sr-users-es March 2009

sr-users-es@lists.kamailio.org

7 participants
9 discussions

[Kamailio-Users-ES] Problema con kamailio 1.5 y freeradius (y posible solución)
by toni cunyat 27 Mar '09

27 Mar '09

Buenas a tod@s. Resulta que ayer decidí migrar de kamailio 1.4.2 a kamailio 1.5 (espero probar la interfaz web en un momento). Después de hacer los ajustes necesarios (basicamente solo tuve que cambiar algun loadmodule) me encuentro que el kamailio no se inicia. Da el siguiente error en el modulo radius: Mar 27 09:00:52 admundsen /usr/local/kamailio/sbin/kamailio[10526]: ERROR:auth_radius:mod_init: auth_radius: can't get code for the Cisco-AVPair attribute Mar 27 09:00:52 admundsen /usr/local/kamailio/sbin/kamailio[10526]: ERROR:core:init_mod: failed to initialize module auth_radius Mar 27 09:00:52 admundsen /usr/local/kamailio/sbin/kamailio[10526]: ERROR:core:main: error while initializing modules Básicamente me dice que no puede coger los codigos para el atributo "Cisco-AVPair", así que googleo un poco y basicamente lo unico que encuentro es que he de cambiar el source. No es que me moleste, pero lo tendria que hacer para cada versión. Para solucionarlo, lo que he hecho es coger el diccionario dictionary.cisco que esta en el freeradius y copiarlo en equipo donde esta el kamailio. Posteriormente he hecho un include en /etc/radiusclient-ng/dictionary para añadirselo al cliente radius: $INCLUDE /etc/radiusclient-ng/dictionary.cisco Así me ha funcionado, y ahora ya inicia el kamailio. No sé si podria decirse que es un bug, no lo he reportado (mi ingles es pésimo) Voy a probar la interfaz web. Un saludo.

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[Kamailio-Users-ES] Llamadas de openser a asterisk solo funcionan en un sentido
by Laura10 18 Mar '09

18 Mar '09

Hola, tengo montado el siguiente escenario: pjsua1 ----- openser1 ----- asterisk ---- openser2 ----- pjsua2 En pjsua1 tengo una cuenta registrada en opensip1: juan(a)192.168.1.54 En pjsua2 tengo una cuenta registrada en opensip2: pepe(a)192.168.2.56 Entre cada openser y asterisk tengo creada una troncal SIP: troncal1 para openser1 y troncal2 para openser2. Y los ficheros de configuración de asterisk los tengo así: sip.conf [general] #Registro ante openser register=>troncal1:troncal1@192.168.1.54/troncal1 register=>troncal2:troncal2@192.168.2.56/troncal2 #Extensión que funciona como troncal SIP [troncal1] type=friend secret=troncal1 username=troncal1 password=troncal1 fromuser=troncal1 fromdomain=192.168.1.54 host=192.168.1.54 context=from-openser insecure=invite [troncal2] type=friend secret=troncal2 username=troncal2 password=troncal2 fromuser=troncal2 fromdomain=192.168.2.56 host=192.168.2.56 context=from-openser insecure=invite #Usuarios registrados en openser [pepe] username=pepe host=192.168.2.56 qualify=no context=from-openser [juan] username=juan host=192.168.1.54 qualify=no context=from-openser extensions.conf [from-openser] exten=>troncal1,1,Dial(SIP/troncal1) exten=>troncal2,1,Dial(SIP/troncal2) exten=>pepe,1,Dial(SIP/pepe(a)192.168.2.56,30,Ttm) exten=>pepe,2,Hangup exten=>juan,1,Dial(SIP/juan(a)192.168.1.54,30,Ttm) exten=>juan,2,Hangup El problema que tengo es que las llamadas o funcionan de pjsua1 a pjsua2 o de pjsua2 a pjsua1, y el sentido en el que funcionan depende de que openser arranco primero. Si es el openser2 van de pjsua2 a pjsua1 y si es el openser1 es de pjsua1 a pjsua2. Yo quiero que las llamadas funcionen en los dos sentidos sin problemas. Viendo el fichero de log de asterisk en ambos sentidos y comparándolo veo las siguientes diferencias: Llamada OK: logger.c: No user 'pepe' in SIP users list logger.c: Found peer for 'troncal2' from '192.168.2.56:5060' logger.c: Looking for juan in from_openser (domain 192.168.1.54) Llamada fallida: logger.c: No user 'juan' in SIP users list logger.c: No matching peer for 'juan' from '192.168.1.54:1053' logger.c: Looking for pepe in default (domain 192.168.2.56) De lo que deduzco que no detecta bien la troncal1, aunque al hacer sip show registry en asterisk aparece correctamente registrada, y que como busca al usuario 'pepe' en el contexto default y este no aparece, por eso devuelve el mensaje '404 Not Found'. He hecho el siguiente cambio para ver si funcionaba: Añadir en extensions.conf [default] include => from_openser Entonces la llamada ya funciona en los dos sentidos porque en default si que aparece el usuario buscado. Sin embargo esta solución no me parece la adecuada, y lo que interesa es que detecte correctamente las dos troncales para que funcione sin problemas en los dos sentidos, y a los usuarios los busque en el contexto adecuado y no en default. El problema es que no sé como hacerlo, si alguien me puede ayudar se lo agradezco de antemano. -- View this message in context: http://www.nabble.com/Llamadas-de-openser-a-asterisk-solo-funcionan-en-un-s… Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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[Kamailio-Users-ES] Consulta un fichero con openser
by Laura10 18 Mar '09

18 Mar '09

Hola, soy novata usando openser, y me cuesta mucho configurarlo. Lo que trato de hacer es: consultar un fichero de texto para saber cúal es el estado de la red, y en función de lo que aparezca en ese fichero, enviar un mensaje de error para que no se establezca la comunicación, pero tengo problemas. Para usar ficheros de texto necesito el módulo db_text y para consultarlo he usado el módulo avpops, pero no se utilizarlo y me da problemas. El fichero que quiero leer es muy simple: estado(int) 1 Y mi fichero de configuración es: # # $Id: opensips.cfg 4423 2008-06-27 10:25:01Z henningw $ # # OpenSIPS basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### 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/opensips/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/opensips/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/opensips/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want opensips 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/opensips/modules/" /* uncomment next line for MySQL DB support */ #loadmodule "db_mysql.so" loadmodule "avpops.so" loadmodule "db_text.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.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" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_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) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ #modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") modparam("avpops", "db_url", "db_text:///usr/local/share/opensips/dbtext/opensi$ modparam("avpops", "avp_table", "avptable") modparam("db_text", "db_mode", 1) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") #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://opensips:opensipsrw@localhost/opensips") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } 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 { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) ##if (!(method=="REGISTER") && from_uri==myself) ##{ ## if (!proxy_authorize("", "subscriber")) { ## proxy_challenge("", "0"); ## exit; ## } ## if (!check_from()) { ## sl_send_reply("403","Forbidden auth ID"); ## exit; ## } ## ## consume_credentials(); ## # caller authenticated ##} # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain /* uncomment this if you want to enable presence server and comment the next 'if' block NOTE: uncomment also the definition of route[2] from below */ ##if( is_method("PUBLISH|SUBSCRIBE")) ## route(2); if (is_method("PUBLISH")) { sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) ##if (!www_authorize("", "subscriber")) ##{ ## www_challenge("", "0"); ## exit; ##} ## ##if (!check_to()) ##{ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} if (!save("location")) sl_reply_error(); exit; } if ($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 ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); route(3); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { avp_db_query("SELECT estado FROM ejemplo", "$avp(s:estado)"); if (!avp_check("$avp(s:estado)","re/(0|1)/g")) { sl_send_reply("404", "Not Found"); exit; } else if ($avp(s:estado)=="0") { sl_send_reply("404", "Not found"); exit; } } branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } } Resalto en negrita lo que he modificado respecto al fichero original. Y al arrancar openser obtengo en el fichero de log lo siguiente: WARNING:core:fix_socket_list: could not rev. resolve 192.168.1.54 INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) NOTICE:core:main: version: opensips 1.4.4-notls (i386/linux) INFO:core:main: using 32 Mb shared memory INFO:core:main: using 1 Mb private memory per process INFO:avpops:avpops_init: initializing... ERROR:core:db_check_api: module db_db_text does not export db_use_table function CRITICAL:avpops:avpops_db_bind: cannot bind to database module! Did you load a database module ? ERROR:core:init_mod: failed to initialize module avpops ERROR:core:main: error while initializing modules Si alguien me puede ayudar se lo agradecería. -- View this message in context: http://www.nabble.com/Consulta-un-fichero-con-openser-tp22563438p22563438.h… Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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[Kamailio-Users-ES] Pregunta Conceptual!
by sadzas 17 Mar '09

17 Mar '09

Pude retomar esto que deje hace mucho tiempo en Stand By... Y tengo una pregunta conceptual: El esquema es el siguiente: PBX (Propietaria) ---- Asterisk ----- Kamailio ------ Phone (que esta detras de un router Linksys mediante NAT) La extension en cuestion es la 6001, la cual esta configurada en Asterisk como: Host: IP kamailio Nat: yes Qualify: no En kamailio esta correctamente registrada, con los flags correctos reconociendo el NAT. (Si llamo de una extension registrada en kamailio a esta extension, la comunicacion es correcta) Lo que sucede es lo siguiente: Cuando desde una extension de la PBX propietaria disco el numero 6001, esto llega al asterisk, el asterisk lo reenvia al kamailio, y kamailio toma el INVITE y lo reenvia a la direccion IP Privada del telefono. Para que quede mas claro: IP PBX propietaria: 192.168.2.1 IP Asterisk: 200.xx.xx.87 IP Kamailio: 200.xx.xx.53 IP privada telefono: 192.168.2.10 U 200.xx.xx.87:5060 -> 200.xx.xx.53:5060 INVITE sip:6001@192.168.2.10:5060 SIP/2.0. Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK66c318cc;rport. From: "Propietario" <sip:220@200.xx.xx.87>;tag=as1f07b3ea. To: <sip:6001@192.168.2.10:5060>. Contact: <sip:220@200.xx.xx.87>. Call-ID: 7fb490c8634757322217b63e51bd0f12(a)200.xx.xx.87. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. INVITE sip:6001@192.168.2.10:5060 SIP/2.0. Record-Route: <sip:200.xx.xx.53;lr=on;ftag=as1f07b3ea>. Via: SIP/2.0/UDP 200.xx.xx.53;branch=z9hG4bK890f.87c7d145.0. Via: SIP/2.0/UDP 200.xx.xx.87:5060;branch=z9hG4bK66c318cc;rport=5060. From: "Propietario" <sip:220@200.xx.xx.87>;tag=as1f07b3ea. To: <sip:6001@192.168.2.10:5060>. Contact: <sip:220@200.xx.xx.87>. Call-ID: 7fb490c8634757322217b63e51bd0f12(a)200.xx.xx.87. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Por que hace este INVITE a la IP privada? y no toma la Publica haciendo correctamente el NAT? Gracias. -- View this message in context: http://www.nabble.com/Pregunta-Conceptual%21-tp22567852p22567852.html Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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[Kamailio-Users-ES] Llamadas a IP's privadas sin NAT
by toni cunyat 17 Mar '09

17 Mar '09

Buenas a todos. En mi empresa hay montado un Callmanager con IP's privadas , pero yo al margen monté un kamailio con IP publica para que los usuarios se puedan registrar y hacer videoconferencias. El tema de la presencia solo lo he conseguido hacer funcionar con el X-Lite. Ahora la idea que tienen es unir ambas cosas, para lo cual en otro PC monté un asterisk con un trunk SIP al callmanager (version 4.1 de callmanager) y en el kamailio le digo que lo reenvie al asterisk (que dispone de IP publica e IP privada). También en la maquina que esta el asterisk, le he instalado el rtpproxy, quedandose el siguiente montaje: |----Ekiga Linux-----| | kamailio | | asterisk | | Callmanager | |Cliente (IP Publica|----->| IP Publica |------->|IP Publica IP--- Privada|------| IP Privada | |-------------------------| | rewritehostport("IP_asterisk:5060");| | rtpproxy | | | No quiero hacer NAT. Las pruebas de momento no me funcionan, lo que no sé es si el esquema es correcto y deberia funcionar, y así centrarme en la configuración. ¿Estoy metiendo la pata en el esquema? ¿Me faltan elementos? Gracias

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[Kamailio-Users-ES] Interconectar Kamailio con callmanager con IP privadas sin NAT
by toni cunyat 17 Mar '09

17 Mar '09

Buenas a todos. En mi empresa hay montado un Callmanager con IP's privadas , pero yo al margen monté un kamailio con IP publica para que los usuarios se puedan registrar y hacer videoconferencias. Ahora la idea que tienen es unir ambas cosas, para lo cual en otro PC monté un asterisk con un trunk SIP al callmanager (version 4.1 de callmanager) y en el kamailio le digo que lo reenvie a lPC con asterisk (que dispone de IP publica e IP privada). También en la maquina que esta el asterisk, le he instalado el rtpproxy, quedandose el siguiente montaje: http://s1.subirimagenes.com/imagen/previo/thump_1982023asterisk.png No quiero hacer NAT. Las pruebas de momento no me funcionan, lo que no sé es si el esquema es correcto y deberia funcionar, y así centrarme en la configuración. ¿Estoy metiendo la pata en el esquema? ¿Me faltan elementos? Gracias a todos.

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[Kamailio-Users-ES] Consulta un fichero con openser
by Laura10 17 Mar '09

17 Mar '09

Hola, soy novata usando openser, y me cuesta mucho configurarlo. Lo que trato de hacer es: consultar un fichero de texto para saber cúal es el estado de la red, y en función de lo que aparezca en ese fichero, enviar un mensaje de error para que no se establezca la comunicación, pero tengo problemas. Para usar ficheros de texto necesito el módulo db_text y para consultarlo he usado el módulo avpops, pero no se utilizarlo y me da problemas. El fichero que quiero leer es muy simple: estado(int) 1 Y mi fichero de configuración es: # # $Id: opensips.cfg 4423 2008-06-27 10:25:01Z henningw $ # # OpenSIPS basic configuration script # by Anca Vamanu # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### 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/opensips/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/opensips/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/opensips/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want opensips 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/opensips/modules/" /* uncomment next line for MySQL DB support */ #loadmodule "db_mysql.so" loadmodule "avpops.so" loadmodule "db_text.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.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" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/opensips_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) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ #modparam("usrloc", "db_mode", 2) #modparam("usrloc", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") modparam("avpops", "db_url", "db_text:///usr/local/share/opensips/dbtext/opensi$ modparam("avpops", "avp_table", "avptable") modparam("db_text", "db_mode", 1) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://opensips:opensipsrw@localhost/opensips") #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://opensips:opensipsrw@localhost/opensips") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } 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 { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) ##if (!(method=="REGISTER") && from_uri==myself) ##{ ## if (!proxy_authorize("", "subscriber")) { ## proxy_challenge("", "0"); ## exit; ## } ## if (!check_from()) { ## sl_send_reply("403","Forbidden auth ID"); ## exit; ## } ## ## consume_credentials(); ## # caller authenticated ##} # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain /* uncomment this if you want to enable presence server and comment the next 'if' block NOTE: uncomment also the definition of route[2] from below */ ##if( is_method("PUBLISH|SUBSCRIBE")) ## route(2); if (is_method("PUBLISH")) { sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) ##if (!www_authorize("", "subscriber")) ##{ ## www_challenge("", "0"); ## exit; ##} ## ##if (!check_to()) ##{ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} if (!save("location")) sl_reply_error(); exit; } if ($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 ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); route(3); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { avp_db_query("SELECT estado FROM ejemplo", "$avp(s:estado)"); if (!avp_check("$avp(s:estado)","re/(0|1)/g")) { sl_send_reply("404", "Not Found"); exit; } else if ($avp(s:estado)=="0") { sl_send_reply("404", "Not found"); exit; } } branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } } Resalto en negrita lo que he modificado respecto al fichero original. Y al arrancar openser obtengo en el fichero de log lo siguiente: WARNING:core:fix_socket_list: could not rev. resolve 192.168.1.54 INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) NOTICE:core:main: version: opensips 1.4.4-notls (i386/linux) INFO:core:main: using 32 Mb shared memory INFO:core:main: using 1 Mb private memory per process INFO:avpops:avpops_init: initializing... ERROR:core:db_check_api: module db_db_text does not export db_use_table function CRITICAL:avpops:avpops_db_bind: cannot bind to database module! Did you load a database module ? ERROR:core:init_mod: failed to initialize module avpops ERROR:core:main: error while initializing modules Si alguien me puede ayudar se lo agradecería. -- View this message in context: http://www.nabble.com/Consulta-un-fichero-con-openser-tp22563432p22563432.h… Sent from the OpenSER Users - ES mailing list archive at Nabble.com.

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[Kamailio-Users-ES] ayuda migrando de openser 1.3x a 1.4x opensips
by troxlinux 13 Mar '09

13 Mar '09

señores tengo algo peculiar que me esta pasando , estoy tratando de migrar mi proxy de la version 1.3x a opensip 1.4.4 , he estado leyendo y se que algunas cosas cambiaron pero lo raro es que cuando llamo de una extensión a otro a me salta directo al voicemail como si estuviera ocupado ... y eso hablando de extensiones en la misma red , no externas ... este pegon si lo tengo duro , no se por donde buscarle ... disculpen que añada este trozote de log sip ... interface: any filter: (ip) and ( port 5060 ) # U +1.038668 192.168.10.19:5060 -> 192.168.10.3:5060 INVITE sip:201@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3> Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:200@192.168.10.19:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 47854 47854 IN IP4 192.168.10.19 s=- c=IN IP4 192.168.10.19 t=0 0 m=audio 16402 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000438 192.168.10.3:5060 -> 192.168.10.19:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.62fe Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="192.168.10.3", nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66" Server: OpenSIPS (1.4.4-notls (i386/linux)) Content-Length: 0 # U +0.015654 192.168.10.19:5060 -> 192.168.10.3:5060 ACK sip:201@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-95de5398 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.62fe Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 101 ACK Max-Forwards: 70 Contact: <sip:200@192.168.10.19:5060> User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 # U +0.003845 192.168.10.19:5060 -> 192.168.10.3:5060 INVITE sip:201@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3> Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a" Contact: <sip:200@192.168.10.19:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 47854 47854 IN IP4 192.168.10.19 s=- c=IN IP4 192.168.10.19 t=0 0 m=audio 16402 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001341 192.168.10.3:5060 -> 192.168.10.19:5060 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3> Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE Server: OpenSIPS (1.4.4-notls (i386/linux)) Content-Length: 0 # U +0.000131 192.168.10.3:5060 -> 192.168.10.3:5070 INVITE sip:201@192.168.10.3:5070 SIP/2.0 Record-Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.0 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3> Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE Max-Forwards: 69 Proxy-Authorization: Digest username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a" Contact: <sip:200@192.168.10.19:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 47854 47854 IN IP4 192.168.10.19 s=- c=IN IP4 192.168.10.19 t=0 0 m=audio 16402 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000688 192.168.10.3:5070 -> 192.168.10.3:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.0;received=192.168.10.3 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 Record-Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3> Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:201@192.168.10.3:5070> Content-Length: 0 # U +0.000598 192.168.10.3:5070 -> 192.168.10.3:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.0;received=192.168.10.3 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 Record-Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=as4a245f87 Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:201@192.168.10.3:5070> Content-Type: application/sdp Content-Length: 238 v=0 o=root 9235 9235 IN IP4 192.168.10.3 s=session c=IN IP4 192.168.10.3 t=0 0 m=audio 17814 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.000278 192.168.10.3:5060 -> 192.168.10.19:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-82527624 Record-Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=as4a245f87 Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:201@192.168.10.3:5070> Content-Type: application/sdp Content-Length: 238 v=0 o=root 9235 9235 IN IP4 192.168.10.3 s=session c=IN IP4 192.168.10.3 t=0 0 m=audio 17814 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.018455 192.168.10.19:5060 -> 192.168.10.3:5060 ACK sip:201@192.168.10.3:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-c1874d9d From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=as4a245f87 Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> Proxy-Authorization: Digest username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a" Contact: <sip:200@192.168.10.19:5060> User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 # U +0.000206 192.168.10.3:5060 -> 192.168.10.3:5070 ACK sip:201@192.168.10.3:5070 SIP/2.0 Record-Route: <sip:192.168.10.3;lr=on;ftag=327eaf6c2c6324f1o0> Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK5afe.11d6393.2 Via: SIP/2.0/UDP 192.168.10.19:5060;branch=z9hG4bK-c1874d9d From: <sip:200@192.168.10.3>;tag=327eaf6c2c6324f1o0 To: "kamailio-14x" <sip:201@192.168.10.3>;tag=as4a245f87 Call-ID: acc6c148-4348dcc5(a)192.168.10.19 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="200",realm="192.168.10.3",nonce="49b6c7130000001145db79bf52de222d013769f25c8daa66",uri="sip:201@192.168.10.3",algorithm=MD5,response="f61b7ab5b597daea13911d7630e9dc4a" Contact: <sip:200@192.168.10.19:5060> User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 # U +4.600183 192.168.10.3:5070 -> 192.168.10.3:5060 OPTIONS sip:192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK60d4a50f;rport From: "asterisk" <sip:asterisk@192.168.10.3:5070>;tag=as5417e9bc To: <sip:192.168.10.3> Contact: <sip:asterisk@192.168.10.3:5070> Call-ID: 2ec730764cddd1bf15e296f168d9b0bf(a)192.168.10.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 10 Mar 2009 20:00:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000269 192.168.10.3:5060 -> 192.168.10.3:5070 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK60d4a50f;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.3:5070>;tag=as5417e9bc To: <sip:192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.0e65 Call-ID: 2ec730764cddd1bf15e296f168d9b0bf(a)192.168.10.3 CSeq: 102 OPTIONS Proxy-Authenticate: Digest realm="192.168.10.3", nonce="49b6c717000000127f72fab6bd7f924f5d24d09098273864" Server: OpenSIPS (1.4.4-notls (i386/linux)) Content-Length: 0 mi segundo problemilla es con los permisos tengo algunas usuarios que hacer llamadas a la pstn y otros a algunos proveedores voip , pero esto no lo respeta el proxy cualquiera puede llamar a la pstn etc... y la extension 201 no tiene permisos para las llamadas locales , tengo el modulo group.so cargado y el permissions.so dentro del opensips.cfg tengo añadido este trozo de codigo if (uri=~"^sip:[2346597][0-9]{6}@.*") { # if (is_user_in("From", "local")){ if (is_user_in("credentials", "local")){ route(4); y veo que esto se mantiene en esta version .... añado algunos log U +0.963418 192.168.10.30:5064 -> 192.168.10.3:5060 INVITE sip:2685249@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac From: "Opensips-14x" <sip:201@192.168.10.3>;tag=f89d240daba57386 To: <sip:2685249@192.168.10.3> Contact: <sip:201@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Call-ID: 52934087b53c9f07(a)192.168.10.30 CSeq: 20723 INVITE User-Agent: Grandstream GXP2020 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=201 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5050 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000418 192.168.10.3:5060 -> 192.168.10.30:5064 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac From: "Opensips-14x" <sip:201@192.168.10.3>;tag=f89d240daba57386 To: <sip:2685249@192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.00fb Call-ID: 52934087b53c9f07(a)192.168.10.30 CSeq: 20723 INVITE Proxy-Authenticate: Digest realm="192.168.10.3", nonce="49b6c5f00000000c81c829543f443c845ee6c9514213aa27" Server: OpenSIPS (1.4.4-notls (i386/linux)) Content-Length: 0 # U +0.022859 192.168.10.30:5064 -> 192.168.10.3:5060 ACK sip:2685249@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK8cb08d7b02da46ac From: "Opensips-14x" <sip:201@192.168.10.3>;tag=f89d240daba57386 To: <sip:2685249@192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.00fb Contact: <sip:201@192.168.10.30:5064;transport=udp> Supported: path Call-ID: 52934087b53c9f07(a)192.168.10.30 CSeq: 20723 ACK User-Agent: Grandstream GXP2020 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.018038 192.168.10.30:5064 -> 192.168.10.3:5060 INVITE sip:2685249@192.168.10.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKd5d3f9baf951bf4e From: "Opensips-14x" <sip:201@192.168.10.3>;tag=f89d240daba57386 To: <sip:2685249@192.168.10.3> Contact: <sip:201@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="201", realm="192.168.10.3", algorithm=MD5, uri="sip:2685249@192.168.10.3", nonce="49b6c5f00000000c81c829543f443c845ee6c9514213aa27", response="269dbb2125ca6984dbb8d427a0f6a053" Call-ID: 52934087b53c9f07(a)192.168.10.30 CSeq: 20724 INVITE User-Agent: Grandstream GXP2020 1.1.6.44 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=201 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5050 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.001127 192.168.10.3:5060 -> 192.168.10.30:5064 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKd5d3f9baf951bf4e From: "Opensips-14x" <sip:201@192.168.10.3>;tag=f89d240daba57386 To: <sip:2685249@192.168.10.3> Call-ID: 52934087b53c9f07(a)192.168.10.30 CSeq: 20724 INVITE Server: OpenSIPS (1.4.4-notls (i386/linux)) Content-Length: 0 alguna idea ... -- rickygm http://gnuforever.homelinux.com

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[Kamailio-Users-ES] Record route en publish
by Jon Bonilla 09 Mar '09

09 Mar '09

Saludos Me he encontrado con un tratamiento que hace record-route a los PUBLISH en un fichero de configuración de Kamailio. Me he dado cuenta de que en el fichero por defecto también aparece así: if (!is_method("REGISTER|MESSAGE")) record_route(); if( is_method("PUBLISH|SUBSCRIBE")){ sl_send_reply("503", "Service Unavailable"); exit; } ¿Por qué se le pone el record-route al publish? No es una initial request que establezca un diálogo y según el rfc 3903: A PUBLISH request does not establish a dialog. A UAC MAY include a Route header field in a PUBLISH request based on a pre-existing route set as described in Section 8.1 of RFC 3261 [4]. The Record-Route header field has no meaning in PUBLISH requests or responses, and MUST be ignored if present. Lo que me mosquea es que los de Kamailio, que saben de esto mucho más que yo, hayan puesto el record-route y se me esté escapando algo.

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sr-users-es March 2009