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<font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I'm trying to page the Cisco SPA 941 by adding the SIP-header
Call-Info: answer-after=0<br>
<br>
dialplan :<br>
<br>
exten => _*XX*,n,SIPAddHeader("Call-Info: answer-after=0")<br>
<br>
<br>
SIP debug :<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:testcorp6@192.168.1.106:5064">sip:testcorp6@192.168.1.106:5064</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6090dca4;rport<br>
Max-Forwards: 70<br>
From: "GXP2010 Grandstream"
<a class="moz-txt-link-rfc2396E" href="sip:50@192.168.1.150"><sip:50@192.168.1.150></a>;tag=as530fd02f<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:testcorp6@192.168.1.106:5064"><sip:testcorp6@192.168.1.106:5064></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:50@192.168.1.150"><sip:50@192.168.1.150></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:1cf4575c13fb25a40338a2fa2ad22cc0@192.168.1.150">1cf4575c13fb25a40338a2fa2ad22cc0@192.168.1.150</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.10<br>
Date: Mon, 05 Sep 2011 10:21:15 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO<br>
Supported: replaces, timer<br>
<b>Call-Info: answer-after=0</b><br>
Content-Type: application/sdp<br>
Content-Length: 266<br>
<br>
<br>
In the SPA 941 I have : tab "User" > Auto Answer Page: yes<br>
</font><font face="Helvetica, Arial, sans-serif"><br>
<br>
But when calling *60* the SPA941 IP-phone just rings :-( in stead
of answering automatically through the speakers.<br>
<br>
<br>
Kind regards,<br>
Jonas.<br>
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