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

Daniel-Constantin Mierla miconda at gmail.com
Mon Sep 27 11:29:22 CEST 2010


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

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