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:2...@192.168.10.150 <sip%3a...@192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@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:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.150 <sip%3a...@192.168.10.150>>. I think that should be: From: "Sip Phone" <sip:1...@192.168.10.150 <sip%3a...@192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:2...@192.168.10.160 <sip%3a...@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!
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