Hi all!

I really don't know why "Mitel" rejects my calls. I'm using Kamailio to forward calls to Mitel.

A little more graphic:

Please see the picture:

http://s3.subirimagenes.com:81/otros/5226539form.jpg

SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone

Mitel rejects my calls with "404 Not Found". Ok, you may think: "the extension that you are calling doesn't exists".. please dont think that.

(One more thing: If I try to make the same scene using Asterisk instead Kamailio everything works fine.)

So, I made a sip capture to see what happens:

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/2.0.
Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a.
From: "Sip Phone" <sip:100@192.168.10.150>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@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>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@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/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>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@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>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@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>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@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.

This is my Kamailio code from reenvites..

route[4] {
t_relay("udp:192.168.10.160:5060");
t_on_reply("1");
exit;
}

If you pay attention to INVITES (Kamailio SIP messages) you will see:

From: "Sip Phone" <sip:100@192.168.10.150>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.150>.

I think that should be:

From: "Sip Phone" <sip:100@192.168.10.150>;tag=d396005aaf3ab9a2o0.
To: "Mitel Phone" <sip:200@192.168.10.160>.

It could be the reason for Mitel rejects? Can I fix it? I can use TEXTOPS but I cant understand why Mitel rejects the Kamailio INVITES.

I will thanks any help!