[SR-Users] Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?

Mon Sep 27 19:39:47 CEST 2010

Thanks Daniel!

I'm using Debian, so this is helpfull!

thanks again.

2010/9/27 Daniel-Constantin Mierla <miconda at 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:
>
>  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 at 192.168.10.150 <sip%3A200 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 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 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 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 at 192.168.10.150 <sip%3A200 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 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 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 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 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.150 <sip%3A200 at 192.168.10.150>>.
>
>  I think that should be:
>
>  From: "Sip Phone" <sip:100 at 192.168.10.150 <sip%3A100 at 192.168.10.150>
> >;tag=d396005aaf3ab9a2o0.
> To: "Mitel Phone" <sip:200 at 192.168.10.160 <sip%3A200 at 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 listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com
>
>
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