<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On 26 Mar 2014, at 19:06, Mickael MONSIEUR <<a href="mailto:mickael.monsieur@gmail.com">mickael.monsieur@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><pre style="font-size: 1.1em; overflow: auto;"><code>Hi,</code></pre><pre style="font-size: 1.1em; overflow: auto;"><code>When Asterisk create a register it uses by default extension "s".</code></pre></div></blockquote>Not if you add an "extension" according to the syntax below.<br><blockquote type="cite"><div dir="ltr">
<pre style="font-size: 1.1em; overflow: auto;"><code>register => user[:secret[:authuser]]@host[:port][/extension]</code></pre><pre style="overflow:auto"><code><font>But that we do not care. I want to replace the 's' from my SIP INVITE.
</font></code></pre></div></blockquote><br><blockquote type="cite"><div dir="ltr"><pre style="overflow:auto"><code><font>How?<span style="font-size:1.1em"><br></span></font></code></pre></div></blockquote>Read the docs... ;-)</div><div><br></div><div>/O<br><blockquote type="cite"><div class="gmail_extra"><br><br><div class="gmail_quote">2014-03-26 16:40 GMT+01:00 Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word"><br><div><div>On 26 Mar 2014, at 16:32, Mickael MONSIEUR <<a href="mailto:mickael.monsieur@gmail.com" target="_blank">mickael.monsieur@gmail.com</a>> wrote:</div>
<br><blockquote type="cite"><div dir="ltr">Hello,<div><br><div><div>I installed kamailio and asterisk with the tutorial of asipto. </div><div>For alias numbers I configured the module alias_db. </div><div><br></div><div>Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) </div>
<div><br></div><div>Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ?</div></div>
</div></div></blockquote><div><br></div>Who sets the "s" in the request URI? Asterisk use "s" when there's no extension given in the dialplan. </div><div><br></div><div>/O<br><blockquote type="cite">
<div dir="ltr"><div>
<div><br></div><div>Example:</div><div><br></div><div><div>INVITE <a href="http://sip:s@10.1.0.191:5060/" target="_blank">sip:s@10.1.0.191:5060</a> SIP/2.0</div><div>Max-Forwards: 69</div><div>From: "0475XXXXXX" <<a href="mailto:sip%3A1053212@sip.domain.com" target="_blank">sip:1053212@sip.domain.com</a>>;tag=as7df9ab18</div>
<div>To: <<a>sip:02XXXXXX@kamailioIP:5060</a>></div><div>Contact: <<a>sip:1053212@asteriskIP:5060</a>></div><div>Call-ID: <a href="mailto:344d42bd16975a54141d11f635bdfc71@sip.domain.com" target="_blank">344d42bd16975a54141d11f635bdfc71@sip.domain.com</a></div>
<div>CSeq: 102 INVITE</div><div>Date: Wed, 26 Mar 2014 15:06:01 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Content-Type: application/sdp</div>
<div>Content-Length: 252</div></div><div><br></div></div><div>Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ...<br>
</div><div><br></div><div>Thank you,</div><div>Mickael</div></div>
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