<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>On 14 Mar 2011, at 15:58, Jonas Kellens wrote:</div><blockquote type="cite"><div text="#000000" bgcolor="#ffffff"><font face="Helvetica, Arial, sans-serif">dialplan :<br>
<br>
exten => 67121212,1,NoOp()<br>
exten => 67121212,n,Set(CALLERID(all)="32596666" <32596666>)<br>
exten => 67121212,n,SIPAddHeader(P-Preferred-Identity:
<a class="moz-txt-link-rfc2396E" href="sip:32596666/;user=phone"><sip:32596666\;user=phone></a>)<br>
exten => 67121212,n,SIPAddHeader(Privacy: id)<br>
exten => 67121212,n,Dial(SIP/32596666/</font><font face="Helvetica, Arial, sans-serif">67121212</font><font face="Helvetica, Arial, sans-serif">)<br>
<br>
<br>
CLI :<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:67121212@sip.voip.tld">sip:67121212@sip.voip.tld</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51<br>
From: "VC" <a class="moz-txt-link-rfc2396E" href="sip:voip2@sip.voip.tld"><sip:voip2@sip.voip.tld></a>;tag=cb415736707fb109o2<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:67121212@sip.voip.tld"><sip:67121212@sip.voip.tld></a><br>
Remote-Party-ID: "VC"
<a class="moz-txt-link-rfc2396E" href="sip:voip2@sip.voip.tld"><sip:voip2@sip.voip.tld></a>;screen=yes;party=calling<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:2a80707a-bdb7c895@192.168.1.106">2a80707a-bdb7c895@192.168.1.106</a><br>
CSeq: 101 INVITE<br>
Max-Forwards: 70<br>
Contact: "VC" <a class="moz-txt-link-rfc2396E" href="sip:voip2@192.168.1.106:5063"><sip:voip2@192.168.1.106:5063></a><br>
Expires: 240<br>
User-Agent: Linksys/SPA941-5.1.8<br>
Content-Length: 399<br>
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br>
Supported: replaces<br>
Content-Type: application/sdp<br></font></div></blockquote></div><br><div>That's the invite from the phone, not from Asterisk... no?</div><div><br></div><div>S</div></body></html>