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