Hello,

the INVITE comes with that Caller ID set from Asterisk. It was very unlikely Kamailio changes it unless you use uac module.

I guess Asterisk in matching on source IP and port and happens to select another (pretty much randomly) caller id.

Try to use type=user in sipusers table.

Another option is to get the caller id from incoming invite to asterisk and set it for outgoing invite from asterisk.

Let me know if any of these works.

Cheers, Daniel

On 10/12/10 5:14 PM, Lucas Alvarez wrote:

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 http://192.168.15.11:5060/ and asterisk at 192.168.15.11:5080 http://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 http://192.168.15.11:5060: INVITE sip:1020@192.168.15.11:5060 http://sip:1020@192.168.15.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.11:5080;branch=z9hG4bK7c1bd27b;rport From: "Lucas Voice" <sip:1020@192.168.15.11 mailto:sip%3A1020@192.168.15.11>;tag=as1a1d0e0e To: <sip:1020@192.168.15.11:5060 http://sip:1020@192.168.15.11:5060> Contact: <sip:1020@192.168.15.11:5080 http://sip:1020@192.168.15.11:5080> Call-ID: 7278984921bca2d55477817467d99103@192.168.15.11 mailto:7278984921bca2d55477817467d99103@192.168.15.11 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 mailto: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