sr-users-es September 2010

sr-users-es@lists.kamailio.org

3 participants
2 discussions

Kamailio --> Mitel (Not Found) ¿Problemas con el INVITE?

by Tincho ylm

Hola a todos! Recurro a ustedes para ver si me pueden ayudar con este problema al cual no le encuentro solucion: Primero les muestro el esquema que poseo: SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone Lo que quiero hacer es llamar desde la extension Linksys a traves de Kamailio a una extension de la central propietaria Mitel. Actualmente Mitel rechaza mis llamadas con un bonito 404 Not Found! lo cual es imposible porque la extension Mitel existe y funciona bien. Ademas, solo para probar, intente el mismo escenario, pero en vez de utilizar Kamailio, puse un asterisk y funciona barbaro. Entonces... en algo le estoy errando en Kamailio. Hice unas capturas para que puedan ver si hay algun inconveniente con los INVITES (por logica no deberian existir, pero por las dudas los pongo) Primero una aclaracion sobre las IPs: Sip Phone -> 100 192.168.10.140 -> Sip Phone 192.168.10.150 -> Kamailio 192.168.10.160 -> Mitel Mitel Phone -> 200 Kamailio U 192.168.10.140:5060 -> 192.168.10.150:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 70. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.150:5060 -> 192.168.10.140:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.10.140:5060 ;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Server: Kamailio (1.5.4-notls (i386/linux)). Content-Length: 0. U 192.168.10.150:5060 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0. Via: SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 69. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Content-Length: 0. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Contact: <sip:192.168.10.160>. Content-Length: 0. Lo mismo pero con Asterisk (que si funciona) * * U 192.168.10.140:5060 -> 192.168.10.150:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d5c5100f. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=59178c25144180dfo0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: 5f5d6a1d-6b287b47 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="100",realm="asterisk",nonce="3ed77171",uri="sip:200@192.168.10.150<sip%3A200(a)192.168.10.150> ",algorithm=... Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 397. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.150:5060 -> 192.168.10.140:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.140:5060 ;branch=z9hG4bK-d5c5100f;received=192.168.10.140. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=59178c25144180dfo0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: 5f5d6a1d-6b287b47 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 102 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Contact: <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Content-Length: 0. U 192.168.10.150:5060 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=as749996b5. To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>. Contact: <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150>>. Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the [!at] with a @). CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 22 Sep 2010 12:36:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 235. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=as749996b5. To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> >;tag=0_3844423584-65506194. Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the [!at] with a @). CSeq: 102 INVITE. Content-Length: 0. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=as749996b5. To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> >;tag=0_3844423584-65506194. Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the [!at] with a @). CSeq: 102 INVITE. Contact: <sip:200@192.168.10.160:5060;transport=udp>. Content-Length: 0. * *La configuracion de Kamailio es default. Basicamente consulto por el numero ingresado, si es 200 lo envio para la Mitel: route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); if (is_method("INVITE") && $rU =~ "200"){ force_rtp_proxy(); t_on_failure("1"); route(5); }; }; } route[5] { t_relay("udp:192.168.2.1:5060"); t_on_reply("1"); exit; } Tienen idea si me esta faltando algo??? Les agradecere cualquier ayuda.

13 years, 6 months

Kamailio --> Mitel (Not Found) ¿Problemas con el INVITE?

by Tincho ylm

> > > Por favor, no le presten atencion al error: t_relay("udp:192.168.2.1:5060"); En realidad es: t_relay("udp:192.168.10.160:5060 <http://192.168.2.1:5060/>"); Aviso por si alguien responde diciendo que puede ser esto, NO LO ES, simplemente me equivoque al ponerlo en el mail. Gracias. > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 22 Sep 2010 11:53:49 -0300 > From: Tincho ylm <sadzas(a)gmail.com> > Subject: [SR-Users-ES] Kamailio --> Mitel (Not Found) ¿Problemas con > el INVITE? > To: sr-users-es(a)lists.sip-router.org > Message-ID: > <AANLkTinYcfBvQ4XtVt_mSvKtNib8M043u-8VnotucDzc(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hola a todos! > > Recurro a ustedes para ver si me pueden ayudar con este problema al cual no > le encuentro solucion: > > Primero les muestro el esquema que poseo: > > SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone > > Lo que quiero hacer es llamar desde la extension Linksys a traves de > Kamailio a una extension de la central propietaria Mitel. Actualmente Mitel > rechaza mis llamadas con un bonito 404 Not Found! lo cual es imposible > porque la extension Mitel existe y funciona bien. > Ademas, solo para probar, intente el mismo escenario, pero en vez de > utilizar Kamailio, puse un asterisk y funciona barbaro. Entonces... en algo > le estoy errando en Kamailio. > > Hice unas capturas para que puedan ver si hay algun inconveniente con los > INVITES (por logica no deberian existir, pero por las dudas los pongo) > > Primero una aclaracion sobre las IPs: > > Sip Phone -> 100 > 192.168.10.140 -> Sip Phone > 192.168.10.150 -> Kamailio > 192.168.10.160 -> Mitel > Mitel Phone -> 200 > > Kamailio > U 192.168.10.140:5060 -> 192.168.10.150:5060 > INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>> SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 101 INVITE. > Max-Forwards: 70. > Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. > Expires: 240. > User-Agent: Linksys/SPA941-5.1.8. > Content-Length: 395. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: replaces. > Content-Type: application/sdp. > > U 192.168.10.150:5060 -> 192.168.10.140:5060 > SIP/2.0 100 Giving a try. > Via: SIP/2.0/UDP 192.168.10.140:5060 > ;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 101 INVITE. > Server: Kamailio (1.5.4-notls (i386/linux)). > Content-Length: 0. > > U 192.168.10.150:5060 -> 192.168.10.160:5060 > INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>> SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0. > Via: SIP/2.0/UDP 192.168.10.140:5060 > ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 101 INVITE. > Max-Forwards: 69. > Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. > Expires: 240. > User-Agent: Linksys/SPA941-5.1.8. > Content-Length: 395. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: replaces. > Content-Type: application/sdp. > > U 192.168.10.160:5060 -> 192.168.10.150:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP > 192.168.10.140:5060 > ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>> > >;tag=0_4044193584-65506210. > Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 101 INVITE. > Content-Length: 0. > > U 192.168.10.160:5060 -> 192.168.10.150:5060 > SIP/2.0 404 Not Found. > Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP > 192.168.10.140:5060 > ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=d396005aaf3ab9a2o0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>> > >;tag=0_4044193584-65506210. > Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 101 INVITE. > Contact: <sip:192.168.10.160>. > Content-Length: 0. > > Lo mismo pero con Asterisk (que si funciona) > * > * > U 192.168.10.140:5060 -> 192.168.10.150:5060 > INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>> SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d5c5100f. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=59178c25144180dfo0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Call-ID: 5f5d6a1d-6b287b47 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 102 INVITE. > Max-Forwards: 70. > Proxy-Authorization: Digest > username="100",realm="asterisk",nonce="3ed77171",uri=" > sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150><sip%3A200(a)192.168.10.150<sip%253A200(a)192.168.10.150> > > > ",algorithm=... > Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. > Expires: 240. > User-Agent: Linksys/SPA941-5.1.8. > Content-Length: 397. > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > Supported: replaces. > Content-Type: application/sdp. > > U 192.168.10.150:5060 -> 192.168.10.140:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.10.140:5060 > ;branch=z9hG4bK-d5c5100f;received=192.168.10.140. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=59178c25144180dfo0. > To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Call-ID: 5f5d6a1d-6b287b47 [!at] 192.168.10.140 (replace the [!at] with a > @). > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. > Supported: replaces. > Contact: <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> < > sip%3A200(a)192.168.10.150 <sip%253A200(a)192.168.10.150>>>. > Content-Length: 0. > > U 192.168.10.150:5060 -> 192.168.10.160:5060 > INVITE sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> < > sip%3A200(a)192.168.10.160 <sip%253A200(a)192.168.10.160>> SIP/2.0. > Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=as749996b5. > To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> < > sip%3A200(a)192.168.10.160 <sip%253A200(a)192.168.10.160>>>. > Contact: <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>>>. > Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the > [!at] with a @). > CSeq: 102 INVITE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Date: Wed, 22 Sep 2010 12:36:12 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 235. > > U 192.168.10.160:5060 -> 192.168.10.150:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=as749996b5. > To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> < > sip%3A200(a)192.168.10.160 <sip%253A200(a)192.168.10.160>> > >;tag=0_3844423584-65506194. > Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the > [!at] with a @). > CSeq: 102 INVITE. > Content-Length: 0. > > U 192.168.10.160:5060 -> 192.168.10.150:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.10.150:5060;branch=z9hG4bK5c9bfd49;rport. > From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> < > sip%3A100(a)192.168.10.150 <sip%253A100(a)192.168.10.150>> > >;tag=as749996b5. > To: <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160> < > sip%3A200(a)192.168.10.160 <sip%253A200(a)192.168.10.160>> > >;tag=0_3844423584-65506194. > Call-ID: 59dfcf505cb2671977c8c9175f50c96e [!at] 192.168.10.150 (replace the > [!at] with a @). > CSeq: 102 INVITE. > Contact: <sip:200@192.168.10.160:5060;transport=udp>. > Content-Length: 0. > * > *La configuracion de Kamailio es default. Basicamente consulto por el > numero > ingresado, si es 200 lo envio para la Mitel: > > route[2]{ > force_rport(); > if (nat_uac_test("19")) { > if (method=="REGISTER") { > fix_nated_register(); > } else { > fix_nated_contact(); > }; > setflag(5); > > if (is_method("INVITE") && $rU =~ "200"){ > force_rtp_proxy(); > t_on_failure("1"); > route(5); > }; > > }; > } > > route[5] { > t_relay("udp:192.168.2.1:5060"); > t_on_reply("1"); > exit; > } > > Tienen idea si me esta faltando algo??? > > Les agradecere cualquier ayuda. > ------------ próxima parte ------------ >

13 years, 7 months

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sr-users-es September 2010