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<TITLE>Re: [Freeswitch-users] &nbsp;Receiving 406 From Freeswitch....Any Clues?</TITLE>
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<OL><LI><FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>you need to include an SDP or enable 3pcc support on the sip profile... 
</SPAN></FONT></FONT><LI><FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>the 3rd invite looks like it doesn&#8217;t include any codecs (or atleast any I am familiar with)<BR>
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<HR ALIGN=CENTER SIZE="3" WIDTH="95%"><B>From: </B>Adeel Ansari &lt;adeel.gnome@gmail.com&gt;<BR>
<B>Reply-To: </B>&lt;freeswitch-users@lists.freeswitch.org&gt;<BR>
<B>Date: </B>Tue, 11 Nov 2008 17:29:50 +0800<BR>
<B>To: </B>&lt;freeswitch-users@lists.freeswitch.org&gt;<BR>
<B>Subject: </B>[Freeswitch-users] &nbsp;Receiving 406 From Freeswitch....Any Clues?<BR>
<BR>
Hi, I am getting 406, no matter what. I have tried 3 different INVITEs.<BR>
<BR>
------<BR>
<B>INVITE sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> ;transport=udp SIP/2.0<BR>
Call-ID: ef1e16976ae32f0f011de0db2ab5804b@192.168.253.101<BR>
CSeq: 1 INVITE<BR>
From: &lt;sip:1001@192.168.253.101 <a href="mailto:sip%3A1001@192.168.253.101">&lt;mailto:sip%3A1001@192.168.253.101&gt;</a> &gt;;tag=5919<BR>
To: &lt;sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> &gt;<BR>
Via: SIP/2.0/UDP 192.168.253.101:7620;branch=z9hG4bKcfac6aaa7ee4a96457335bdad787cf31<BR>
Max-Forwards: 2<BR>
Contact: &lt;sip:1001@192.168.253.101:7620;transport=udp&gt;<BR>
Accept: audio/gsm,audio/x-gsm,text/plain<BR>
Content-Length: 0<BR>
------<BR>
<BR>
and I tried,<BR>
<BR>
-------<BR>
INVITE sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> ;transport=udp SIP/2.0<BR>
Call-ID: 7a612541e08e583e0069c988c8323425@192.168.253.101<BR>
CSeq: 1 INVITE<BR>
From: &lt;sip:1001@192.168.253.101 <a href="mailto:sip%3A1001@192.168.253.101">&lt;mailto:sip%3A1001@192.168.253.101&gt;</a> &gt;;tag=7387<BR>
To: &lt;sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> &gt;<BR>
Via: SIP/2.0/UDP 192.168.253.101:6453;branch=z9hG4bK7ccbc7f8e7bc0c4ac0483cca54a48489<BR>
Max-Forwards: 2<BR>
Contact: &lt;sip:1001@192.168.253.101:6453;transport=udp&gt;<BR>
Accept: audio/gsm,audio/x-gsm,text/plain<BR>
Authorization: Digest username=&quot;1001&quot;,realm=&quot;192.168.253.101 <a href="http://192.168.253.101">&lt;http://192.168.253.101&gt;</a> &quot;,uri=&quot;sip:192.168.253.101:5060 <a href="http://192.168.253.101:5060">&lt;http://192.168.253.101:5060&gt;</a> &quot;,algorithm=MD5,opaque=&quot;&quot;,nonce=&quot;b1f5e8d8-afd0-11dd-ad17-f7c4d6a988fd&quot;,response=&quot;2c70f260de2292d9e01406bdd1f90e28&quot;<BR>
Content-Length: 0<BR>
-------<BR>
<BR>
and then I tried,<BR>
<BR>
------<BR>
INVITE sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> ;transport=udp SIP/2.0<BR>
Call-ID: 12e7c3af8d94ba048789f75e7036ea74@192.168.253.101<BR>
CSeq: 1 INVITE<BR>
From: &lt;sip:1001@192.168.253.101 <a href="mailto:sip%3A1001@192.168.253.101">&lt;mailto:sip%3A1001@192.168.253.101&gt;</a> &gt;;tag=6309<BR>
To: &lt;sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> &gt;<BR>
Via: SIP/2.0/UDP 192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd<BR>
Max-Forwards: 2<BR>
Contact: &lt;sip:1001@192.168.253.101:4487;transport=udp&gt;<BR>
Accept: audio/gsm,audio/x-gsm,text/plain<BR>
Authorization: Digest username=&quot;1001&quot;,realm=&quot;192.168.253.101 <a href="http://192.168.253.101">&lt;http://192.168.253.101&gt;</a> &quot;,uri=&quot;sip:192.168.253.101:5060 <a href="http://192.168.253.101:5060">&lt;http://192.168.253.101:5060&gt;</a> &quot;,algorithm=MD5,opaque=&quot;&quot;,nonce=&quot;85c12b22-afd2-11dd-ad17-f7c4d6a988fd&quot;,response=&quot;c8d6083112fc68b6e53d7e5b38e990eb&quot;<BR>
Content-Type: application/sdp<BR>
Content-Length: 111<BR>
<BR>
v=0<BR>
o=1001 279445 280814 IN IP4 192.168.253.101 <a href="http://192.168.253.101">&lt;http://192.168.253.101&gt;</a> <BR>
s=-<BR>
c=IN IP4 192.168.253.101 <a href="http://192.168.253.101">&lt;http://192.168.253.101&gt;</a> <BR>
t=0 0<BR>
m=audio 1436 RTP/AVP <BR>
------<BR>
<BR>
<BR>
But I always receive the same 406. Any idea. The response I am receiving from Freeswitch is,<BR>
<BR>
-------<BR>
received response : SIP/2.0 406 Not Acceptable<BR>
Via: SIP/2.0/UDP 192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd<BR>
From: &lt;sip:1001@192.168.253.101 <a href="mailto:sip%3A1001@192.168.253.101">&lt;mailto:sip%3A1001@192.168.253.101&gt;</a> &gt;;tag=6309<BR>
To: &lt;sip:1002@192.168.253.101 <a href="mailto:sip%3A1002@192.168.253.101">&lt;mailto:sip%3A1002@192.168.253.101&gt;</a> &gt;;tag=H1DH6S41c0U2N<BR>
Call-ID: 12e7c3af8d94ba048789f75e7036ea74@192.168.253.101<BR>
CSeq: 1 INVITE<BR>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M<BR>
Accept: application/sdp<BR>
Accept-Encoding: <BR>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE,REGISTER,INFO,PUBLISH<BR>
Supported: 100rel,timer,precondition,path,replaces<BR>
Allow-Events: talk,presence,dialog,call-info,sla,include-session-description,presence.winfo,message-summary<BR>
Content-Length: 0<BR>
-------<BR>
<BR>
Thanks.<BR>
<BR>
-- <BR>
Best,<BR>
Adeel Ansari<BR>
<BR>
<a href="http://www.linkedin.com/in/adeelansari">http://www.linkedin.com/in/adeelansari</a><BR>
<BR>
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