<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">Hi Daniel-Constantin, thank for your quick response. This is the link to the SIP trace: </span><div><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><br>
</span></div><div><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><font size="3"><a href="http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa">http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa</a></font></span><div>
<a href="http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa"></a><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse; font-size: medium;"><br>
</span></font><div><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">I didn't send it through the list cause the body size needed approval.</span><div>
<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">The trace is a call from the extension 1090 to 1020. Kamailio is listening at </span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><a href="http://192.168.15.11:5060/" target="_blank" style="color: rgb(0, 0, 204); ">192.168.15.11:5060</a></span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "> and asterisk at </span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><a href="http://192.168.15.11:5080/" target="_blank" style="color: rgb(0, 0, 204); ">192.168.15.11:5080</a></span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">. 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. </span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><br>
</span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">I really appreciate you help. Regards.</span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><br>
</span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; "><br></span><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">Lucas</span></div>
<div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse; "><br></span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br>
</span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;">CLI trace:</span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br>
</span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><div>
-- Executing [1020@longdistance:1] NoOp("SIP/1090-00000037", "Callerid number: 1090 Name: Lucas Voice ") in new stack</div><div> -- Executing [1020@longdistance:2] Dial("SIP/1090-00000037", "SIP/1020") in new stack</div>
<div>[Oct 12 10:44:26] DEBUG[17631]: chan_sip.c:3462 update_call_counter: Call to peer '1020' is 1 out of 10</div><div>Audio is at 192.168.15.11 port 18106</div><div>Adding codec 0x4 (ulaw) to SDP</div><div>Adding codec 0x8 (alaw) to SDP</div>
<div>Adding codec 0x2 (gsm) to SDP</div><div>Adding non-codec 0x1 (telephone-event) to SDP</div><div>Reliably Transmitting (no NAT) to <a href="http://192.168.15.11:5060">192.168.15.11:5060</a>:</div><div>INVITE <a href="http://sip:1020@192.168.15.11:5060">sip:1020@192.168.15.11:5060</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 192.168.15.11:5080;branch=z9hG4bK7c1bd27b;rport</div><div>From: "Lucas Voice" <<a href="mailto:sip%3A1020@192.168.15.11">sip:1020@192.168.15.11</a>>;tag=as1a1d0e0e</div><div>To: <<a href="http://sip:1020@192.168.15.11:5060">sip:1020@192.168.15.11:5060</a>></div>
<div>Contact: <<a href="http://sip:1020@192.168.15.11:5080">sip:1020@192.168.15.11:5080</a>></div><div>Call-ID: <a href="mailto:7278984921bca2d55477817467d99103@192.168.15.11">7278984921bca2d55477817467d99103@192.168.15.11</a></div>
<div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX</div><div>Max-Forwards: 70</div><div>Date: Tue, 12 Oct 2010 14:44:26 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</div><div>
Supported: replaces</div><div>Content-Type: application/sdp</div><div>Content-Length: 287</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></span></font><br><div class="gmail_quote">On Mon, Oct 11, 2010 at 7:36 PM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"> Hello,<div><div></div><div class="h5"><br>
<br>
On 10/11/10 11:28 PM, Lucas Alvarez wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi, I'm having a problem with the caller ID, I have implemented an<br>
integration between asterisk and kamailio following this tutorial:<br>
<a href="http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb</a><br>
and the problem is that when I call from extension, let's say 1000, to<br>
another extension, let's say 2000, the callerid number is always the<br>
number I'm calling, in this case 2000. Using xlog and printing $fu,<br>
$fU variables I realize that when the call came from asterisk to the<br>
destination number, kamailio changes the "From" headers. I will<br>
appreciate any kind of help.<br>
Regards.<br>
<br>
</blockquote></div></div>
can you take a SIP trace of such case on kamailio server? preferably with ngrep:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
Cheers,<br>
Daniel<br><font color="#888888">
<br>
-- <br>
Daniel-Constantin Mierla<br>
<a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br>
<br>
</font></blockquote></div><br></div></div></div></div>