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Hello,<br>
<br>
the INVITE comes with that Caller ID set from Asterisk. It was very
unlikely Kamailio changes it unless you use uac module.<br>
<br>
I guess Asterisk in matching on source IP and port and happens to
select another (pretty much randomly) caller id.<br>
<br>
Try to use type=user in sipusers table.<br>
<br>
Another option is to get the caller id from incoming invite to
asterisk and set it for outgoing invite from asterisk.<br>
<br>
Let me know if any of these works.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<br>
On 10/12/10 5:14 PM, Lucas Alvarez wrote:
<blockquote
cite="mid:AANLkTim1UXHKFniy9QeTeDiu53BZ3P0103+GQfgr7xEA@mail.gmail.com"
type="cite"><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 moz-do-not-send="true"
href="http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa">http://www.euscorp.com/images/Wedpooi321989812j1DD9ddd9PaHw8XCa</a></font></span>
<div>
<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 moz-do-not-send="true"
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 moz-do-not-send="true"
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
moz-do-not-send="true"
href="http://192.168.15.11:5060">192.168.15.11:5060</a>:</div>
<div>INVITE <a moz-do-not-send="true"
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 moz-do-not-send="true"
href="mailto:sip%3A1020@192.168.15.11">sip:1020@192.168.15.11</a>>;tag=as1a1d0e0e</div>
<div>To: <<a moz-do-not-send="true"
href="http://sip:1020@192.168.15.11:5060">sip:1020@192.168.15.11:5060</a>></div>
<div>Contact: <<a moz-do-not-send="true"
href="http://sip:1020@192.168.15.11:5080">sip:1020@192.168.15.11:5080</a>></div>
<div>Call-ID: <a moz-do-not-send="true"
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
moz-do-not-send="true"
href="mailto:miconda@gmail.com">miconda@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt
0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;"> Hello,
<div>
<div class="h5"><br>
<br>
On 10/11/10 11:28 PM, Lucas Alvarez wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt
0pt 0pt 0.8ex; border-left: 1px solid rgb(204,
204, 204); 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 moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://www.asipto.com" target="_blank">http://www.asipto.com</a><br>
<br>
</font></blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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