Configuration of endpoints (establishing trusted CAs and installing certificates) is still very slow, but once the endpoints were configured, iteroperability of SIP over TLS was very high. (However, see the related note on the SRTP test). The automated tests started at SIPit 25 continue to be refined and will soon be available for tests between SIPit events. * Early Media The early media test was also well attended, and the automated tests for early media scenarios continue to be refined. We tested handling early media in forking situations (one branch sends early media, another branch answers) and taking early media on and off hold. The early media test was developed as a self-test and will be available for tests between SIPit events. The test involved forking the call to multiple servers sending early media in the same dialog. The participants argued that early media for ring tones could and propably should be separated from early media for operator messages by using 180 with SDP for ring tones and 183 with SDP for other messages. This way, a UA can decide to stop listening to the 180 ring tone if an 183 message is sent. UAs showed very different behaviour, from jumping between the various streams to mixing them. * Presence The multiparty presence tests successfully exercised basic inter-domain presence exchange, including exchange of winfo information. The participants tried to set up a back-end-subscription loop between two RLSes, but configuration issues got in the way. We will try to exercise that scenario early at the next event. * IPv6 The IPv6 multiparty was set up in order to run various calls through gateways between IPv4 and IPv6, as well as to discuss and setup permanent self-tests for UAs and proxys. IPv4 UAs successfully placed calls to IPv6 UAs through servers that handled the gateway situation. IPv4 UAs mostly had a hard time handling getting SDPs or Contact: headers with IPv6 addresses when placing calls through a proxy that supported dual stack. Successful sessions were established on an IPv6 only networkA lot of experience was gained and we will continue to refine these tests on following SIPits _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

3 2

Help with configuring SER to append a plus (+)
by Vu, Tuan N 14 Jun '10

14 Jun '10

T o whom it may concern, I am looking for an SER configuration that change the number to be a global number Incoming to SER - From: <sip:[telephone number]@ <hosts> SER changes it to - From:sip:+[telephone number]@ <host<sip:-[telephone%20number]@%20%3chost>> before sending it on. Thanks, Tuan Vu, THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the e-mail and its attachments from all computers

2 1

ser problem
by Maciej Olejnik 14 Jun '10

14 Jun '10

Hi, I just installed SER on my server, I'm having problem, which I can't solve Despite mysql server works fine, when I try to register to SIP server i can't manage. I see in log this: Jun 11 12:26:21 elka /usr/local/sbin/ser[23474]: submit_query(): MySQL server has gone away Jun 11 12:26:21 elka /usr/local/sbin/ser[23474]: submit_query(): Error while submitting query Jun 11 12:26:21 elka /usr/local/sbin/ser[23474]: get_ha1(): Error while querying database What is the solution? Maciej

2 1

Re: [SR-Users] play early media in kamailio
by dotnetdub 14 Jun '10

14 Jun '10

On 14 June 2010 00:16, zhou tianjun <zhoutj(a)gmail.com> wrote: > Hi Stephen, > thanks for your rapid reply. > I use kamailio as SIP proxy server . when ua1 is calling to ua2, > it will receive early media from kamailio while ua2 is busy(send > 486 reply code). > > All sip messages pass kamailio, ua1 and ua2 are not natted cliend. > and i want not to use any media server such as sems etc. > > > Hi Zhou, > > You will need to use something like rtpproxy - > > There is a thread discussing it here: > > http://www.mail-archive.com/users@lists.kamailio.org/msg06532.html > > HTH. > > Regards, > Stephen >

1 0

Amoocon 2010 presentations
by Daniel-Constantin Mierla 14 Jun '10

14 Jun '10

Hello, the slides of the two presentations I gave at Amoocon 2010 are available online on event's site: The SIP Router Project - Asynchronous SIP Routing - http://bit.ly/9Lm83T Do your SIP routing in Lua! - http://bit.ly/c2uRYM First talk went through the presentation of the project, features in 3.0 and devel versions, ending with short description of offered capabilities for asynchronous processing. The second one focused on Lua API that can be used to do SIP routing, showing an example of asynchronous processing (as continuation to first talk): notifications to Twitter in case of missed calls using Lua API. Cheers, Daniel -- Daniel-Constantin Mierla Asterisk SIP Router Masterclass Malaga, Spain - July 5-9, 2010 http://www.asipto.com/

1 0

auth_radius - Segmentation fault
by Dmitry Kirillov 14 Jun '10

14 Jun '10

Hi I have build SIP-Router and modules from sources from GIT repository (development branch). auth_radius module results in a "Segmentation fault: 11" when I try to register (radius_www_authorize). sercmd> core.version Server: ser (3.0.99-dev1 (i386/freebsd)) eabca5 Any help for solving the problem would be appreciated. Thanks. -- Dmitry Kirillov, System Engineer KDS-RIPE Network Operational Center, JSC Neotelecom Tomsk, Russia Telephone: +7(3822)710-008, ICQ: 22413725 Web: http://www.neotelecom.ru/ | mailto:noc@neotelecom.ru Looking Glass: http://lg.neotelecom.ru/ __________ Information from ESET NOD32 Antivirus, version of virus signature database 5191 (20100611) __________ The message was checked by ESET NOD32 Antivirus. http://www.esetnod32.ru/.ml

2 1

play early media in kamailio
by zhou tianjun 13 Jun '10

13 Jun '10

HI,all Is kamailio.cfg can be configured to play early media using nathelper module with rtpproxy_stream2uac() function. my SIP application environment is in only one domain, no NAT. thanks in advance for any help.

3 2

(no subject)
by rajesh volluru 11 Jun '10

11 Jun '10

http://lbwalltr.webs.com

1 0

Connecting Two OpenSER machines
by SR-USER 11 Jun '10

11 Jun '10

Hi everyone. I have two machines running OpenSER, one machines runs OpenSER A and the other runs OpenSER B. Phone A registers in OpenSER A while Phone B registers in OpenSER B. I would like to allow Phone A call Phone B, but not the other way around. I am a newbie so sorry for my lack of knowledge. Would should be my configurations files for OpenSER A and B? Thanks in advanced

4 7

kamailio 3 and cpl - 483 too many hops
by Karl Harris 11 Jun '10

11 Jun '10

I am trying to incorporate a cpl script for testing purposes with kamailio 3 but have been met with failure with each try. Test scenario I have two sip phones registered to kamailio successfully (I can make phone calls between the two phones). The CPL XML is as follows: <?xml version="1.0" encoding="UTF-8"?> <cpl> <incoming > <location clear="yes" url="sip:1000@192.168.254.144<sip%3A1000(a)192.168.254.144>" > <log comment="cpl-log - just location" /> </location> </incoming> </cpl> This compiled successfully and is populating the cpl table in the openser database. When I call this from kamailio.cfg I get the 483 (Too many hops) error. If someone could spot an obvious error, I would appreciate it. The current kamailio.cfg file is as follows: ------------------ debug=8 log_stderror=yes 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 revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* 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:192.168.254.144:5060 ####### 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 # pstn.gw_ip = "" desc "PSTN GW Address" ####### Modules Section ######## #set module path mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" 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" #!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 # ----kh add------# loadmodule "cpl-c.so" # ----kh end -----# # ----------------- setting module-specific parameters --------------- # ----- 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", 0) # ----- 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", "") # ----- 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://openser:openserrw@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") #!endif # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif # ----- cpl -------- modparam("cpl-c","db_url","mysql://openser:openserrw@localhost/openser") modparam("cpl-c","cpl_dtd_file","/etc/kamailio/cpl-06.dtd") modparam("cpl-c","log_dir","/var/log/kamailio/cpl") # ----- cpl -------- ####### Routing Logic ######## route{ 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; } # NAT detection route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... 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() ) { # non 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.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # 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(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { 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(); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(PSTN); # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); 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 setflag(2); route(RELAY); } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag("6"); } if (isflagset(5) || isbflagset("6")) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); xlog("--kh-- before cpl call"); cpl_run_script("incoming", "force_stateful"); xlog("--kh-- after cpl call"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence server route route[PRESENCE] { #!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("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "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 } } #!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(5); } #!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; } # 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; } # 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(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[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } } -- Karl Harris

2 1

Re: [SR-Users] Fail to add user account
by JinKevin 11 Jun '10

11 Jun '10

Sorry, pls ingore my previous email. I just forgot to remove some echo command added by me before the if statement. From: kevin.jzh(a)hotmail.com To: klaus.mailinglists(a)pernau.at; sr-users(a)lists.sip-router.org Subject: RE: [SR-Users] Fail to add user account Date: Fri, 11 Jun 2010 07:57:47 +0800 Have changed the kamctl from if [ $? -eq 0 ] ; then minfo "user '$1' already exists" exit 1 fi to if [ $? -ne 0 ] ; then minfo "user '$1' already exists" exit 1 fi Now I can add user successfully. Seems it's a bug in the script in kamailio 3.0.2 Thanks, Kevin > Date: Thu, 10 Jun 2010 16:48:13 +0200 > From: klaus.mailinglists(a)pernau.at > To: kevin.jzh(a)hotmail.com > Subject: Re: [SR-Users] Fail to add user account > > I have no idea. > > Try supplying a domain too: > > kamctl add user(a)yourdomain.com passwd > > regards > klaus > > Am 10.06.2010 16:32, schrieb JinKevin: > > Thanks Klaus, > > > > After change the LAST_LINE in the kamctlrc, the tail error is going, but > > it still says user "xxxx" already exists. > > > > # tail tool > > LAST_LINE="tail -1" > > > > I find the script below returns "false" which makes the kamctl exit. > > > > # params: user > > # output: false if exists, true otherwise > > is_user() { > > set_user $1 > > QUERY="select count(*) from $SUB_TABLE where \ > > $SUBSCRIBER_COLUMN='$OSERUSER' and $REALM_COLUMN='$OSERDOMAIN';" > > CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` > > mdbg "is_user: user counter=$CNT" > > if [ "$CNT" = "0" ] ; then > > false > > else > > true > > fi > > } > > > > > > What could be the cause? > > > > Thanks, > > Kevin > > > > > > > Date: Thu, 10 Jun 2010 10:11:18 +0200 > > > From: klaus.mailinglists(a)pernau.at > > > To: kevin.jzh(a)hotmail.com > > > CC: sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] Fail to add user account > > > > > > > > > > > > Am 10.06.2010 03:12, schrieb JinKevin: > > > > AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 > > > > usage: tail [+/-[n][lbc][f]] [file] > > > > tail [+/-[n][l][r|f]] [file] > > > > INFO: user '2000' already exists > > > > > > > > > between your kamctl command and the error message (user '2000' already > > > exists) is an error produced by the tail command. > > > > > > Looks like kamctl uses tail in a way which is not supported on your OS. > > > > > > regards > > > klaus > > > > ------------------------------------------------------------------------ > > 更多热辣资讯尽在新版MSN首页！立刻访问！ <http://cn.msn.com/> 聊天+搜索+邮箱想要轻松出游,手机MSN帮你搞定! 立刻下载！ _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

1 0

Re: [SR-Users] Fail to add user account
by JinKevin 10 Jun '10

10 Jun '10

Have changed the kamctl from if [ $? -eq 0 ] ; then minfo "user '$1' already exists" exit 1 fi to if [ $? -ne 0 ] ; then minfo "user '$1' already exists" exit 1 fi Now I can add user successfully. Seems it's a bug in the script in kamailio 3.0.2 Thanks, Kevin > Date: Thu, 10 Jun 2010 16:48:13 +0200 > From: klaus.mailinglists(a)pernau.at > To: kevin.jzh(a)hotmail.com > Subject: Re: [SR-Users] Fail to add user account > > I have no idea. > > Try supplying a domain too: > > kamctl add user(a)yourdomain.com passwd > > regards > klaus > > Am 10.06.2010 16:32, schrieb JinKevin: > > Thanks Klaus, > > > > After change the LAST_LINE in the kamctlrc, the tail error is going, but > > it still says user "xxxx" already exists. > > > > # tail tool > > LAST_LINE="tail -1" > > > > I find the script below returns "false" which makes the kamctl exit. > > > > # params: user > > # output: false if exists, true otherwise > > is_user() { > > set_user $1 > > QUERY="select count(*) from $SUB_TABLE where \ > > $SUBSCRIBER_COLUMN='$OSERUSER' and $REALM_COLUMN='$OSERDOMAIN';" > > CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` > > mdbg "is_user: user counter=$CNT" > > if [ "$CNT" = "0" ] ; then > > false > > else > > true > > fi > > } > > > > > > What could be the cause? > > > > Thanks, > > Kevin > > > > > > > Date: Thu, 10 Jun 2010 10:11:18 +0200 > > > From: klaus.mailinglists(a)pernau.at > > > To: kevin.jzh(a)hotmail.com > > > CC: sr-users(a)lists.sip-router.org > > > Subject: Re: [SR-Users] Fail to add user account > > > > > > > > > > > > Am 10.06.2010 03:12, schrieb JinKevin: > > > > AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 > > > > usage: tail [+/-[n][lbc][f]] [file] > > > > tail [+/-[n][l][r|f]] [file] > > > > INFO: user '2000' already exists > > > > > > > > > between your kamctl command and the error message (user '2000' already > > > exists) is an error produced by the tail command. > > > > > > Looks like kamctl uses tail in a way which is not supported on your OS. > > > > > > regards > > > klaus > > > > ------------------------------------------------------------------------ > > 更多热辣资讯尽在新版MSN首页！立刻访问！ <http://cn.msn.com/> _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

1 0

Match ip source of response into failure_route
by inge 10 Jun '10

10 Jun '10

Hi all, I try to do something like this on my failure_route[1] : if (t_check_status("503")) { if (ip_src==A.B.C.D) { ... } } The goal, is to prevent a 503 from a PSTN gateway and try another one for terminating the call. But, I realize that "src_ip" is the IP source of SIP Request. What could be a possible way to match IP source of the 503 responses ? Regards, Adrien L.

3 4

Fail to add user account
by JinKevin 10 Jun '10

10 Jun '10

Hi All, Finally, the kamalio was installed on the Solaris 10 sparc and I can start it with the 'kamctl start', and it is running as showed below. However I cann't add user by using the kamctl, it always says the user already exists. And actually the susbciber table is still empty in mysql. Mysql is accessable with command "mysql -u openser -p ". What could be the issue? What are the log files I can check except the syslog? Didn't find an error in the syslog. ======= AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl add 2000 2000 usage: tail [+/-[n][lbc][f]] [file] tail [+/-[n][l][r|f]] [file] INFO: user '2000' already exists AppSer01:root@/usr/local/kamailio-3.0/sbin$ kamctl ps Process:: ID=0 PID=8888 Type=attendant Process:: ID=1 PID=8890 Type=udp receiver child=0 sock=210.x.x.x:5060 Process:: ID=2 PID=8891 Type=udp receiver child=1 sock=210.x.x.x:5060 Process:: ID=3 PID=8892 Type=udp receiver child=2 sock=210.x.x.x:5060 Process:: ID=4 PID=8893 Type=udp receiver child=3 sock=210.x.x.x:5060 Process:: ID=5 PID=8894 Type=slow timer Process:: ID=6 PID=8895 Type=timer Process:: ID=7 PID=8896 Type=MI FIFO Process:: ID=8 PID=8897 Type=ctl handler Process:: ID=9 PID=8898 Type=tcp receiver child=0 Process:: ID=10 PID=8899 Type=tcp receiver child=1 Process:: ID=11 PID=8905 Type=tcp receiver child=2 Process:: ID=12 PID=8906 Type=tcp receiver child=3 Process:: ID=13 PID=8907 Type=tcp main process AppSer01:root@/usr/local/kamailio-3.0/sbin$ ======= Thanks, Kevin _________________________________________________________________ SkyDrive电子画册，带你领略精彩照片，分享“美”时“美”刻! http://www.windowslive.cn/campaigns/e-magazine/ngmchina/?a=c

2 2

domain name
by Anders 10 Jun '10

10 Jun '10

Hi, I have a carrier who insists and have configured their system so they can receive INVITEs to their domain name only, and not the IP. I cannot configure the gw for anything but IPs, but thought that I might be able to replace the IP address in the cfg. So I tried with setting $dd = "domain.com"; but that's not working, - I guess $dd is read-only or something. Any ideas? Thanks!! //Anders

3 5

Re: [SR-Users] 2 Asterisk servers re-inviting around kamailio even with t_relay being used?
by JR Richardson 09 Jun '10

09 Jun '10

> If that's really your complete Kamailio config, it's missing a number > of critical checks and handlers, such as loose_route(), which handles > sequential (in-dialog) requests differently from initial requests due > to the presence of a Route: header and a To header tag. > > You can't just stick cr_route() in the main route { ... } block > without those canonical checks and expect everything to work as desired. > > The reason that the proxy is not seeing the sequential requests > (reinvites, BYE, etc.) is because you don't have a Record-Route > header, which tells the endpoints to relay sequential requests through > the proxy on the network and transport level. Add this to your > initial request route prior to (cr_route())[1]: > > if(!is_method("REGISTER|OPTIONS|MESSAGE")) > record_route(); > > That should take care of it. > > -- Alex > > [1] And ideally, add quite a few other things as well. See the > stock configuration file for more insight. Thanks Alex. I am testing this in the lab so I don't have a production config loaded, I am really focused on the carrierroute module and trying to figure that out so I'm running bare bones using the module example. I looked into the RR functions and read in the CR module that it invokes the TM when necessary so I was thinking it would take care of keeping track of dialogs. The thing that threw me off was the handling of the scenario 1 session with sipp was ok but with asterisk scenario 2 it was not. So I loaded in the record_route function in the main routing block and it did take care of the complete session with the asterisk server, so that problem is solved. What I am running into now is the failure route. If the initial session can not compete to the fist carrierroute, the call switches over to the failure-route[1] which invokes carrierroute again with another domain. Then the receiving asterisk server retransmits the initial OK 6 times, so the transaction is lost or not held in kamailio state, then the whole session drops after a few more seconds. I think there is something going on with the append_branch function that is throwing me off. It is needed so the failure route completes but it's not handling the complete dialog or I could be totally off. Thanks. JR -- JR Richardson Engineering for the Masses

1 0

Kamailio 3.0.1 and NAT
by Information 09 Jun '10

09 Jun '10

hi, I have configured kamailio 3.0.1 on fedora 12 and its working fine for ips on the network. But I am unable to get a working configuration for a natted client. If one client is natted and other is not, both can call each other through the kamailio server but only the natted client sends voice to the un-natted client while on tcpdump i found that the un-natted client is actually seeing the private ip of the natted client and trying to deliver packets there. So no sound reaches the natted client from the un-natted client. I have been searching for some days for some help in this regard but have not found anything i can use. help, regards. ***********************Confidentiality Notice*************************** The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Divinet or info(a)divinetaccess.com immediately and destroy all copies of this message and any attachments. Please print this message only if extremely necessary. Plant a tree today. Save the EARTH. ************************************************************************

2 3

OT: phones which support out-of-dialog REFER
by Klaus Darilion 09 Jun '10

09 Jun '10

Hi! Does somebody know any SIP clients (hard/soft) which support out-of-dialog REFERs? thanks klaus

3 3

2 Asterisk servers re-inviting around kamailio even with t_relay being used?
by JR Richardson 08 Jun '10

08 Jun '10

Hi All, I'm using kamailio with carrierroute to load balance calls to other servers. I have 2 testing scenarios set up: sipp><kamailio><asterisk server In the above scenario all SIP transactions, dialogs go through kamailio, INVITE, Trying, ACK, OK, BYE, OK Here is another one: phone><asterisk><kamailio><asterisk server In this scenario only the initial INVITE, ACK and OK go through kamailio, then the 2 asterisk servers finish the session directly to each other with ACK, BYE and OK In both asterisk servers, canreinvite=no is set in peers and general section in sip.conf. The kamailio cfg is using t_relay. No errors are coming up anywhere. Here is a pastebin of the 2 asterisk traces and the kamailio config. http://pastebin.com/Uk9qVhX2 Any ideas on why the call is not maintaining SIP session through the proxy? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Strange behavior when rewriting "To" header from failure_route (kamailio - 3.0.2)
by Kosilov Fedor 08 Jun '10

08 Jun '10

piece of code: ........ failure_route[2] { if (!t_is_canceled()) { revert_uri(); sql_query("main", "select username from subscriber where alias='$tU' and alias_seq='2'", "alias_name"); sql_query("main", "select domain from subscriber where alias='$tU' and alias_seq='2'", "alias_domain"); subst_uri('/^sip:.*@.*$/sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])/'); subst('/^To:.*<sip:.*>/To: <sip:$dbr(alias_name=>[0,0])@$dbr(alias_domain=>[0,0])>\r/'); #<-----THIS ONE DOESN'T WORK PROPERLY sql_result_free("alias_name"); sql_result_free("alias_domain"); setflag(10); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_reply("404", "Not Found"); exit; case -2: exit; } } append_branch(); route(5); } } route[5] { if (!t_relay()) { sl_reply_error(); } exit; } And with ngrep i can see: # U 10.10.10.10.:5060 -> 10.10.10.15:53286 INVITE sip:ncm0000a@10.20.5.72:5060 SIP/2.0. Record-Route: <sip:10.10.10.10;lr=on;ftag=D1D33F74-16CA;nat=yes>. Via: SIP/2.0/UDP 10.10.10.10;branch=z9hG4bKc1ce.a89602e5.1. Via: SIP/2.0/UDP 10.20.5.10:5060;rport=57078;branch=z9hG4bK6F2A7E0. Remote-Party-ID: <sip:85559955500@10.20.5.10 <sip%3A85559955500(a)10.20.5.10> >;party=calling;screen=yes;privacy=off. From: <sip:9955500@10.20.5.10 <sip%3A9955500(a)10.20.5.10>>;tag=D1D33F74-16CA. To: <sip:ncm0000b@domain.com <sip%3Ancm0000b(a)domain.com>>.To: < sip:ncm0000a@domain.com <sip%3Ancm0000a(a)domain.com>>. <---------------DUPLICATE HEADERS Date: Tue, 08 Jun 2010 06:39:41 GMT. Call-ID: 73CF85D3-71FF11DF-B5C1D8D5-9276F617(a)10.20.5.10. Supported: 100rel,timer,resource-priority,replaces. Min-SE: 1800. Cisco-Guid: 1942740043-1912541663-2892365836-2242214896. User-Agent: Cisco-SIPGateway/IOS-12.x. Accept-Language: ru. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. Timestamp: 1275979181. Contact: <sip:85559955500@10.20.5.10:57078>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 401. ............... I'm trying to rewrite this one: To: <sip:ncm0000b@domain.com<sip%3Ancm0000b(a)domain.com> >. with this one: To: <sip:ncm0000a@domain.com<sip%3Ancm0000a(a)domain.com>>. I'm a newbie, so, please, show me where is my mistake. Sincerely yours, Kosilov Fedor.

2 3

Problems starting Kamailio
by Toyima Dias 07 Jun '10

07 Jun '10

Hello sip-users, I'm having a problem configuring Kamailio on a Debian server, following the ERROR: madas3009:/home/ddias# /etc/init.d/kamailio start Starting Kamailio SIP server: kamailioListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 212.230.0.117 [212.230.0.117]:5060 Aliases: . madas3009:/home/ddias# Jun 7 14:31:42 [11789] NOTICE:core:main: version: kamailio 1.5.4-notls (x86_64/linux) Jun 7 14:31:42 [11789] INFO:core:main: using 64 Mb shared memory Jun 7 14:31:42 [11789] INFO:core:main: using 4 Mb private memory per process Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module db_mysql Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module mi_fifo Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module sl Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module tm Jun 7 14:31:42 [11789] INFO:tm:mod_init: fr_inv_timer_next value is <30> Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module rr Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module pv Jun 7 14:31:42 [11789] INFO:pv:shvar_init_locks: locks array size 16 Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module maxfwd Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module usrloc Jun 7 14:31:42 [11789] INFO:usrloc:ul_init_locks: locks array size 512 Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module registrar Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module textops Jun 7 14:31:42 [11789] INFO:core:init_mod: initializing module uri_db Jun 7 14:31:42 [11789] ERROR:db_mysql:db_mysql_new_connection: driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Jun 7 14:31:42 [11789] ERROR:core:db_do_init: could not add connection to the poolJun 7 14:31:42 [11789] ERROR:uri_db:uridb_db_ver: unable to open database connection Jun 7 14:31:42 [11789] ERROR:uri_db:mod_init: Error while querying table version Jun 7 14:31:42 [11789] ERROR:core:init_mod: failed to initialize module uri_db Jun 7 14:31:42 [11789] ERROR:core:main: error while initializing modules Am i doing something wrong? what should i check? Thanks in advance

2 1

A Problem With serctl
by Nahum Nir 07 Jun '10

07 Jun '10

Hi All, I have just installed SER 2.0 and when I run serctl I get command not found. What to do? Thanks, Nahum

2 1

Ending a dialog via MI
by Iñaki Baz Castillo 05 Jun '10

05 Jun '10

Hi, I'm building an external application in order to end dialogs via XMLRPC MI (Kamailio 1.5.4). For each call of a client I set a dialog profile: dlg_manage(); set_dlg_profile("out","$avp(AVP_CLIENT_USERNAME)"); The problem is that there is no way to find the current dialog information based on dialog profiles, as such information is not provided when running MI dlg_list or dlg_list_ctx. So, is there something I can do? I've realized that dlg_list_ctx also displays the dialog "context": "associated context from modules sitting on top of the dialog module" But of course such field is empty as I don't use a module on top of dialog module. But perhaps I could code something in the dialog module so such value could be set in the script (I need it to display $avp(AVP_CLIENT_USERNAME)). In this way, I would get the dialog list and filter these with "context = SOME_CLIENT_USERNAME". Then I get the h_entry and h_id and can perform the dlg_end_dlg MI command. Any suggestion about it please? BTW: Running dlg_list_ctx via MI FIFO I get this for each dialog: dialog:: hash=3960:1144646733 Should I assume that h_entry is 3960 and h_id is 1144646733? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 1

mysql queries per sec forever rising...almost
by Anders 05 Jun '10

05 Jun '10

Hi, I use Munin to try to understand what's going on on the server that runs Kamailio (with Siremis). One thing that puzzles me is that if I look at the MySql queries per second, it just keeps going up. The number rises during the time with calls - the more calls the more it rises - and then when there are no calls, it stays at the same level. And this is what I don't understand - when Kamailio is doing nothing, why are there 5k queries per second? Sometimes - every few weeks - it takes a steep drop to very few queries per second... - and then back to rising again. Is this a database issue, a Siremis issue, or a Munin issue? Or is it just something I don't understand that it's supposed to keep rising forever? Thanks! //Anders

4 3

kamailio project presence at linuxtag 2010
by Henning Westerholt 04 Jun '10

04 Jun '10

Hello, the kamailio, sip-router and sems projects will be present next week on the linuxtag 2010 - "the most important place for Linux and open source software in Europe." [1]. We're at booth 106, this is in hall 7.2b. If you're during next week in Berlin and visit the exibition, then we're looking forward to meet you there! The kamailio project will be also present in the conference program, with a talk about building geographical redundant VoIP systems [2]. Best regards, Henning Westerholt [1] http://www.linuxtag.org/2010/en.html [2] http://www.linuxtag.org/2010/en/program/free- conference/wednesday/details.html?talkid=104 -- Henning Westerholt - Technical Leader VoIP backend development 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 1

Re: [SR-Users] [sr-dev] pua_dialoginfo deletes AVPs?
by Henning Westerholt 04 Jun '10

04 Jun '10

On Friday 04 June 2010, Jasmin Schnatterbeck wrote: > onreply_avp_mode does not seem to be available in kamailio 3.0.2 (at > least not in the module doc) > I think this parameter is for avps in reply_route? or does it work for > branch_route also? Hi Jasmin, ah, sorry. Its not available or necessary anymore in 3.0. In 1.5 it also affected the branch_route, i think, but for the newer version this behaviour is now the default. Then in your case its probably another problem.. Cheers, Henning

1 0

gmake: *** [cfg.tab.h] Broken Pipe while make kamailio on solaris 10
by JinKevin 04 Jun '10

04 Jun '10

Hi All, I'm trying to install kamailio3.0.x on solaris 10 (sparc) as following: gmake prefix="/usr/local/kamailio-3.0" include_modules="db_mysql dialplan" cfg gmake prefix="/usr/local/kamailio-3.0" all but get the Broken Pipe error as below: ----snip ----- from cfg/cfg_struct.c:40: cfg/../atomic/atomic_sparc.h:39:2: warning: #warning "sparc32 atomic operations support not tested" gcc -g -O9 -funroll-loops -mtune=v8 -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"sparc"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc -D__OS_solaris -DSER_VER=3000002 -DCFG_DIR='"/usr/local/kamailio-3.0/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -c cfg/cfg.c -o cfg/cfg.o -MMD -MP In file included from cfg/../atomic/atomic_native.h:68, from cfg/../atomic_ops.h:166, from cfg/cfg_struct.h:38, from cfg/cfg.c:36: cfg/../atomic/atomic_sparc.h:39:2: warning: #warning "sparc32 atomic operations support not tested" yacc -d -b cfg cfg.y gmake: *** [cfg.tab.h] Broken Pipe ---end of snip --- what is the cause of the issue? How should I fix it? Your help are greatly appreciated! Thanks, Kevin _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

3 8

New debian and ubuntu repository for kamailio released!
by Jon Bonilla 04 Jun '10

04 Jun '10

We are proud to announce the release of a Debian and Ubuntu repository for Kamailio. We hope this will make easier installation and update of Kamailio on Debian and Ubuntu servers. Kamailio project's web page hosts the description and instructions: http://www.kamailio.org/dokuwiki/doku.php/packages:debs cheers, Sipdoc Team

3 2

join the freeswitch public conference today to listen to daniel!
by Meftah Tayeb 03 Jun '10

03 Jun '10

hi sr folk, please join #freeswitch public weekly conference to listen to daniel, our guest about integration betwan freeswitch and kamailio sip-URI: sip:888@conference.freeswitch.org iveryone is welcome to join! -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

1 0

Failed to make module db_mysql on Solaris 10 Sparc
by JinKevin 03 Jun '10

03 Jun '10

Hi All, While [ gmake prefix="/usr/local/kamailio-3.0" all ] I got the error below regarding the db_mysql, however I have already installed the mysql server in the path /usr/local/mysql and all the .h and lib files are there. Please suggest what's wrong here and how to fix the issue. ========= Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/db_mysql' gcc -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -Wall -DNAME='"kamailio"' -DVERSION='"3.0.2"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3000002 -DCFG_DIR='"/usr/local/kamailio-3.0/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DF_MALLOC -DUSE_TLS -DTLS_HOOKS -DSTATISTICS -DMALLOC_STATS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_dbase.c -o km_dbase.o -MMD -MP km_dbase.c:38:25: warning: mysql/mysql.h: No such file or directory km_dbase.c:39:26: warning: mysql/errmsg.h: No such file or directory km_dbase.c:40:33: warning: mysql/mysql_version.h: No such file or directory In file included from km_dbase.c:47: km_my_con.h:47: error: syntax error before "MYSQL_RES" km_my_con.h:47: warning: no semicolon at end of struct or union km_my_con.h:48: warning: type defaults to `int' in declaration of `con' km_my_con.h:48: warning: data definition has no type or storage class km_my_con.h:49: error: syntax error before "row" km_my_con.h:49: warning: type defaults to `int' in declaration of `row' km_my_con.h:49: warning: data definition has no type or storage class km_my_con.h:51: error: syntax error before '}' token km_dbase.c: In function `db_mysql_submit_query': km_dbase.c:81: error: dereferencing pointer to incomplete type km_dbase.c:82: warning: implicit declaration of function `mysql_ping' km_dbase.c:82: error: dereferencing pointer to incomplete type km_dbase.c:83: warning: implicit declaration of function `mysql_error' km_dbase.c:83: error: dereferencing pointer to incomplete type km_dbase.c:83: warning: format argument is not a pointer (arg 7) km_dbase.c:83: error: dereferencing pointer to incomplete type km_dbase.c:83: error: dereferencing pointer to incomplete type km_dbase.c:83: warning: format argument is not a pointer (arg 6) km_dbase.c:83: error: dereferencing pointer to incomplete type km_dbase.c:83: error: dereferencing pointer to incomplete type km_dbase.c:91: error: dereferencing pointer to incomplete type km_dbase.c:109: warning: implicit declaration of function `mysql_real_query' km_dbase.c:109: error: dereferencing pointer to incomplete type km_dbase.c:112: warning: implicit declaration of function `mysql_errno' km_dbase.c:112: error: dereferencing pointer to incomplete type km_dbase.c:113: error: `CR_SERVER_GONE_ERROR' undeclared (first use in this function) km_dbase.c:113: error: (Each undeclared identifier is reported only once km_dbase.c:113: error: for each function it appears in.) km_dbase.c:113: error: `CR_SERVER_LOST' undeclared (first use in this function) km_dbase.c:117: error: dereferencing pointer to incomplete type km_dbase.c:117: warning: format argument is not a pointer (arg 7) km_dbase.c:117: error: dereferencing pointer to incomplete type km_dbase.c:117: error: dereferencing pointer to incomplete type km_dbase.c:117: warning: format argument is not a pointer (arg 6) km_dbase.c:117: error: dereferencing pointer to incomplete type km_dbase.c:117: error: dereferencing pointer to incomplete type km_dbase.c: In function `db_mysql_store_result': km_dbase.c:166: error: dereferencing pointer to incomplete type km_dbase.c:166: warning: implicit declaration of function `mysql_store_result' km_dbase.c:166: error: dereferencing pointer to incomplete type km_dbase.c:167: error: dereferencing pointer to incomplete type km_dbase.c:168: warning: implicit declaration of function `mysql_field_count' km_dbase.c:168: error: dereferencing pointer to incomplete type km_dbase.c:173: error: dereferencing pointer to incomplete type km_dbase.c:173: warning: format argument is not a pointer (arg 7) km_dbase.c:173: error: dereferencing pointer to incomplete type km_dbase.c:173: error: dereferencing pointer to incomplete type km_dbase.c:173: warning: format argument is not a pointer (arg 6) km_dbase.c:173: error: dereferencing pointer to incomplete type km_dbase.c:173: error: dereferencing pointer to incomplete type km_dbase.c:188: warning: implicit declaration of function `mysql_free_result' km_dbase.c:188: error: dereferencing pointer to incomplete type km_dbase.c:195: error: dereferencing pointer to incomplete type km_dbase.c: In function `db_mysql_free_result': km_dbase.c:228: error: dereferencing pointer to incomplete type km_dbase.c:229: error: dereferencing pointer to incomplete type km_dbase.c: In function `db_mysql_fetch_result': km_dbase.c:294: error: dereferencing pointer to incomplete type km_dbase.c:294: error: dereferencing pointer to incomplete type km_dbase.c:295: error: dereferencing pointer to incomplete type km_dbase.c:296: error: dereferencing pointer to incomplete type km_dbase.c:301: error: dereferencing pointer to incomplete type km_dbase.c:301: warning: format argument is not a pointer (arg 7) km_dbase.c:301: error: dereferencing pointer to incomplete type km_dbase.c:301: error: dereferencing pointer to incomplete type km_dbase.c:301: warning: format argument is not a pointer (arg 6) km_dbase.c:301: error: dereferencing pointer to incomplete type km_dbase.c:301: error: dereferencing pointer to incomplete type km_dbase.c:312: warning: implicit declaration of function `mysql_num_rows' km_dbase.c:312: error: dereferencing pointer to incomplete type km_dbase.c:351: error: dereferencing pointer to incomplete type km_dbase.c:351: warning: implicit declaration of function `mysql_fetch_row' km_dbase.c:351: error: dereferencing pointer to incomplete type km_dbase.c:352: error: dereferencing pointer to incomplete type km_dbase.c:353: error: dereferencing pointer to incomplete type km_dbase.c:353: warning: format argument is not a pointer (arg 7) km_dbase.c:353: error: dereferencing pointer to incomplete type km_dbase.c:353: error: dereferencing pointer to incomplete type km_dbase.c:353: warning: format argument is not a pointer (arg 6) km_dbase.c:353: error: dereferencing pointer to incomplete type km_dbase.c:353: error: dereferencing pointer to incomplete type km_dbase.c: In function `db_last_inserted_id': km_dbase.c:465: warning: implicit declaration of function `mysql_insert_id' km_dbase.c:465: error: dereferencing pointer to incomplete type gmake[1]: *** [km_dbase.o] Error 1 gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/kamailio/modules/db_mysql' gmake: *** [modules] Error 1 AppSer01:root@/usr/local/src/kamailio-3.0.0/kamailio$ AppSer01:root@/usr/local$ ls -l total 48 drwxr-xr-x 4 bin bin 4608 Jun 1 02:58 bin drwxr-xr-x 46 bin bin 1024 Jun 1 02:58 doc drwxr-xr-x 4 bin bin 512 Apr 15 04:32 etc drwxr-xr-x 12 bin bin 1024 Jun 1 02:58 include drwxr-xr-x 2 bin bin 1024 Apr 15 04:32 info drwxr-xr-x 11 bin bin 4608 Jun 1 02:58 lib drwxr-xr-x 4 bin bin 512 Apr 15 04:31 libexec drwxr-xr-x 8 bin bin 512 Apr 15 04:32 man lrwxrwxrwx 1 root root 35 Jun 1 06:04 mysql -> mysql-5.1.47-solaris10-sparc-64bit/ drwxr-xr-x 13 root mysql 512 Jun 1 06:29 mysql-5.1.47-solaris10-sparc-64bit drwxr-xr-x 10 bin bin 512 Apr 15 04:31 samba drwxr-xr-x 2 bin bin 512 Apr 15 04:32 sbin drwxr-xr-x 12 bin bin 512 Apr 15 04:32 share drwxr-xr-x 3 root root 512 May 31 03:10 src drwxr-xr-x 9 bin bin 512 Apr 15 04:31 ssl drwxr-xr-x 3 bin bin 512 Apr 15 04:30 var AppSer01:root@/usr/local$ cd mysql /usr/local/mysql AppSer01:root@/usr/local/mysql$ ls include/ decimal.h m_string.h my_dbug.h my_list.h my_sys.h mysql_time.h mysqld_error.h sslopt-case.h errmsg.h my_alloc.h my_dir.h my_net.h my_xml.h mysql_version.h plugin.h sslopt-longopts.h keycache.h my_attribute.h my_getopt.h my_no_pthread.h mysql_com.h mysql.h sql_common.h sslopt-vars.h m_ctype.h my_config.h my_global.h my_pthread.h mysql_embed.h mysqld_ername.h sql_state.h typelib.h AppSer01:root@/usr/local/mysql$ ls lib/ libdbug.a libmysqlclient_r.la libmysqlclient.la libmystrings.a plugin libheap.a libmysqlclient_r.so libmysqlclient.so libmysys.a libmyisam.a libmysqlclient_r.so.16 libmysqlclient.so.16 libvio.a libmyisammrg.a libmysqlclient_r.so.16.0.0 libmysqlclient.so.16.0.0 libz.a libmysqlclient_r.a libmysqlclient.a libmysqld.a libz.la AppSer01:root@/usr/local/mysql$ === Thanks, Kevin Jin _________________________________________________________________ MSN十年回馈，每位用户可免费获得价值25元的卡巴斯基反病毒软件2010激活码，快来领取！ http://kaba.msn.com.cn/?k=1

2 2

check_from and check_to: case sensitive?
by Iñaki Baz Castillo 03 Jun '10

03 Jun '10

Hi, when using check_from or check_to, is the username part of the uri checked in mode case sensitive or insensitive? It should be case sensitive as per SIP grammar. But in the code (uri_db/checks.c) I see the opposite: /* URI table not used, simply compare digest username and From/To * username, the comparison is case insensitive */ if (_uri->user.len == c->digest.username.user.len) { if (!strncasecmp(_uri->user.s, c->digest.username.user.s, _uri->user.len)) { LM_DBG("Digest username and URI username match\n"); return 1; } } -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 12

call routing using kamailio 3.0.2
by Meftah Tayeb 02 Jun '10

02 Jun '10

hello, how do i route calls from local registered users to a pstn gateway? thank you

2 2

Audio Conference: Kamailio and FreeSWTICH - Wed, June 2, 2010
by Daniel-Constantin Mierla 02 Jun '10

02 Jun '10

Hello, next Wednesday (June 2, 2010), at 18:00UTC, I will participate to FreeSWITCH weekly conference call, talking about using Kamailio and FreeSWITCH together. The draft of agenda is: * Goals of Kamailio, how it differentiates from FreeSWITCH and why using them together creates a very powerful framework to build large VoIP systems. * Kamailio for sip routing and FreeSWITCH as media server (conferencing, voicemail, IVR, announcements) * FreeSWITCH as a B2BUA for Kamailio (topology hiding, transcoding, call interrupt detection) * FreeSWITCH as prepaid engine for Kamailio * Load balancing FreeSWITCH servers with Kamailio Is an open debate, join and ask your questions. You can dial in via SIP, Skype, PSTN, see access details at: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_02 Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

3 3

fix_nated_contact() on REGISTER
by Klaus Darilion 02 Jun '10

02 Jun '10

Hi! Does somebody know what happens when I call fix_nated_contact() and save() on a REGISTER request? What will be stored and used as RURI? The original contact URI or the fixed one? I would guess the original one, as message manipulations are not visible when calling save() - true? regards Klaus

5 12

VoIPToday Kamailio interview
by Daniel-Constantin Mierla 01 Jun '10

01 Jun '10

FYI: VoIPToday.org has published an interview I gave related to Kamailio project: http://bit.ly/9gVzvS Meanwhile, asyncrhonous TLS is already in GIT, thanks to Andrei. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Kamailio and FreeSWITCH as media server
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, a new tutorial about using Kamailio as SIP server and FreeSWITCH as media server is published at: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

New way of Kamailio and Asterisk Real Time Integration
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, several days ago, I published a new tutorial about Kamailio and Asterisk realtime integration, using a different approach than in the previous tutorials on this topic I wrote. Specifically, this time the Asterisk database structure is used. You can find it at: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Two presentations at Amoocon 2010
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, next weekend (June 4-6, 2010) in Rosotck, Germany, takes place 2010 edition of Amoocon. http://www.amoocon.com I will give two talks about Kamailio - SIP Router: - Asynchronous SIP Routing - Develop Your SIP Routing in Lua More details are posted on news section of the web sites: http://www.kamailio.org/w/2010/05/kamailio-at-amoocon-2010/ If you are around and want to meet, drop me an email. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Re: [SR-Users] Dialplan modinit failed
by Daniel-Constantin Mierla 30 May '10

30 May '10

Hello, please use the sr-users(a)lists.sip-router.org to post questions about ser or kamailio - apart of first one, most private messages are discarded. There are many people that can answer, faster and better. In this particular case, seems you haven't loaded pv module (located in modules_k). Cheers, Daniel On 5/30/10 9:50 AM, Dmitry Kirillov wrote: > Hi > > i tried to start SIP-Router (ser-3.0.2_src_2010-05-29) by including > dialplan.so in the list of loaded > modules and empty dialplan modparam. when i start sip-router, i get > > 0(23147) ERROR: <core> [pvapi.c:501]: error searching pvar "ruri.user" > 0(23147) ERROR: <core> [pvapi.c:705]: wrong char [r/114] in > [$ruri.user] at [9 (0)] > 0(23147) ERROR: dialplan [dialplan.c:163]: input pv is invalid > 0(23147) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > With modparam("dialplan", "attrs_pvar", "$avp(s:dest)"): > > 0(23158) ERROR: <core> [pvapi.c:501]: error searching pvar "avp" > 0(23158) ERROR: <core> [pvapi.c:705]: wrong char [s/115] in > [$avp(s:dest)] at [5 (5)] > 0(23158) ERROR: dialplan [dialplan.c:149]: invalid pvar name > 0(23158) ERROR: <core> [sr_module.c:874]: init_mod(): Error while > initializing module dialplan > ERROR: error while initializing modules > > there is no dialplan function call in ser.cfg. > > Please, help > -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Siremis with grandchild(?)
by Anders 29 May '10

29 May '10

Hi, This might not be directly Siremis-related, but in any case it is a consequence of using Siremis: The stored procedure for calculating and populating the 'cdrs' table has started to act out all of a sudden - it won't do anything. In my syslog it looks like this: May 29 09:24:01 vae /USR/SBIN/CRON[25555]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u root -pxxx kamailio >/dev/null) May 29 09:24:23 vae /USR/SBIN/CRON[25548]: (CRON) error (grandchild #25549 failed with exit status 1) May 29 09:24:23 vae /USR/SBIN/CRON[25548]: (CRON) info (No MTA installed, discarding output) ..and that's every time the cron job runs, the only thing that changes is the "grandchild" number. I have no clue what "grandchild" means in this context - ...anyone? Thanks!! //Anders

1 1

Issue with starting SER
by Jesse B Scites 28 May '10

28 May '10

I am not sure if this mailing list is for any type of product support, but I am at a loss as how to continue, so any type of guidance would be appreciated. I am new to using SER, however I downloaded the source baseline, then made it, etc. according to the "Getting Started" guide. The very basic configuration (minus MySQL, etc.) seemed to build fine and all binaries, the ser.cfg file, etc. were installed to the correct locations. Since I just wanted to start using a basic configuration, I chose to use the "Hello World" configuration file as outlined in the guide, so I swapped it out with the default one. As the super user, I ran >>ser -V and received the expected version feedback as listed within the "Getting Started" guide. However, when I tried to start SER normally with >>ser I received the following error: 0(6989) parse error (11,1-5): syntax error 0(6989) parse error (11,1-5): ERROR: bad config file (2 errors) So I went back and checked the "Hello World" config file and I could find no syntax errors since the exact replica can be compared to the example in the "Getting Started" guide. So I swapped back in the original ser.cfg file and I still got errors. Of course I also got additional errors with that file since I did not have MySQL, etc configured. So I guess my question would be, what may be causing SER to error out on the config file (I assume it means there is some issue with parsing out the contents of the ser.cfg file). I am using Fedora 11 on an i386 platform with gcc 4.4.1, bison 2.4.1, flex and all other needed dependencies listed within the INSTALL file. Any suggestions would be much appreciated. Regards, Jesse Scites

2 1

new developer access: Jon Bonilla
by Daniel-Constantin Mierla 28 May '10

28 May '10

Hello, lately, most of the bugs related to Debian packaging were filled by Jon Bonilla (Manwe), so the solution was to ask him to join and help to fix/improve the spec files. Jon has been around for quite some time, working with other SIP related projects as well. I am looking forward to his contributions to packaging system, part which was not much under focus lately, but needs to be brought up-to-date to ease the installation on various OSes. His commit id is: manwe Welcome Jon! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

4 3

Multiple domains in different servers
by Ricardo Coelho 28 May '10

28 May '10

Hi, Right now I have 2 machines (one on domainA and the other on domainB). I want to allow a phone registered in the openser from domainA to call a phone registered in the openser from domainB (and vice-versa). I have already inserted both domains in both openser but when I try to call a "user not found" message is returned. Can anyone help me? Thanks in advanced, Ricardo Coelho

5 11

continue processing route script if no response from gateway
by JR Richardson 28 May '10

28 May '10

Hi All, I am lab testing carrierroute modue on kamailio 1.5.4-notls (i386/linux) and have a question on how to continue processing a call if kamailio sends a call with t_relay() but does not get a response from the gateway. I read about the timers in the tm module and set them pretty low in the config file: # ----- tm params ----- modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer_next", 3) But nothing happens in the routing script after an invite is sent and there is no response. Debug just shows the invite to the gateway being re-transmited several times. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses

3 7

kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 28 May '10

28 May '10

Hello, I am planning to release kamailio 3.0.2 this Thursday. There were some fixes since 3.0.1 that worth to be packaged. If you have major reports for current stable version, please write to sr-dev(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

3 7

RPID Question
by dotnetdub 28 May '10

28 May '10

Hi All, Kamailio 3 I currently store a default RPID in our database. We allow some asterisk customers to send us the RPID. I want to store the RPID sent by the customer on the radius record. We test: if!(is_present_hf("Remote-Party-ID")) { #RPID Not Set by Asterisk - Will set now from DB append_rpid_hf(); } Do I use textops to get the RPID that the Asterisk sends and then overwrite what we read from the database? $avp(s:rpid) = ? Not really sure how to extract the RPID from the SIP message sent from the asterisk. TIA. Stephen.

3 2

new feature: embedded http 1.0 server
by Daniel-Constantin Mierla 28 May '10

28 May '10

Hello, there is a new module named xhttp which provides an embedded http server inside the SIP server. It reuses the transport layer from the core, so you can send http requests via TCP and TLS. The port is the same as for SIP traffic. The documentation of the module is available at: http://sip-router.org/docbook/sip-router/branch/master/modules/xhttp/xhttp.… Since sip and http are very similar, you can use other modules in handling the http requests. For example, you can do www-authentication: event_route[xhttp:request] { if (!www_authorize("xcap", "subscriber")) { www_challenge("sip-router.org", "0"); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Welcome $au, you are connected from $si:$sp</body></html>"); } allowing in this way only local SIP users to do http requests. For now supports only http 1.0, meaning it cannot take chunked requests (usually happening in PUT requests with large body). Hope it is useful for some of you. Its main goal right now is to offer the transport layer for embedded xcap server (the new xcap_server module), but it can be used for service monitoring, server control, etc. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

Kamailio v3.0.2 Released
by Daniel-Constantin Mierla 27 May '10

27 May '10

Hello, Kamailio v3.0.2 was just released. It is a patch release that includes the fixes done since release of v3.0.1. You should consider to upgrade if you run v3.0.0 or or v3.0.2, there is no change that you should do to configuration file or database, just reinstall the binaries. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/3.0.2/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/3.0.2/ChangeLog Download via GIT: # git clone –depth 1 git://git.sip-router.org/sip-router kamailio # cd kamailio # git checkout -b kamailio_3.0 origin/kamailio_3.0 Binaries and packages will be uploaded at : http://www.kamailio.org/pub/kamailio/3.0.2/ Modules’ documentation: http://www.kamailio.org/docs/modules/3.0.x/ What is new in 3.0.x release series is summarized in the announcement of v3.0.0: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ If you are eager to see what's up to next major release (3.1.0), visit: http://sip-router.org/wiki/features/new-in-devel Enjoy, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

Carrierroute failover domain not working
by JR Richardson 27 May '10

27 May '10

Hi All, I'm testing carrierroute modue in kamailio 1.5.4-notls (i386/linux). I have things working like this: sipp><kamailio carrierroute>< 2 asterisk servers A & B even distribution of calls So A & B are in carrier1 domain1, I have failureroute setup with 2 more asterisk servers C& D in carrier1 domain2. While I'm processing calls from sipp and sending to A & B, I can fail A and then all calls go to B as expected. But if I fail both A & B, the calls fail, so I'm guessing I don't have the faliure_route configured correctly. Here is my current config: route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!allow_trusted()) { sl_send_reply("403", "Forbidden"); exit; }; if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); route(2); } route[1] { if (!t_relay()) { sl_reply_error(); } exit; } route[2] { if(!cr_route("1", "1", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { t_on_failure("2"); route(1); } } failure_route[2] { if (t_check_status("408|5[0-9][0-9]")) { if(!cr_route("1", "2", "$rU", "$rU", "call_id")){ t_reply("403", "Not allowed"); } else { t_relay(); } } } I think the database entries are setup ok, I used the example in the module doc, but I can share that if needed. Any guidance will be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 4

functions for modifying pAsserted
by Alex rsm 27 May '10

27 May '10

Hi, I couldn't find the module that has the functions for modifying pAsserted field. Your help would be appreciated. Thanks, AR _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

3 2

Siremis 1.0 login fails (or, does nothing)
by Joe Micciche 27 May '10

27 May '10

RHEL5.5 server Kamailio 3.0 built from source via Git Apache/2.2.3 PHP 5.3.2 MySQL 5.1.47 chown -R apache /var/www/html/siremis-1.0.0, apache is the httpd user. Used the instructions from Asipto's site for Siremis install and configure, added the alias to httpd.conf, Config.xml is: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> </DataSource> </Config> I navigate to the Siremis page, http://10.5.10.21/siremis/ and get the login screen, enter the login credentials, Submit, and nothing happens. httpd/error_log: [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ I have wiped Siremis and done a fresh install, restarted httpd, and still nothing. Does anyone have ideas or suggestions on where else to look? -- ================================================================== Joe Micciche jmiccich(a)redhat.com Red Hat, Inc. http://www.redhat.com Senior Communications Engineer X(81) 44554 +1.919.754.4554 ==================================================================

6 15

t_check_trans return false on CANCEL
by Ernest Mavrel 27 May '10

27 May '10

3 2

MAX_BRANCHES
by Alex Balashov 27 May '10

27 May '10

This may seem like an absurd question, but is there a particular reason why MAX_BRANCHES is #defined as 12 in config.h? Any particular reason for 12? Is it okay to make it higher? -- 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 5

prefix with kamailio
by alexis heron 27 May '10

27 May '10

hello I want to absorb a number with kamailio just to make out only three but I can not. In fact I send 1941 kamailio and i spring 941 with kamailio so I absorb a number. Here is how I : if ($rU=~"^1+941$") { log(1, "kamailio vers CCM 9xx"); route(10); } route[10] { log(1, "In route[10]"); rewritehostport("142.145.98.138:5060"); append_hf("P-hint: redirection CCM\r\n"); xlog("L_INFO", "$rm from $fu to $tu"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } Can someone help me?? best regards

3 4

Question about inbound Alias Routing
by dotnetdub 26 May '10

26 May '10

Hi List, At the moment I use dbalias to send inbound from the PSTN <--> subscribers. I have a new opportunity that requires me to send inbound to a URI - SIP/353xxxxxxx(a)customer.proxy - not really sure how to accomplish this. I have a feeling there is a module I should be looking at to do this possibly? If someone could point me in the correct direction. TIA. Stephen.

2 1

A Couple of Questions about carrierroute moduel
by JR Richardson 26 May '10

26 May '10

Hi All, I'm testing carrierroute module on version: kamailio 1.5.4-notls (i386/linux). I'm distributing calls to a group of Asterisk servers, all in the same carrier and domain, wanted to load balance the calls and I'm curious how this works. I see the 'probability' parameter in the carrierroute table and set it to equal value across all servers, I have 4 servers I'm sending calls to, so I set each prob value to '0.25'. I'm testing with sipp and hashing over call_id so this is the same for every call. I send 40 calls and the distribution of calls to the asterisk servers are 11, 12, 6, 11, pretty close to even but not 100%. How does the distribution algorithm work for carrieroute? Does it statefully track all active calls and load balance across each server in destination group? Is there another algorithm that can offer that can get closer to a round robin effect? Another question is do I need to include the record_route() and the t_check_trans() functions prior to calling the carrierroute functions? I see in the debug "DBG:core:forward_reply: found module tm, passing reply to it" and dialogs seem to process fine, so does the carrierroute module handle those requirements for stateful processing? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Routing from Asterisk to Microsoft UM for voicemail.
by Nelson Pereira 26 May '10

26 May '10

If someone has a known good working route configuration for use with Microsoft Unified Messaging, please post your config. Thanks Nelson Pereira

2 1

Calculate ha1 with mysql trigger
by Ole Kaas 26 May '10

26 May '10

Hello, When editing the subscriber table using phpmyadmin, updating the ha1 fields are no fun and error prone. The trigger below will calculate the hashes if the input fields are left empty. Navigate to the subscriber table in phpmyadmin and drop the code in the query field under the "SQL" tab. Remeber to set the delimiter to // /Ole DROP TRIGGER IF EXISTS calcha1 // CREATE TRIGGER calcha1 BEFORE INSERT ON subscriber FOR EACH ROW BEGIN IF NEW.ha1 = "" THEN SET NEW.ha1 = MD5(CONCAT(NEW.username,":",NEW.domain,":",NEW.password)); END IF; IF NEW.ha1b = "" THEN SET NEW.ha1b = MD5(CONCAT(NEW.username,"@",NEW.domain,":",NEW.domain,":",NEW.password)); END IF; END//

4 3

'serctl add <name> <password> <email>'
by Sanjeev Singh 26 May '10

26 May '10

Hi All, Please help me this i am using below command to add subscribers in the ser so that i can make a phone call. 1) what is the meaning of this command 2) how to use this command? *'serctl add <name> <password> <email>'* can you please help me on this also. 1) how to register my sip client with ser server 2) how to add subscribers so that i can make a phone call. Regards, Sanjeev Singh

1 0

permissions module
by Asim Riaz 25 May '10

25 May '10

Hello List, from the documentation, permissions.allow, allows the calls either by matching From-URI or R-URI and permission.deny denies the call from matching uri. Do I need to divide the file into different sections, for R-URI and for From-URI? Not sure exact syntax could someone explain me that if I want to match in permissions.allow if a phone 10.10.10.10 dial(r-uri) 1234 allow it and everything else is blocked? What is the preference does kamailio check for permissions.deny first if there is no matching then checks permissions.allow? Thanks in advance, Asim

2 1

Re: [SR-Users] sipp and stateful transaction problem
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, On 5/22/10 2:22 AM, JR Richardson wrote: > On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 5/21/10 10:47 PM, JR Richardson wrote: >> >>> Hi All, >>> >>> I'm doing some testing with kamailio 1.5: >>> >>> kamailio1:/etc/kamailio# kamailio -V >>> version: kamailio 1.5.4-notls (i386/linux) >>> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >>> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> svnrevision: 2:6005M >>> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >>> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 >>> >>> Using dispatcher module trying to load balance SIP calls across some >>> Asterisk servers. I have it working fine when I test in this >>> scenario: >>> >>> sip phone dial out><asterisk><kamailio><round robin to several asterisk >>> servers >>> >>> This works stateful and stateless, handles everything gracefully. >>> >>> This scenario is giving me fits: >>> >>> sipp dial out><kamailio><round robin to several asterisk servers >>> >>> I get retransmits on every call back to sipp with errors like >>> "Discarding message which can't be mapped to a known SIPp call" and >>> "SIP/2.0 481 Call leg/transaction does not exist" >>> >>> This happens with kamailio setup stateful or stateless. I'm wondering >>> if sipp is the problem or just doesn't play well with kamailio? >>> >>> I've kept the config as simple as possible for testing, it is listed >>> here http://pastebin.com/BZ8hJvJv >>> >>> Here is my sipp usage: >>> >>> sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 >>> -trace_err >>> >>> Any insight would be appriciated. >>> >>> >>> >> the problem is in your sipp scenario. The uac calls do not map to uas. >> kamailio does not reply 481, check the uas scenario, that is the one that >> sends back the 481. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio (OpenSER) Advanced Training >> Miami, Fl, USA - June 21-23, 2010 >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> > Thanks Daniel, I reveiwed the sipp docs, '-sn uas' just sits there as > a responder, it will not initiate calls to kamailio. I don't > understand what you are getting at? How would I use this type of > scenario to test? > keeping the mailing list cc-ed is recommended, since others can respond faster and new people can benefit of the discussion. What I wanted to say is that kamailio does not reply 481. So the problem is in the responder of requests sent by UAC and forwarded by Kamailio. Somehow, the dialog is destroyed before the BYE (or other in-dialog request) is sent by UAS. If you can grab the SIP trace of such call (e.g., using ngrep on kamailio server), I can give more hits (try to select the sip flow for one such call only, sending full sip trace will be too big). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

2 2

Re: [SR-Users] kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 25 May '10

25 May '10

On 5/25/10 8:51 PM, SIP wrote: > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am planning to release kamailio 3.0.2 this Thursday. There were some >> fixes since 3.0.1 that worth to be packaged. If you have major reports >> for current stable version, please write to sr-dev(a)lists.sip-router.org >> >> Cheers, >> Daniel >> >> > Is this a drop-in upgrade from 3.0.1 (i.e. recompile modules and core > and keep the rest in place)? > > yes, changes to last number in version string mean that config language and database structure are the same. This is a minor release, bringing only fixes to the code. So upgrading from 3.0.0 or 3.0.1 to 3.0.2 is exactly what you said, just recompile and reinstall modules and core, keep the rest in place. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

could not bind to the TM-Module
by Javier Menendez 25 May '10

25 May '10

Hi, I have been configuring an Kamailio 3.0.1, with the dispatcher module. Everything seems to work just fine but when a gateway is in Probing mode it remains in that state for ever. checking the logs I found some weird messages, I think that is the rrot cause of the problem but I have no clue how to solve it. These are the messages I mentioned before May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:516]: find_export_record: <load_tm> not found May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: WARNING: dispatcher [dispatcher.c:358]: could not bind to the TM-Module, automatic re-activation disabled. May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: sl May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [statistics.c:105]: statistics manager successfully initialized May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [md5utils.c:87]: DEBUG: MD5 calculated: f8f2ab2c1295e90ed7dbb499b30f44b2 May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:506]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] As you may see load_tm is not found in the dispatcher but it is found in md5utils?? These are the modules included mpath="/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_k/" loadmodule "tm.so" loadmodule "dispatcher.so" loadmodule "sl.so" loadmodule "db_mysql.so" loadmodule "mi_fifo.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 "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "permissions.so" loadmodule "drouting.so Any clue?

2 3

could not open module kamailio
by Bruno Bresciani 25 May '10

25 May '10

I compiled the route.so module to the kamailio but when I start kamailio the follow error is generated in the log file: "ERROR:core:sr_load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: add_hash" Someone can help me? Cheers, Bruno

2 1

kamailio 3.0.1 rpm for centos 5.5
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, for those interested, kamailio 3.0.1 rpm packages for centos 5.5 are available for download at: http://www.kamailio.org/pub/kamailio/3.0.1/packages/centos-5.5/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

R: SIP Router devel meeting, Berlin, June 8, 2010
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, a short reminder for those that want to participate to SIP Router devel meeting and haven't been in contact with me yet, you need to send email to: *registration(a)lists.sip-router.org* See more details at: http://sip-router.org/2010/05/12/sip-router-devel-meeting-june-8-2010-berli… Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

1 0

no Contact in REFER generated with dlg_bridge
by Anton Roman 25 May '10

25 May '10

Hi all, I'm trying to implement a Click2Dial service by using the dlg_bridge command from the dialog module. Although it works in this case, I found two problems in the scenario and I would like to read your opinion about them. 1) Fisrt problem: the REFER generated by Kamailio doesn't contain a contact header. As per RFC 3515, this request should have a Contact header. This REFER is not being accepted by a proprietary device. This is can be worked around with the append_hf() command from the textops module, but I don know if it is a good solution. 2) The second one arises when the ACK generated by Kamailio, goes through Kamailio (caller and calle involved in click2dial are registered on the Kamailio server): the branch of the second via header is not correctly populated. Its content is '0' when it must be tha same as in the INVITE. Below you can find the traces of the whole scenario. Could you take a look at them and give me your feedback? Thanks in advance, best regards Antón Román T +8.969957 127.0.0.1:44785 -> 127.0.0.1:5060 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 331 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>mi_fifo</methodName> <params> <param> <value><string>dlg_bridge</string></value> </param> <param> <value><string>sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103></string></value> </param> <param> <value><string>sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103></string></value> </param> </params> </methodCall> ## U +0.001779 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # U +0.001330 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7160 req_src_ip=10.1.20.103 req_src_port=5060 in_uri=sip:111@10.1.20.103<sip%3A111(a)10.1.20.103>out_uri= sip:111@10.1.3.15:5061 via_cnt==1" # U +0.000727 10.1.20.103:5060 -> 10.1.3.15:5061 INVITE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # T +0.000442 127.0.0.1:5060 -> 127.0.0.1:44785 [AP] HTTP/1.0 200 OK Via: SIP/2.0/TCP 127.0.0.1:44785 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 211 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=7172 req_src_ip=127.0.0.1 req_src_port=44785 in_uri=sip:127.0.0.1:9 out_uri=sip: 127.0.0.1:9 via_cnt==1" <?xml version="1.0"?> <methodResponse> <params> <param> <value><array><data> <value><string>200 OK </string></value> <value><string> </string></value> </data></array></value> </param> </params> </methodResponse> ##### U +0.021063 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.030585 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000384 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +2.440228 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000315 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001190 10.1.20.103:5060 -> 10.1.20.103:5060 ACK sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK Route: <sip:10.1.20.103;lr=on;did=768.90707151> User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.001249 10.1.20.103:5060 -> 10.1.3.15:5061 ACK sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 *Via: SIP/2.0/UDP 10.1.20.103;branch=0* Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.000418 10.1.20.103:5060 -> 10.1.20.103:5060 REFER sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> * No Contact!*! # U +0.005133 10.1.20.103:5060 -> 10.1.3.15:5061 REFER sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> # U +0.046925 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000520 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="10.1.20.103", nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813" Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7162 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> via_cnt==1" # U +0.004377 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000193 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000286 10.1.20.103:5060 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.003618 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.005223 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 ACK Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.010954 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> ",algorithm=MD5,response="2baeb40875c0cc7040857daa6e8192fa" Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001353 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7159 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.3.18:5061 via_cnt==1" # U +0.000226 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:112@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.007697 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000280 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.012355 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.034919 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000260 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.032160 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.002343 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.461009 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.028104 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.760273 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000246 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.022157 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.009321 10.1.20.103:5060 -> 10.1.3.18:5061 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.447617 10.1.3.18:5061 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.007101 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 69 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.014754 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000198 10.1.20.103:5060 -> 10.1.3.18:5061 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0

3 4

sipp and stateful transaction problem
by JR Richardson 25 May '10

25 May '10

Hi All, I'm doing some testing with kamailio 1.5: kamailio1:/etc/kamailio# kamailio -V version: kamailio 1.5.4-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:6005M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 Using dispatcher module trying to load balance SIP calls across some Asterisk servers. I have it working fine when I test in this scenario: sip phone dial out><asterisk><kamailio><round robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial out><kamailio><round robin to several asterisk servers I get retransmits on every call back to sipp with errors like "Discarding message which can't be mapped to a known SIPp call" and "SIP/2.0 481 Call leg/transaction does not exist" This happens with kamailio setup stateful or stateless. I'm wondering if sipp is the problem or just doesn't play well with kamailio? I've kept the config as simple as possible for testing, it is listed here http://pastebin.com/BZ8hJvJv Here is my sipp usage: sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 -trace_err Any insight would be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

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