June 2009 - sr-users-es

[OpenSER-Users-ES] Instalacción Erronea
by Javier Allende Astigarraga 22 Oct '25

22 Oct '25

Bueno pos tras cargar mi memoria virtual intento arrancar Openser y me dice que el modulo de mysql.so esta obsoleto con lo cual me quedo como asi :| , pero bueno pienso habra sido problema de la instalación de TLS que tendria que haber añadido el modulo de mysql por realizo otra instalación (sin hacer clean,eso lo voy a a probar ahora),me lo instala pero no me arranca openser me dice listening : Aliases: y de repente la petada del mi_fifo que si lo tengo puesto en el openser.cfg y en openserctrl . alguna idea?? -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

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[Kamailio-Users-ES] Asterisk Virtualizado - Sirve Kamailio?
by sadzas 15 Jul '09

15 Jul '09

Hola a todos... Supongan el siguiente escenario (que es en realidad lo que tengo funcionando)... Tengo 3 asterisk virtualizados. Un telefono en cada * virtualizado. El tema es que cada telefono apunta a una IP diferente. Esto es: Asterisk-1 : 192.168.2.1 Asterisk-2 : 192.168.2.2 Asterisk-3 : 192.168.2.3 Cada * tiene diferentes features, como voicemail, meetme, etc etc. Me interesa unificar esto, para que cada telefono apunte a sola una IP. Y luego de acuerdo al interno, lo pueda desviar hacia el * indicado. Todo lo que quiero hacer es un bypassing. Siempre sabiendo que las IPs de los telefonos que se van a registrar son dinamicas y cambian constantemente. Se puede utilizar OpenSer para esto? El problema que le encuentro es que con OpenSer tendria que compartir la BD con Asterisk, o ir a buscar los datos ahi, pero no me imagino como hacer con 3 BD diferentes. mas aun, si asi fuera, me queda la eterna duda de como hacer que la llamada vuelva al telefono desde el Asterisk pasando por OpenSer. Para graficarlo un poco: Asterisk-1 \ Asterisk-2 - IP ------> Telefonos de diferentes * Asterisk-3 / La verdad es que no me imagino con que hacer esto... les agradeceria toda ayuda posible... gracias a todos! -- View this message in context: http://www.nabble.com/Asterisk-Virtualizado---Sirve-Kamailio--tp24185223p24… Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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Re: [SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
by rubenrojas - Trc.es 01 Jul '09

01 Jul '09

Gracias por la pronta respuesta, asi da gusto unirse a una comunidad, he colocado varios xlog en onreply_route[1] y lo he dejado asi: onreply_route[1] { xdbg("incoming reply\n"); xlog("L_INFO", "*** onreply_route antes de forzar rtpproxy ***"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { xlog("L_INFO", "*** rtpproxy forzado en onreply_route ***"); force_rtp_proxy(); } if (isbflagset(6)) { xlog("L_INFO", "*** fix_nated_contact en onreply_route ***"); fix_nated_contact(); } } haciendo un tail -f /var/log/syslog veo lo siguiente: Jun 25 13:39:51 kamailio /usr/local/sbin/kamailio[22088]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:51 kamailio /usr/local/sbin/kamailio[22087]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22088]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22088]: *** rtpproxy forzado en onreply_route *** Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22088]: ACC: transaction answered: timestamp=1245929995;method=INVITE;from_tag=5c805948a0455e5b;to_tag=c0a80101-3569b;call_id=d1b81cbb5fbb9e6e@192.168.254.110;code=200;reason=OK;src_user=20000004;src_domain=212.4.107.250;dst_ouser=20000000;dst_user=20000000;dst_domain=192.168.254.101 Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22089]: ACC: request acknowledged: timestamp=1245929995;method=ACK;from_tag=5c805948a0455e5b;to_tag=c0a80101-3569b;call_id=d1b81cbb5fbb9e6e@192.168.254.110;code=200;reason=OK;src_user=20000004;src_domain=212.4.107.250;dst_ouser=20000000;dst_user=20000000;dst_domain=192.168.254.101 Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22088]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:55 kamailio /usr/local/sbin/kamailio[22088]: *** rtpproxy forzado en onreply_route *** Jun 25 13:39:56 kamailio /usr/local/sbin/kamailio[22089]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:56 kamailio /usr/local/sbin/kamailio[22089]: *** rtpproxy forzado en onreply_route *** Jun 25 13:39:58 kamailio /usr/local/sbin/kamailio[22087]: *** onreply_route antes de forzar rtpproxy *** Jun 25 13:39:58 kamailio /usr/local/sbin/kamailio[22087]: *** rtpproxy forzado en onreply_route *** por lo que veo si hace el force_rtp_proxy() pero no hace nunca el fix_nated_contact() en onreply_route ----- Mensaje original ----- De: "Iñaki Baz Castillo" <ibc(a)aliax.net> Para: users-es(a)lists.kamailio.org Enviados: Jueves, 25 de Junio de 2009 13:25:18 GMT +01:00 Amsterdam / Berlín / Berna / Roma / Estocolmo / Viena Asunto: Re: [SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy) El Jueves, 25 de Junio de 2009, Iñaki Baz Castillo escribió: > El Jueves, 25 de Junio de 2009, rubenrojas - Trc.es escribió: > > # Caller NAT detection route > > /* uncomment the whole following route for enabling Caller NAT Detection > > */ route[4]{ > > force_rport(); > > if (nat_uac_test("19")) { > > if (method=="REGISTER") { > > fix_nated_register(); > > } else { > > fix_nated_contact(); > > } > > setflag(5); > > } > > return; > > } > > Lo anterior NO se está ejecutando. Fíjate que el Contact del 200 Ok llega > con la IP provada al llamante, por lo que el ACK que envía llega al proxy > con el RURI conteniendo una IP privada del llamado, así que el proxy ruta > el ACK a la IP privada (que obiviamente no llega). Por eso no oyes audio, > porque la negociación del INVITE no acaba (no llega el ACK al llamado, y > por eso hay tantas retransmissiones del 200 OK que no pareces haber > observado). Perdona, lo de arriba sí se está ejecutando (el Contact del INVITE cambia). Lo que no cambia es el de la respuesta, o sea, no se ejecuta esto: onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } Lo que no sé es porqué no se ejecuta puesto que tienes puesto en t_onreply("1") en route[1]. Eso tendrás que descubrirlo a base de poner "xlog" para ver si se están eejcutando los bloques o no. Nadie garantiza que el config file que viene por defecto sea 100% correcto. -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ SR-Users-ES mailing list SR-Users-ES(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es _________________________________________________________________ Este mensaje ha sido escaneado en busca de virus por MailScanner.

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Re: [SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
by rubenrojas - Trc.es 25 Jun '09

25 Jun '09

Gracias Iñaki ya lo he solucionado (temporalmente), de hecho hay dos bugs en la configuracion por defecto el 1ro en el failure_route[1] la linea if (is_method("INVITE") viene sin la segunda comilla, osea originalmente es asi: if (is_method("INVITE) al activar el nat kamailio no inicia hasta que no se le coloque la comilla que faltaba el 2do (el que nos trajo hasta aqui): en onreply_route[1] no llega el flag 6 activo de momento he colocado el fix_nated_contact(); en el mismo bloque que el force_rtp_proxy() y va perfecto si veis donde podra estar la perdida de este flag, por favor colocarlo aqui en la lista para dejar la configuracion por defecto del kamailio lo mejor posible muchas gracias ----- Mensaje original ----- De: "Iñaki Baz Castillo" <ibc(a)aliax.net> Para: users-es(a)lists.kamailio.org Enviados: Jueves, 25 de Junio de 2009 13:44:55 GMT +01:00 Amsterdam / Berlín / Berna / Roma / Estocolmo / Viena Asunto: Re: [SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy) El Jueves, 25 de Junio de 2009, rubenrojas - Trc.es escribió: > por lo que veo si hace el force_rtp_proxy() pero no hace nunca el > fix_nated_contact() en onreply_route Sí, como dije el problema no es el SDP de las respuestas (la IP y puerto se ven claramente modificados por el proxy) sino en el Contact. Pues te toca investigar lo mismo pero con el fix_nated_contact (añade xlogs a porrillo). Tal vez has descubierto un bug en el config por defecto de Kamailio. -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ SR-Users-ES mailing list SR-Users-ES(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-es _________________________________________________________________ Este mensaje ha sido escaneado en busca de virus por MailScanner.

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No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
by rubenrojas - Trc.es 25 Jun '09

25 Jun '09

Hola, este es mi primer post en esta lista, Tengo instalado el kamailio 1.5.1 y estoy utilizando un kamailio.cfg utilizando el que trae por defecto, luego he modificado el cfg para activar mysql, domain, presence, nathelper y authentication con md5, todo funciona como se supone que deberia, los clientes pueden registrarse, enviar mensajes de texto y hablar entre ellos. El unico problema es el audio cuando dos clientes estan detras de una NAT, los telefonos pueden realizar la llamada y suena el ring, pero cuando se descuelga no hay audio en ninguna direccion. cuendo los telefonos tienen una IP publica todo funciona bien, tambien funciona cuando utilizo un Linksys PAP2T con las opciones "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" y "NAT mapping enable" encendidas, con el softphone de Qutecom stambien funciona. Este problema me esta sucediendo con los telefonos fisicos Thomson phones (model ST 2022) y GrandStream Budge Tone 200, este problema ocurre sin importar que opciones le coloque para el tipo de nateo dentro de los telefonos, Incluso he utilizado stun con stunserver.org o con el servidor de stun de ekiga, los telefonos se registran y pueden hacer y recibir llamadas, pero no hay audio cuando atiendes la llamada. Con kamctl ul show, puedes ver que han registrado el Contact con su IP local y el Received con la IP publica y los puertos para el NAT La unica diferencia con los Linksys que si funcionan es que los Linksys registran el Contact con la IP publica. Aqui se puede ver dos telefonos con NAT en el proxy Domain:: location table=512 records=2 max_slot=1 AOR:: 20000004(a)212.4.107.250 Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q= Expires:: 1150 Callid:: 72ed03f6d2f390f9(a)192.168.254.110 Cseq:: 10003 User-agent:: Grandstream BT200 1.1.6.27 Received:: sip:212.4.97.115:35379 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 7807 AOR:: 20000000(a)212.4.107.250 Contact:: sip:20000000@192.168.254.101:5060;user=phone Q= Expires:: 2945 Callid:: 17fe-c0a80101-5-1(a)192.168.254.101 Cseq:: 6 User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06 Received:: sip:212.4.97.115:55128 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 4294967295 Estoy utilizando rtpproxy y no hay ningun error en el log que indique que el rtpproxy no esta funcionando, de hecho haciendo un SIP trace muestra al rtpproxy seteando puertos para el audio. Ejecuto el rtpproxy con este comando: rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F Cualquier ayuda sera apreciada, llevo dos semanas buscando una solucion Adjunto mi kamailio.cfg para que puedan mirarlo, al final de este mensaje voy a adjuntar el SIP Trace de una llamada entre dos telefonos detras de una NAT (un Thomson y un GrandStream) en caso que puedan ayudarme a decifrar que esta mal aqui: este es mi cfg ************************************************************************************************** # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # 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. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.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" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* 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" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so"#manually added loadmodule "nathelper.so" # ----------------- 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=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #a#modparam("auth_db", "calculate_ha1", yes) #a#modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") #parametros de autentificacion modificados manualmente modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "use_domain", 0) modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque utilizaremos multi-domain modparam("auth_db", "load_credentials", "rpid") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- 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 ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:212.4.107.250:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@212.4.107.250") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("nathelper", "sipping_method", "OPTIONS") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } ##if (!uri==myself) /* replace with following line if multi-domain support is used */ if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) { route(2); } if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); setbflag(7);# sipping } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { 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; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } # Auth only on registration #a# } else { #a# # authenticate if from local subscriber (uncomment to enable auth) #a# if (from_uri==myself) #a# { #a# if (!proxy_authorize("", "subscriber")) { #a# proxy_challenge("", "0"); #a# exit; #a# } #a# if (is_method("PUBLISH")) #a# { #a# if ($au!=$tU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } else { #a# if ($au!=$fU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } #a# #a# consume_credentials(); #a# # caller authenticated #a# } #a# } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { 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(); ##} } ************************************************************************************************** ************************************************************************************************** Aqui va el SIP Trace para una llamada de telefonos fisicos NATed to NATed: ************************************************************************************************** U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060 INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 332 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 192.168.254.110 t=0 0 m=audio 40000 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 # U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 # U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128 INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 348 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 212.4.107.250 t=0 0 m=audio 35752 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 a=nortpproxy:yes # U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Content-Length: 0 # U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797 Route: <sip:212.4.107.250;lr=on;nat=yes> From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes

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[Kamailio-Users-ES] Consulta Basica acerca de kamailio
by sadzas 23 Jun '09

23 Jun '09

Hola a todos. Con el modulo Textops, puedo cambiar partes de la senalizacion, reemplazando, agregando, en resumen, editando. Me interesa crear de 0 un INVITE. Por ejemplo: Hago esto: if (is_method("INVITE") && $rU =~ "loquesea"){ force_rtp_proxy(); t_on_failure("1"); route(x); }; route[x] { log(1, "Reenvia creando un INVITE \n"); --->CREAR EL INVITE<----- t_relay("udp:adondequiero:5060"); t_on_reply("1"); exit; } Y poner en el Invite a modo de variables, los valores que quiero, por ejemplo: numA= loquesea numB= adondequiero "SIP to address: sip:numA@numB" No se si me explique bien... pretendo crear un INVITE desde 0 utilizando una especie de plantilla de senalizacion donde le aregaria los valores de acuerdo a mi criterio... tal vez se podria levantar esta plantilla desde un TXT, o algo parecido... es esto posible? Muchas gracias. -- View this message in context: http://www.nabble.com/Consulta-Basica-acerca-de-kamailio-tp24147809p2414780… Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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