sr-users-es

sr-users-es@lists.kamailio.org

487 discussions

Re: [Kamailio-Users-ES] Conexion de 2 Kamailio con TLS

by ingdavidcespedes＠cable.net.co

----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Viernes, Octubre 3, 2008 11:00 am Asunto: Re: [Kamailio-Users-ES] Conexion de 2 Kamailio con TLS > El Viernes, 3 de Octubre de 2008, ingdavidcespedes(a)cable.net.co > escribió:> > Nota: Algunos proveedores SIP te permiten indicar una > IP origen y todo > > > lo que venga de ahí se te factura. Todo esto para que no te > tengas que > > > autenticar. > > > > En el mio me tengo que autenticar con nombre de usuario y > contraseña. Lo > > anterior sugiere que mejor utilice asterisk para conectarme al > proveedor? > Con mucha suerte es posible que te funcione con el módulo "uac" de > OpenSer, > pero partiendo de la base, como he explicado antes, que es una > chapuza anti > estándares SIP y que puede que te funcione -de "chiripa"- con algún > teléfono > concreto y con algún proveedor concreto. > > Sí, Asterisk es una opción, como lo sería cualquier otro B2BUA. Gracias Iñaki, basado en esto lo haré con asterisk, pues como ya les he comentado, estoy haciendo la tesis sobre las recomendaciones técnicas de SIP Connect, y debo tratar de cumplir con todos los RFC's que allí enuncian sobre todo el de SIP 3261. Entonces no me sirve que me salga de chiripa, creo que con asterisk sera más claro el intercambio de mensajes SIP entre las diferentes entidades. > > -- > Iñaki Baz Castillo > > _______________________________________________ > Users-es mailing list > Users-es(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es >

16 years, 2 months

Re: [Kamailio-Users-ES] Conexion de 2 Kamailio con TLS

by ingdavidcespedes＠cable.net.co

----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Viernes, Octubre 3, 2008 3:13 am Asunto: Re: [Kamailio-Users-ES] Conexion de 2 Kamailio con TLS > El 3/10/08, ingdavidcespedes(a)cable.net.co > <ingdavidcespedes(a)cable.net.co> escribió: > > > Alguien a interconectado alguna vez dos Kamailio con TLS entre > ellos (Solo entre los Kamailio, no los teléfonos)?? alguna pista . > > Poder se puede, creo recordar que viene un howto completo en la > documentación. Lo buscare y lo leeré, solo quería saber si alguien ya lo había hecho para que me contara como le va. > > > Ahhh y otra pregunta, se puede conectar Kamailio directamente a > un Proveedor SIP, o > > es necesario hacerlo a través de asterisk, ya había preguntado > esto antes pero para > > tener salida directa desde Kamailio hacia la PSTN, pero me habían > contestado que > > necesitaba un GW que podía ser asterisk, pero no se si en el caso > de que el TRUNK > > sea SIP habrá también que hacer lo mismo, pienso que de pronto se > puede con el > > modulo de UAC pero todavía no lo he leído, cualquier guía sera > bienvenida. > El problema es que si se trata de un proveedor que pide autenticación > entonces OpenSer no puede añadir él la autenticación. Bueno, en > realidad sí puede, para eso es el módulo "uac" y uno de sus métodos, > pero es una chapuza ya que no se incrementa el CSeq y es muy posible > que el proveedor no lo acepte. OpenSer no puede aumentar el CSeq > porque los UAC's no lo sabrían y no aceptarían las respuestas del > proveedor con un CSeq incrementado. > > Nota: Algunos proveedores SIP te permiten indicar una IP origen y todo > lo que venga de ahí se te factura. Todo esto para que no te tengas que > autenticar. > En el mio me tengo que autenticar con nombre de usuario y contraseña. Lo anterior sugiere que mejor utilice asterisk para conectarme al proveedor? > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > _______________________________________________ > Users-es mailing list > Users-es(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es >

16 years, 2 months

[Kamailio-Users-ES] Fallo al transferir llamadas externas

by Jose Fernandez

Hola. Tengo una instalación con FXO Grandstream y Openser con varios teléfonos IP registrados, y el problema que tengo es que las llamadas que entran a través del FXO, al transferirlas entre teléfonos IP se cortan. No se pueden transferir llamadas externas entre teléfonos IP. Sin embargo con las llamadas internas entre los propios teléfonos IP no hay ningún problema para transferirlas. Este problema ha surgido a raíz de cambiar el modo de funcionamiento del FXO para que las llamadas entrantes suenen directamente en los teléfonos IP. Con la configuración anterior según la cual cuando entraba la llamada en el FXO era necesario una segunda marcación para seleccionar la extensión IP, la transferencia de llamadas funcionaba correctamente. ¿Me puede por favor echar alguien una mano a identificar la causa del corte en las transferencias? Muchas gracias de antemano. Adjunto traza ngrep, de una llamada externa entrante a la extensión 43 a través del canal FXO 9873, y el intento de transferencia de esta llamada desde la extensión 43 hacia la extensión 42. La traza a partir de haberse atendido ya la llamada externa en la extensión 42. # U 2008/09/19 14:30:19.983100 192.168.20.13:5060 -> 192.168.20.1:5060 INVITE sip:42@192.168.20.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060> Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 70 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 313 v=0 o=43 32364100 22806179 IN IP4 192.168.20.13 s=A conversation c=IN IP4 192.168.20.13 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:19.984128 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060> Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Server: OpenSer (1.1.1-notls (i386/linux)) Content-Length: 0 Warning: 392 192.168.20.1:5060 "Noisy feedback tells: pid=1627 req_src_ip=192.168.20.13 req_src_port=5060 in_uri=sip:42@192.168.20.1:5060 out_uri=sip:42@192.168.20.12:5060 via_cnt==1" # U 2008/09/19 14:30:19.984308 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060> Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 313 P-hint: usrloc applied v=0 o=43 32364100 22806179 IN IP4 192.168.20.13 s=A conversation c=IN IP4 192.168.20.13 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:20.202511 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060> Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 313 P-hint: usrloc applied v=0 o=43 32364100 22806179 IN IP4 192.168.20.13 s=A conversation c=IN IP4 192.168.20.13 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:20.205681 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060> Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Content-Length: 0 # U 2008/09/19 14:30:20.226804 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Content-Length: 0 # U 2008/09/19 14:30:20.227247 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Content-Length: 0 # U 2008/09/19 14:30:20.298327 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Content-Length: 0 # U 2008/09/19 14:30:20.299019 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Content-Length: 0 # U 2008/09/19 14:30:23.142771 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 13188124 91532501 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:23.143540 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK26887157682213617086;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 13188124 91532501 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:23.612392 192.168.20.13:5060 -> 192.168.20.1:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2420243251744332345 Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:23.613303 192.168.20.1:5060 -> 192.168.20.12:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.d21cd317.2 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2420243251744332345 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:27.150694 192.168.20.13:5060 -> 192.168.20.1:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 70 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 307 v=0 o=43 32364100 22806180 IN IP4 192.168.20.13 s=A conversation c=IN IP4 0.0.0.0 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly # U 2008/09/19 14:30:27.151252 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Server: OpenSer (1.1.1-notls (i386/linux)) Content-Length: 0 Warning: 392 192.168.20.1:5060 "Noisy feedback tells: pid=1628 req_src_ip=192.168.20.13 req_src_port=5060 in_uri=sip:42@192.168.20.12:5060 out_uri=sip:42@192.168.20.12:5060 via_cnt==1" # U 2008/09/19 14:30:27.151423 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 307 P-hint: rr-enforced v=0 o=43 32364100 22806180 IN IP4 192.168.20.13 s=A conversation c=IN IP4 0.0.0.0 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly # U 2008/09/19 14:30:27.209761 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 307 P-hint: rr-enforced v=0 o=43 32364100 22806180 IN IP4 192.168.20.13 s=A conversation c=IN IP4 0.0.0.0 t=0 0 m=audio 10320 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly # U 2008/09/19 14:30:27.406839 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 22322225 14523311 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly # U 2008/09/19 14:30:27.407193 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 22322225 14523311 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly # U 2008/09/19 14:30:27.491887 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 22322225 14523311 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly # U 2008/09/19 14:30:27.492279 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2273717731513410102;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 334 v=0 o=42 22322225 14523311 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10176 RTP/AVP 8 4 18 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly # U 2008/09/19 14:30:27.539823 192.168.20.13:5060 -> 192.168.20.1:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK392818939904813707 Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:27.540247 192.168.20.1:5060 -> 192.168.20.12:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.2 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK392818939904813707 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:27.709861 192.168.20.13:5060 -> 192.168.20.1:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK392818939904813707 Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:27.710267 192.168.20.1:5060 -> 192.168.20.12:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=1176531356> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK8676.852fdee7.2 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK392818939904813707 From: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 To: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 2 ACK Contact: <sip:43@192.168.20.13:5060> Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:27.762162 192.168.20.13:5060 -> 192.168.20.1:5060 REFER sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK183391788069093717;rport Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 2 REFER Contact: <sip:43@192.168.20.13:5060> refer-to: <sip:42@192.168.20.1:5060?Replaces=66593269826955-13455355112414%40192.168.20.13%3Bto-tag%3D3026415234%3Bfrom-tag%3D1176531356> referred-by: <sip:43@192.168.20.1> Max-Forwards: 70 User-Agent: Voip Phone 1.0 Event: refer Content-Length: 0 # U 2008/09/19 14:30:27.762693 192.168.20.1:5060 -> 192.168.10.2:5066 REFER sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK6263.9c369da.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK183391788069093717;rport=5060 From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 2 REFER Contact: <sip:43@192.168.20.13:5060> refer-to: <sip:42@192.168.20.1:5060?Replaces=66593269826955-13455355112414%40192.168.20.13%3Bto-tag%3D3026415234%3Bfrom-tag%3D1176531356> referred-by: <sip:43@192.168.20.1> Max-Forwards: 69 User-Agent: Voip Phone 1.0 Event: refer Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:27.859294 192.168.10.2:5066 -> 192.168.20.1:5060 SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK6263.9c369da.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK183391788069093717;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 2 REFER User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Contact: <sip:9873@192.168.10.2:5066;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U 2008/09/19 14:30:27.859610 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK183391788069093717;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 2 REFER User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Contact: <sip:9873@192.168.10.2:5066;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U 2008/09/19 14:30:27.880527 192.168.10.2:5066 -> 192.168.20.1:5060 NOTIFY sip:43@192.168.20.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK18161fe54437b433 Route: <sip:192.168.20.1;lr=on;ftag=3d05668484314852> From: "unknown"<sip:unknown@192.168.20.1>;tag=3d05668484314852 To: <sip:43@192.168.20.1>;tag=2478014423 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 10757 NOTIFY User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: refer Subscription-State: active Content-Type: message/sipfrag;version=2.0 Content-Length: 20 SIP/2.0 100 Trying # U 2008/09/19 14:30:27.881054 192.168.20.1:5060 -> 192.168.20.13:5060 NOTIFY sip:43@192.168.20.13:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=3d05668484314852> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKcd3.d84f00f3.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK18161fe54437b433 From: "unknown"<sip:unknown@192.168.20.1>;tag=3d05668484314852 To: <sip:43@192.168.20.1>;tag=2478014423 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 10757 NOTIFY User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: refer Subscription-State: active Content-Type: message/sipfrag;version=2.0 Content-Length: 20 P-hint: rr-enforced SIP/2.0 100 Trying # U 2008/09/19 14:30:27.912377 192.168.10.2:5066 -> 192.168.20.1:5060 INVITE sip:42@192.168.20.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060> Contact: <sip:9873@192.168.10.2:5066;user=phone> Replaces: 66593269826955-13455355112414(a)192.168.20.13;to-tag=3026415234;from-tag=1176531356 Supported: replaces, timer, path Referred-By: <sip:43@192.168.20.1> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 235 v=0 o=9873 8007 8000 IN IP4 192.168.10.2 s=SIP Call c=IN IP4 192.168.10.2 t=0 0 m=audio 5032 RTP/AVP 8 0 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U 2008/09/19 14:30:27.912889 192.168.20.1:5060 -> 192.168.10.2:5066 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Server: OpenSer (1.1.1-notls (i386/linux)) Content-Length: 0 Warning: 392 192.168.20.1:5060 "Noisy feedback tells: pid=1626 req_src_ip=192.168.10.2 req_src_port=5066 in_uri=sip:42@192.168.20.1:5060 out_uri=sip:42@192.168.20.12:5060 via_cnt==1" # U 2008/09/19 14:30:27.913010 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060> Contact: <sip:9873@192.168.10.2:5066;user=phone> Replaces: 66593269826955-13455355112414(a)192.168.20.13;to-tag=3026415234;from-tag=1176531356 Supported: replaces, timer, path Referred-By: <sip:43@192.168.20.1> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 235 P-hint: usrloc applied v=0 o=9873 8007 8000 IN IP4 192.168.10.2 s=SIP Call c=IN IP4 192.168.10.2 t=0 0 m=audio 5032 RTP/AVP 8 0 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U 2008/09/19 14:30:28.140252 192.168.20.13:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKcd3.d84f00f3.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK18161fe54437b433 Record-Route: <sip:192.168.20.1;lr=on;ftag=3d05668484314852> From: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 To: <sip:43@192.168.20.1>;tag=2478014423 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 10757 NOTIFY Contact: <sip:43@192.168.20.13:5060> Content-Length: 0 # U 2008/09/19 14:30:28.140839 192.168.20.1:5060 -> 192.168.10.2:5066 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK18161fe54437b433 Record-Route: <sip:192.168.20.1;lr=on;ftag=3d05668484314852> From: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 To: <sip:43@192.168.20.1>;tag=2478014423 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 10757 NOTIFY Contact: <sip:43@192.168.20.13:5060> Content-Length: 0 # U 2008/09/19 14:30:28.193697 192.168.20.13:5060 -> 192.168.20.1:5060 BYE sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:28.194430 192.168.20.1:5060 -> 192.168.10.2:5066 BYE sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK7263.af092c33.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport=5060 From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:28.210618 192.168.20.1:5060 -> 192.168.20.12:5060 INVITE sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060> Contact: <sip:9873@192.168.10.2:5066;user=phone> Replaces: 66593269826955-13455355112414(a)192.168.20.13;to-tag=3026415234;from-tag=1176531356 Supported: replaces, timer, path Referred-By: <sip:43@192.168.20.1> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 235 P-hint: usrloc applied v=0 o=9873 8007 8000 IN IP4 192.168.10.2 s=SIP Call c=IN IP4 192.168.10.2 t=0 0 m=audio 5032 RTP/AVP 8 0 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U 2008/09/19 14:30:28.211176 192.168.20.1:5060 -> 192.168.10.2:5066 BYE sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK7263.af092c33.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport=5060 From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:28.286970 192.168.10.2:5066 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK7263.af092c33.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Contact: <sip:9873@192.168.10.2:5066;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U 2008/09/19 14:30:28.287776 192.168.20.1:5060 -> 192.168.20.13:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:6) 1.0.1.2 Contact: <sip:9873@192.168.10.2:5066;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U 2008/09/19 14:30:28.326083 192.168.10.2:5066 -> 192.168.20.1:5060 SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK7263.af092c33.0 Via: SIP/2.0/UDP 192.168.20.13:5060;branch=z9hG4bK2144212896305094925;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=2478014423> From: <sip:43@192.168.20.1>;tag=2478014423 To: unknown <sip:unknown@192.168.20.1>;tag=3d05668484314852 Call-ID: 4263e231cf65ba11(a)192.168.10.2 CSeq: 3 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:0) 1.0.1.2 Content-Length: 0 # U 2008/09/19 14:30:28.364322 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Content-Length: 0 # U 2008/09/19 14:30:28.393542 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=42 42889345 25269231 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10178 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:28.394149 192.168.20.1:5060 -> 192.168.10.2:5066 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=42 42889345 25269231 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10178 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:28.468273 192.168.20.12:5060 -> 192.168.20.1:5060 BYE sip:43@192.168.20.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK17746209931754612026;rport Route: <sip:192.168.20.1;lr=on;ftag=1176531356> From: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 To: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:28.469112 192.168.20.1:5060 -> 192.168.20.13:5060 BYE sip:43@192.168.20.13:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=3026415234> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.e21cd317.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK17746209931754612026;rport=5060 From: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 To: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 BYE Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:28.500801 192.168.10.2:5066 -> 192.168.20.1:5060 CANCEL sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 CANCEL User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U 2008/09/19 14:30:28.501604 192.168.20.1:5060 -> 192.168.10.2:5066 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 CANCEL Server: OpenSer (1.1.1-notls (i386/linux)) Content-Length: 0 Warning: 392 192.168.20.1:5060 "Noisy feedback tells: pid=1627 req_src_ip=192.168.10.2 req_src_port=5066 in_uri=sip:42@192.168.20.12:5060 out_uri=sip:42@192.168.20.12:5060 via_cnt==1" # U 2008/09/19 14:30:28.506648 192.168.20.12:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=42 42889345 25269231 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10178 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:28.507237 192.168.20.1:5060 -> 192.168.10.2:5066 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK2b17d6e34a56abe6 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 INVITE Contact: <sip:42@192.168.20.12:5060> Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=42 42889345 25269231 IN IP4 192.168.20.12 s=A conversation c=IN IP4 192.168.20.12 t=0 0 m=audio 10178 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2008/09/19 14:30:28.521309 192.168.10.2:5066 -> 192.168.20.1:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK8497d3318fc0bc45 Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U 2008/09/19 14:30:28.522076 192.168.20.1:5060 -> 192.168.20.12:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.2 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bK8497d3318fc0bc45 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:28.638974 192.168.20.13:5060 -> 192.168.20.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKb676.e21cd317.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK17746209931754612026;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=3026415234> From: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 To: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 BYE Contact: <sip:43@192.168.20.13:5060> Content-Length: 0 # U 2008/09/19 14:30:28.639596 192.168.20.1:5060 -> 192.168.20.12:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK17746209931754612026;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=3026415234> From: 42 <sip:42@192.168.20.1:5060>;tag=3026415234 To: usuario43 <sip:43@192.168.20.1:5060>;tag=1176531356 Call-ID: 66593269826955-13455355112414(a)192.168.20.13 CSeq: 1 BYE Contact: <sip:43@192.168.20.13:5060> Content-Length: 0 # U 2008/09/19 14:30:28.664921 192.168.10.2:5066 -> 192.168.20.1:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bKf6d0f6c1ee93df11 Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U 2008/09/19 14:30:28.665670 192.168.20.1:5060 -> 192.168.20.12:5060 ACK sip:42@192.168.20.12:5060 SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bKed4e.8b482304.2 Via: SIP/2.0/UDP 192.168.10.2:5066;branch=z9hG4bKf6d0f6c1ee93df11 From: "unknown"<sip:unknown@192.168.20.1>;tag=10263706333383c3 To: <sip:42@192.168.20.1:5060>;tag=983812395 Contact: <sip:9873@192.168.10.2:5066;user=phone> Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 21300 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:7) 1.0.1.2 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:41.078326 192.168.20.12:5060 -> 192.168.20.1:5060 BYE sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK1309813609243777314;rport Route: <sip:192.168.20.1;lr=on;ftag=10263706333383c3> From: <sip:42@192.168.20.1:5060>;tag=983812395 To: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 1 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 # U 2008/09/19 14:30:41.078856 192.168.20.1:5060 -> 192.168.10.2:5066 BYE sip:9873@192.168.10.2:5066;user=phone SIP/2.0 Record-Route: <sip:192.168.20.1;lr=on;ftag=983812395> Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK48e9.dc5f3ea5.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK1309813609243777314;rport=5060 From: <sip:42@192.168.20.1:5060>;tag=983812395 To: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 1 BYE Max-Forwards: 69 User-Agent: Voip Phone 1.0 Content-Length: 0 P-hint: rr-enforced # U 2008/09/19 14:30:41.183529 192.168.10.2:5066 -> 192.168.20.1:5060 SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.20.1;branch=z9hG4bK48e9.dc5f3ea5.0 Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK1309813609243777314;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=983812395> From: <sip:42@192.168.20.1:5060>;tag=983812395 To: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 1 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:0) 1.0.1.2 Content-Length: 0 # U 2008/09/19 14:30:41.183840 192.168.20.1:5060 -> 192.168.20.12:5060 SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.20.12:5060;branch=z9hG4bK1309813609243777314;rport=5060 Record-Route: <sip:192.168.20.1;lr=on;ftag=983812395> From: <sip:42@192.168.20.1:5060>;tag=983812395 To: unknown <sip:unknown@192.168.20.1>;tag=10263706333383c3 Call-ID: e400cdc544f16303(a)192.168.10.2 CSeq: 1 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:0) 1.0.1.2 Content-Length: 0

16 years, 2 months

[Kamailio-Users-ES] Redireccionar mensajes MESSAGE desde el openser a un servidor de aplicaciones

by patricia avila sendino

He decidido abrir un nuevo tema sobre esta consulta porque parece que no estaba llendo muy bien el otro, creo q no mandaba los mensajes a todos los usuarios o algo así. He conseguido concretar mas con mi jefe que es lo que tengo que hacer exactamente. Esto es hacer q los message q me llegen al openser me sean redireccionados a un servidor de aplicaciones. Ahora mismo tengo registrado el usuario 200 del openser en el eyebeam y desde el eyebeam es desde donde mando los mensajes. El problema es que por el momento no tengo mas usuarios registrados y no se como hacer para poder enviar mensajes a otro usuario sin que este registrado, o como puedo registrarlo desde el mismo pc en el que tengo el eyebeam instalado, ya que no peudo instalar nada en ningún otro pc. Alguien puede darme alguna idea?? Otra cosa, para la redireccion de los mensajes lo que he hecho es poner dentro del route lo siguiente, ya que no encuentro ningun apartado en el que se procesen los message:if (is_method("MESSAGE")) { rewritehostport('10.95.44.224:5090'); } Creeis que está bien? Un saludo y muchas gracias por la ayuda. _________________________________________________________________ ¿Sigue el calor? Consulta MSN El tiempo http://eltiempo.es.msn.com/

16 years, 2 months

[Kamailio-Users-ES] Conexion de 2 Kamailio con TLS

by ingdavidcespedes＠cable.net.co

Hola Lista! Alguien a interconectado alguna vez dos Kamailio con TLS entre ellos (Solo entre los Kamailio, no los teléfonos)?? alguna pista . Ahhh y otra pregunta, se puede conectar Kamailio directamente a un Proveedor SIP, o es necesario hacerlo a través de asterisk, ya había preguntado esto antes pero para tener salida directa desde Kamailio hacia la PSTN, pero me habían contestado que necesitaba un GW que podía ser asterisk, pero no se si en el caso de que el TRUNK sea SIP habrá también que hacer lo mismo, pienso que de pronto se puede con el modulo de UAC pero todavía no lo he leído, cualquier guía sera bienvenida. Saludos David Céspedes

16 years, 2 months

Re: [Kamailio-Users-ES] Problemas con SIPp

by ingdavidcespedes＠cable.net.co

Seria Bueno ver los mensajes SIP anteriores para ver en que momento se pierde la comunicación. Parecería que uno de los UA no se da cuenta que el otro se desconecto. Hola a todos. Estoy haciendo unas pruebas de stress en un asterisk instalado en un servidor potente con la intención de ver cuantas llamadas es capaz de aguantar (bueno eso ya os lo imaginabais :). El caso es que en el SIPp client cuando el flujo de Audio de la primera llamada enviada finaliza, el BYE se envía una y otra vez, no para de restransmitirse...No he mirado si a las siguientes llamadas le sucede lo mismo pero estoy seguro que sí porque al finalizar la pruebas tengo mucho mas reenvios de BYE que llamadas propiamente. Cuando empieza a fallar la carga tanto del servidor como del cliente no llega ni al 50% por lo que no es problema de recursos de CPU ni de memoría. Despues miro en lo logs de SIPp y me encuentro con estos errores, que se repiten con frecuencia: 2008-09-29 18:31:18:269 1222705878.269500: Continuing call on unexpected message for Call-Id '3-2320(a)192.168.1.140': while expecting '200' (index 10), received 'SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-2320-3-9;received=192.168.1.140 From: sipp <sip:sipp@192.168.1.140:5060>;tag=2320SIPpTag003 To: sut <sip:9876543210@192.168.1.246:5060>;tag=as407f5326 Call-ID: 3-2320(a)192.168.1.140 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 2008-09-29 18:31:18:642 1222705878.642500: Continuing call on unexpected message for Call-Id '17-2320(a)192.168.1.140': while expecting '200' (index 10), received 'SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-2320-17-9;received=192.168.1.140 From: sipp <sip:sipp@192.168.1.140:5060>;tag=2320SIPpTag0017 To: sut <sip:9876543210@192.168.1.246:5060>;tag=as77779841 Call-ID: 17-2320(a)192.168.1.140 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Hay a millones... Por lo que me he podido informar es un problema de que el destinatario de la llamada ha colgado y un NOTIFY no ha informado a al origen, por tanto el origen al enviar el BYE se encuentra con el problema. En principio creo que este no es el problema, porque en el escenario que uso en el SIPpSERVER no cuelga, es siempre el SIPpCLIENTE el que lo hace. También he visto que puede haber un problema con los tags del From, To, pero no lo he llegado a entender del todo bien. He visto que con este tema al parecer hay un patch para Asterisk (que por cierto la versión que estoy utilizando es la 1.4.21.2). La pena es que no encuentro ahora la Web con el bug. Cuando la encuentre os la hago saber. Espero que podais ayudarme porque tal vez conozcais este error de antemano. Muchas gracias :) _________________________________________________________________ Llega la nueva temporada. Consulta las nuevas tendencias en MSN Estilo http://estilo.es.msn.com/moda/ _______________________________________________ Users-es mailing list Users-es(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es

16 years, 2 months

[Kamailio-Users-ES] Problemas con SIPp

by ZiLi0n

Hola a todos. Estoy haciendo unas pruebas de stress en un asterisk instalado en un servidor potente con la intención de ver cuantas llamadas es capaz de aguantar (bueno eso ya os lo imaginabais :). El caso es que en el SIPp client cuando el flujo de Audio de la primera llamada enviada finaliza, el BYE se envía una y otra vez, no para de restransmitirse...No he mirado si a las siguientes llamadas le sucede lo mismo pero estoy seguro que sí porque al finalizar la pruebas tengo mucho mas reenvios de BYE que llamadas propiamente. Cuando empieza a fallar la carga tanto del servidor como del cliente no llega ni al 50% por lo que no es problema de recursos de CPU ni de memoría. Despues miro en lo logs de SIPp y me encuentro con estos errores, que se repiten con frecuencia: 2008-09-29 18:31:18:269 1222705878.269500: Continuing call on unexpected message for Call-Id '3-2320(a)192.168.1.140': while expecting '200' (index 10), received 'SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-2320-3-9;received=192.168.1.140 From: sipp <sip:sipp@192.168.1.140:5060>;tag=2320SIPpTag003 To: sut <sip:9876543210@192.168.1.246:5060>;tag=as407f5326 Call-ID: 3-2320(a)192.168.1.140 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 2008-09-29 18:31:18:642 1222705878.642500: Continuing call on unexpected message for Call-Id '17-2320(a)192.168.1.140': while expecting '200' (index 10), received 'SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-2320-17-9;received=192.168.1.140 From: sipp <sip:sipp@192.168.1.140:5060>;tag=2320SIPpTag0017 To: sut <sip:9876543210@192.168.1.246:5060>;tag=as77779841 Call-ID: 17-2320(a)192.168.1.140 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Hay a millones... Por lo que me he podido informar es un problema de que el destinatario de la llamada ha colgado y un NOTIFY no ha informado a al origen, por tanto el origen al enviar el BYE se encuentra con el problema. En principio creo que este no es el problema, porque en el escenario que uso en el SIPpSERVER no cuelga, es siempre el SIPpCLIENTE el que lo hace. También he visto que puede haber un problema con los tags del From, To, pero no lo he llegado a entender del todo bien. He visto que con este tema al parecer hay un patch para Asterisk (que por cierto la versión que estoy utilizando es la 1.4.21.2). La pena es que no encuentro ahora la Web con el bug. Cuando la encuentre os la hago saber. Espero que podais ayudarme porque tal vez conozcais este error de antemano. Muchas gracias :) _________________________________________________________________ Llega la nueva temporada. Consulta las nuevas tendencias en MSN Estilo http://estilo.es.msn.com/moda/

16 years, 2 months

Re: [Kamailio-Users-ES] FW: Problema al conectar openser con servidor de aplicaciones

by ingdavidcespedes＠cable.net.co

Pues yo estoy empesando con Kamailio, pero lo que yo haria seria crear otra extension, por ejemplo la 201 en otro pc y llamar desde la 200 a la nueve extension 201. Y en el kamailio pongo una ruta que si la 201 no contesta o esta ocupada, me reenvie la llamada al servidor de aplicaciones, y ahi haces lo que quieras, enviarlo al buzon, o mandarlo a un ivr que tengas, o no se... Pero mientras te sigas llamando a ti misma, va a seguir pasando lo que hasta ahora. David Cespedes ----- Mensaje original ----- De: patricia avila sendino <patavil3(a)hotmail.com> Fecha: Martes, Septiembre 30, 2008 5:30 am Asunto: RE: [Kamailio-Users-ES] FW: Problema al conectar openser con servidor de aplicaciones > > Hola, > > pues lo que hice fue agregar un usuario al openser, q es el 200, > luego registro ese usuario en el eyebeam y realizo una llamada al > 200 para llamar al openser. Asi que si estoy llamando al mismo > usuario! No lo habia visto desde ese punto de vista. Tengo > instalado en una maquina el openser y en otra el eyebeam y el > servidor de aplicaciones. El objetivo es redireccionar las llamadas > q hago desde el eyebeam al openser al servidor de aplicaciones. > > como tengo que hacerlo entonces? alguien puede darme alguna idea de > como hacerlo? > > Muchas gracias! > > > > Date: Mon, 29 Sep 2008 16:36:08 -0500From: > ingdavidcespedes(a)cable.net.coTo: users- > es(a)lists.kamailio.orgSubject: Re: [Kamailio-Users-ES] FW: Problema > al conectar openser con servidor de aplicaciones > > > > > > > From: patavil3(a)hotmail.comTo: users-es(a)lists.kamailio.orgDate: Mon, > 29 Sep 2008 08:46:45 +0000Subject: Re: [Kamailio-Users-ES] Problema > al conectar openser con servidor de aplicaciones > > Hola, he hecho el ngrep que me inidcas y lo que obtengo es lo > siguiente: Mientras hago el ngrep lo que hago es registrar el > eyebeam y realizar una llamada, como ya dije antes me redirecciona > la llamada a la línea dos del eyebeam asi q desde esa linea cojo la > llamada y cuelgo. El resultado del ngrep con la realización de > estas operaciones es:U +0.000421 10.95.44.29:5060 -> > 10.95.40.224:10284SIP/2.0 503 Service Unavailable To: > <sip:200@10.95.44.29>;tag=329cfeaa6ded039da25ff8cbb8668bd2.d2b6 From: patri<sip:200@10.95.44.29>;tag=fa104450 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-435439142-1--d87543-;rport=10284 Call-ID: b27ccb59ef3a5b01 CSeq: 1 PUBLISH Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 #U +0.001236 10.95.40.224:10284 -> 10.95.44.29:5060SUBSCRIBE sip:200@10.95.44.29 SIP/2.0 To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Contact: <sip:200@10.95.40.224:10284> Expires: 3600 Max-Forwards: 70 Accept: application/watcherinfo+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 3004t stamp 16741 Event: presence.winfo Content-Length: 0 #U +0.000234 10.95.44.29:5060 -> 10.95.40.224:10284SUBSCRIBE sip:200@10.95.40.224:10284 SIP/2.0 Record-Route: <sip:10. 95.44.29;lr=on> To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bKc2a1.033c2294.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Contact: <sip:200@10.95.40.224:10284> Expires: 3600 Max-Forwards: 69 Accept: application/watcherinfo+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 3004t stamp 16741 Event: presence.winfo Content-Length: 0 #U +0.003088 10.95.40.224:10284 -> 10.95.44.29:5060SIP/2.0 489 Event Package Not Supported To: <sip:200@10.95.44.29>;tag=6a18d272 From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bKc2a1.033c2294.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Content-Length: 0 #U +0.000177 10.95.44.29 :5060 -> 10.95.40.224:10284SIP/2.0 489 Event Package Not Supported To: <sip:200@10.95.44.29>;tag=6a18d272 From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Content-Length: 0 #U +8.997574 10.95.40.224:10284 -> 10.95.44.29:5060 #U +8.999294 10.95.40.224:10284 -> 10.95.44.29:5060 #U +9.010038 10.95.40.224:10284 -> 10.95.44.29:5060 #U +9.000251 10.95.40.224:10284 -> 10.95.44.29:5060 #U +6.484777 10.95.40.224:10284 -> 10.95.44.29:5060INVITE sip:200@10.95.44.29 SIP/2.0 To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam relea se 3004t stamp 16741 Content-Length: 221 v=0 o=- 5035572 5035638 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10286 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv #U +0.000137 10.95.44.29:5060 -> 10.95.40.224:10284SIP/2.0 100 Giving a try To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 #U +0.000152 10.95.44.29:5060 -> 10.95.40.224:10284INVITE sip:200@10.95.40.224:10284 SIP/2.0 Record-Route: <sip:10.95.44.29;lr=on> To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 221 v=0 o=- 5035572 5035638 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10286 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv #U +0.080853 10.95.40.224:10284 -> 10.95.44.29:5060SIP/2.0 180 Ringing To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 #U +0.000397 10.95.44.29:5060 -> 10.95.40.224:10284SIP/2.0 180 Ringing To: <sip:2 00(a)10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 #U +2.440919 10.95.40.224:10284 -> 10.95.44.29:5060 #U +0.633981 10.95.40.224:10284 -> 10.95.44.29:5060SIP/2.0 200 OK To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 221 v=0 o=- 5035657 5035723 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 1 0.95.40.224 t=0 0 m=audio 10290 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv #U +0.000390 10.95.44.29:5060 -> 10.95.40.224:10284SIP/2.0 200 OK To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 221 v=0 o=- 5035657 5035723 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10290 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv #U +0.002764 10.95.40.224:10284 -> 10.95.44.29:5060ACK sip:200@10.95.40.224:10284 SIP/2.0 To: <sip:200@10.95.44.29>;tag=d659c300 From: patr i<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-372666725-1--d87543-;rport Call-ID: 43640a0eb13ab71f CSeq: 1 ACK Route: <sip:10.95.44.29;lr> Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 #U +0.000345 10.95.44.29:5060 -> 10.95.40.224:10284ACK sip:200@10.95.40.224:10284 SIP/2.0 To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.2 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-372666725-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 ACK Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 #U +3.347698 10.95.40.224:10284 -> 10.95.44.29:5060BYE sip:200@10.95.40.224:10284 SIP/2.0 To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.40 .224:10284;branch=z9hG4bK-d87543-941314466-1--d87543-;rport Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Route: <sip:10.95.44.29;lr> Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 #U +0.000158 10.95.44.29:5060 -> 10.95.40.224:10284BYE sip:200@10.95.40.224:10284 SIP/2.0 To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK6364.3160b9b5.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 #U +0.067856 10.95.40.224:10284 -> 10.95.44.29:5060SIP/2.0 200 OK To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK6364.3160b9b5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.22 4:10284;branch=z9hG4bK-d87543-941314466-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 #U +0.000613 10.95.44.29:5060 -> 10.95.40.224:10284SIP/2.0 200 OK To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 Alguien puede ayudarme a ver que es lo que está pasando? Muchas gracias de antemano. > > > > Juega a las preguntas de Live Quiz con tus contactos de Messenger > ¡Empieza ahora! > > > > ¡Trónchate de risa con los mejores capítulos de South Park en MSN > Vídeo! > _________________________________________________________________ > ¿Sigue el calor? Consulta MSN El tiempo > http://eltiempo.es.msn.com/

16 years, 2 months

Re: [Kamailio-Users-ES] FW: Problema al conectar openser con servidor de aplicaciones

by ingdavidcespedes＠cable.net.co

Hola Patricia Yo lo que veo es que desde el comienzo estas llamándote a ti misma, de la extensión 200 a la 200. Podrías aclarar desde y hacia que extensión estas llamando. Que dirección tiene tu servidor de aplicaciones? esta en la misma maquina que el Open SER? From: patavil3(a)hotmail.comTo: users-es(a)lists.kamailio.orgDate: Mon, 29 Sep 2008 08:46:45 +0000Subject: Re: [Kamailio-Users-ES] Problema al conectar openser con servidor de aplicaciones Hola, he hecho el ngrep que me inidcas y lo que obtengo es lo siguiente: Mientras hago el ngrep lo que hago es registrar el eyebeam y realizar una llamada, como ya dije antes me redirecciona la llamada a la línea dos del eyebeam asi q desde esa linea cojo la llamada y cuelgo. El resultado del ngrep con la realización de estas operaciones es: U +0.000421 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 503 Service Unavailable To: <sip:200@10.95.44.29>;tag=329cfeaa6ded039da25ff8cbb8668bd2.d2b6 From: patri<sip:200@10.95.44.29>;tag=fa104450 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-435439142-1--d87543-;rport=10284 Call-ID: b27ccb59ef3a5b01 CSeq: 1 PUBLISH Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.001236 10.95.40.224:10284 -> 10.95.44.29:5060 SUBSCRIBE sip:200@10.95.44.29 SIP/2.0 To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Contact: <sip:200@10.95.40.224:10284> Expires: 3600 Max-Forwards: 70 Accept: application/watcherinfo+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 3004t stamp 16741 Event: presence.winfo Content-Length: 0 # U +0.000234 10.95.44.29:5060 -> 10.95.40.224:10284 SUBSCRIBE sip:200@10.95.40.224:10284 SIP/2.0 Record-Route: <sip:10.95.44.29;lr=on> To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bKc2a1.033c2294.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Contact: <sip:200@10.95.40.224:10284> Expires: 3600 Max-Forwards: 69 Accept: application/watcherinfo+xml Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 3004t stamp 16741 Event: presence.winfo Content-Length: 0 # U +0.003088 10.95.40.224:10284 -> 10.95.44.29:5060 SIP/2.0 489 Event Package Not Supported To: <sip:200@10.95.44.29>;tag=6a18d272 From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bKc2a1.033c2294.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543-;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Content-Length: 0 # U +0.000177 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 489 Event Package Not Supported To: <sip:200@10.95.44.29>;tag=6a18d272 From: patri<sip:200@10.95.44.29>;tag=10230d60 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-789935123-1--d87543- ;rport=10284 Call-ID: 68145063306f8542 CSeq: 1 SUBSCRIBE Content-Length: 0 # U +8.997574 10.95.40.224:10284 -> 10.95.44.29:5060 # U +8.999294 10.95.40.224:10284 -> 10.95.44.29:5060 # U +9.010038 10.95.40.224:10284 -> 10.95.44.29:5060 # U +9.000251 10.95.40.224:10284 -> 10.95.44.29:5060 # U +6.484777 10.95.40.224:10284 -> 10.95.44.29:5060 INVITE sip:200@10.95.44.29 SIP/2.0 To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 221 v=0 o=- 5035572 5035638 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10286 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv # U +0.000137 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 100 Giving a try To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000152 10.95.44.29:5060 -> 10.95.40.224:10284 INVITE sip:200@10.95.40.224:10284 SIP/2.0 Record-Route: <sip:10.95.44.29;lr=on> To: <sip:200@10.95.44.29> From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 221 v=0 o=- 5035572 5035638 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10286 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv # U +0.080853 10.95.40.224:10284 -> 10.95.44.29:5060 SIP/2.0 180 Ringing To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 # U +0.000397 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 180 Ringing To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 # U +2.440919 10.95.40.224:10284 -> 10.95.44.29:5060 # U +0.633981 10.95.40.224:10284 -> 10.95.44.29:5060 SIP/2.0 200 OK To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543-;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 221 v=0 o=- 5035657 5035723 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10290 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv # U +0.000390 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 200 OK To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-793733969-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 INVITE Record-Route: <sip:10.95.44.29;lr=on> Contact: <sip:200@10.95.40.224:10284> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 221 v=0 o=- 5035657 5035723 IN IP4 10.95.40.224 s=eyeBeam c=IN IP4 10.95.40.224 t=0 0 m=audio 10290 RTP/AVP 100 6 0 8 3 18 5 101 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv # U +0.002764 10.95.40.224:10284 -> 10.95.44.29:5060 ACK sip:200@10.95.40.224:10284 SIP/2.0 To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-372666725-1--d87543- ;rport Call-ID: 43640a0eb13ab71f CSeq: 1 ACK Route: <sip:10.95.44.29;lr> Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 # U +0.000345 10.95.44.29:5060 -> 10.95.40.224:10284 ACK sip:200@10.95.40.224:10284 SIP/2.0 To: <sip:200@10.95.44.29>;tag=d659c300 From: patri<sip:200@10.95.44.29>;tag=4d47cc4b Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK9364.5b9e6ec5.2 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-372666725-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 1 ACK Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 # U +3.347698 10.95.40.224:10284 -> 10.95.44.29:5060 BYE sip:200@10.95.40.224:10284 SIP/2.0 To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543- ;rport Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Route: <sip:10.95.44.29;lr> Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 70 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 # U +0.000158 10.95.44.29:5060 -> 10.95.40.224:10284 BYE sip:200@10.95.40.224:10284 SIP/2.0 To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK6364.3160b9b5.0 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Max-Forwards: 69 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 # U +0.067856 10.95.40.224:10284 -> 10.95.44.29:5060 SIP/2.0 200 OK To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.44.29;branch=z9hG4bK6364.3160b9b5.0;received=10.95.44.29 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 # U +0.000613 10.95.44.29:5060 -> 10.95.40.224:10284 SIP/2.0 200 OK To: patri<sip:200@10.95.44.29>;tag=4d47cc4b From: <sip:200@10.95.44.29>;tag=d659c300 Via: SIP/2.0/UDP 10.95.40.224:10284;branch=z9hG4bK-d87543-941314466-1--d87543- ;rport=10284 Call-ID: 43640a0eb13ab71f CSeq: 2 BYE Contact: <sip:200@10.95.40.224:10284> Content-Length: 0 Alguien puede ayudarme a ver que es lo que está pasando? Muchas gracias de antemano. Juega a las preguntas de Live Quiz con tus contactos de Messenger ¡Empieza ahora! _________________________________________________________________ ¡Entra en el Club oficial de Messenger y te enterarás de todas las novedades! http://www.vivelive.com/ilovemessenger _______________________________________________ Users-es mailing list Users-es(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es

16 years, 2 months

Re: [Kamailio-Users-ES] Instalacion OpenSER en Fedora 9

by ingdavidcespedes＠cable.net.co

----- Mensaje original ----- De: ingdavidcespedes(a)cable.net.co Fecha: Jueves, Septiembre 25, 2008 3:43 pm Asunto: Re: [Kamailio-Users-ES] Instalacion OpenSER en Fedora 9 > Ese ya lo había usado > > [root@localhost sbin]# /usr/local/sbin/kamdbctl create > ERROR: database engine not specified, please setup one in the > config script ***creo que este error es precisamente por que no la > he creado, igual no la he habilitado en el kamailio.cfg por que > estoy siguiendo la guía paso a paso y eso viene después*** > > La habilito primero en el kamailio.cfg y luego la creo con > /usr/local/sbin/kamdbctl create??? voy a intentarlo aunque en la > guía está al revés. Me respondo a mi mismo: Efectivamente, primero modifique el kamctlrc, y para esto me toco antes modificar los permisos de ejecución y escritura de este archivo por que no me dejaba ni abrirlo, no se por que!! Una vez hecho esto, pude entrar y habilitar a kamailio para que usara la base de datos db_mysql, y ya con esto me dejo crearla, para luego modificar el kamailio.cfg y empezar a añadir usuarios!!!!! casi lloro de la felicidad cuando pude registrar los usuarios con éxito. Estuvo un pococ complicado por todos los problemas de mysql y el SElinux, pero al final pude dejándolo inhabilitado, aunque esto no me gusta mucho. Espero crear un documento que actualice la forma de instalar kamailio en fedora, para que otra gente no tenga los problemas que tuve yo. Ahora a integrarlo con Asterisk, vamos a ver como me va 1.¿Donde puedo encontrar la información mas reciente para hacer esto?. 2.Y a conectar los dos servidores Kamailio con TLS, ¿alguien ha hecho esto antes? 3.Todos los usuarios SIP se deben registrar en Kamailio, y Asterisk para este caso actuaria como un usuario mas registrado en kamailio, o se deben registrar tanto en kamailio como en asterisk?, esto no le tengo muy claro. > > > ----- Mensaje original ----- > De: Saúl Ibarra <saghul(a)gmail.com> > Fecha: Jueves, Septiembre 25, 2008 2:30 pm > Asunto: Re: [Kamailio-Users-ES] Instalacion OpenSER en Fedora 9 > > > > ¡¡¡¡ALGUIEN SABE COMO ES ESTE COMANDO PARA KAMAILIO!!!! > > > Se debería actualizar la documentación para kamailio, o si no > > creo que me va a tocar instalar un OpenSER viejo, pero como > > desinstalo el Kamailio?? > > > > > > > Usar kamdbctl :) > > > > > > -- > > Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de > > disketes."-------------------------------------------------------- > -- > > ------ > > http://www.saghul.net/ > > > > _______________________________________________ > > Users-es mailing list > > Users-es(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es > > > > > _______________________________________________ > Users-es mailing list > Users-es(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users-es >

16 years, 2 months

← Newer
1
...
16
17
18
19
20
21
22
...
49
Older →

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

sr-users-es