Hi Daniel-Constantin, thank for your quick response. This is the link to the SIP trace: 

http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa

I didn't send it through the list cause the body size needed approval.
The trace is a call from the extension 1090 to 1020. Kamailio is listening at 192.168.15.11:5060 and asterisk at 192.168.15.11:5080. Additionally I have pasted below a short CLI trace on asterisk showing up a NoOp with the caller id followed by the dial and the first invite. 
I really appreciate you help. Regards.

Lucas


CLI trace:


    -- Executing [1020@longdistance:1] NoOp("SIP/1090-00000037", "Callerid number: 1090      Name: Lucas Voice ") in new stack
    -- Executing [1020@longdistance:2] Dial("SIP/1090-00000037", "SIP/1020") in new stack
[Oct 12 10:44:26] DEBUG[17631]: chan_sip.c:3462 update_call_counter: Call to peer '1020' is 1 out of 10
Audio is at 192.168.15.11 port 18106
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.15.11:5060:
Via: SIP/2.0/UDP 192.168.15.11:5080;branch=z9hG4bK7c1bd27b;rport
From: "Lucas Voice" <sip:1020@192.168.15.11>;tag=as1a1d0e0e
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Oct 2010 14:44:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 287






On Mon, Oct 11, 2010 at 7:36 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
 Hello,


On 10/11/10 11:28 PM, Lucas Alvarez wrote:
Hi, I'm having a problem with the caller ID, I have implemented an
integration between asterisk and kamailio following this tutorial:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
and the problem is that when I call from extension, let's say 1000, to
another extension, let's say 2000, the callerid number is always the
number I'm calling, in this case 2000. Using xlog and printing $fu,
$fU variables I realize that when the call came from asterisk to the
destination number,  kamailio changes the "From" headers. I will
appreciate any kind of help.
Regards.

can you take a SIP trace of such case on kamailio server? preferably with ngrep:

ngrep -d any -qt -W byline port 5060

Cheers,
Daniel

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