Thanks Daniel!

I'm using Debian, so this is helpfull!

thanks again.

2010/9/27 Daniel-Constantin Mierla <miconda@gmail.com>
btw, if you want to install from sources, here is a tutorial for 3.0.x:
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git

If you work with debian or ubuntu, there are apt repos for them:

http://www.kamailio.org/dokuwiki/doku.php/packages:debs

Cheers,
Daniel


Hello,

the r-uri is not rewritten with ip address of the phone, I guess you don't use user location to locate the phone. Is the phone registered to kamailio?

You say about the code for re-invites where you have a t_relay with outbound proxy. Normally, that should go via record-routing. If that code is also for initial invites and you must do it in this way, then you need to rewrite the r-uri domain and port to match phone's ip and port.

I suggest you use kamailio 3.0.x with default config file. It is easy to enable features such as authentication and use location. Create accounts for you phones, set them to register to kamailio and make calls. Then adapt the config to meet extra needs you may have.

Cheers,
Daniel

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:


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!

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Daniel-Constantin Mierla
http://www.asipto.com

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


-- 
Daniel-Constantin Mierla
http://www.asipto.com