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

Mon Sep 27 11:31:29 CEST 2010

  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 <http://192.168.10.140:5060> -> 
>> 192.168.10.150:5060 <http://192.168.10.150:5060>
>> INVITE sip:200 at 192.168.10.150 <mailto: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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto: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 
>> <http://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 <http://192.168.10.150:5060> -> 
>> 192.168.10.140:5060 <http://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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto: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 <http://192.168.10.150:5060> -> 
>> 192.168.10.160:5060 <http://192.168.10.160:5060>
>> INVITE sip:200 at 192.168.10.150 <mailto: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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto: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 
>> <http://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 <http://192.168.10.160:5060> -> 
>> 192.168.10.150:5060 <http://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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto: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 <http://192.168.10.160:5060> -> 
>> 192.168.10.150:5060 <http://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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto: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 <http://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 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.150 
>> <mailto:sip%3A200 at 192.168.10.150>>.
>>
>> I think that should be:
>>
>> From: "Sip Phone" <sip:100 at 192.168.10.150 
>> <mailto:sip%3A100 at 192.168.10.150>>;tag=d396005aaf3ab9a2o0.
>> To: "Mitel Phone" <sip:200 at 192.168.10.160 
>> <mailto: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 list
>> sr-users at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20100927/4ac5e0b9/attachment-0001.htm>