[Freeswitch-users] Receiving 406 From Freeswitch....Any Clues?

Ken Rice krice at suspicious.org
Tue Nov 11 01:38:30 PST 2008


1. you need to include an SDP or enable 3pcc support on the sip profile...
2. the 3rd invite looks like it doesn¹t include any codecs (or atleast any I
am familiar with)



From: Adeel Ansari <adeel.gnome at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 11 Nov 2008 17:29:50 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users]  Receiving 406 From Freeswitch....Any Clues?

Hi, I am getting 406, no matter what. I have tried 3 different INVITEs.

------
INVITE sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101>
;transport=udp SIP/2.0
Call-ID: ef1e16976ae32f0f011de0db2ab5804b at 192.168.253.101
CSeq: 1 INVITE
From: <sip:1001 at 192.168.253.101 <mailto:sip%3A1001 at 192.168.253.101>
>;tag=5919
To: <sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101> >
Via: SIP/2.0/UDP 
192.168.253.101:7620;branch=z9hG4bKcfac6aaa7ee4a96457335bdad787cf31
Max-Forwards: 2
Contact: <sip:1001 at 192.168.253.101:7620;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Content-Length: 0
------

and I tried,

-------
INVITE sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101>
;transport=udp SIP/2.0
Call-ID: 7a612541e08e583e0069c988c8323425 at 192.168.253.101
CSeq: 1 INVITE
From: <sip:1001 at 192.168.253.101 <mailto:sip%3A1001 at 192.168.253.101>
>;tag=7387
To: <sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101> >
Via: SIP/2.0/UDP 
192.168.253.101:6453;branch=z9hG4bK7ccbc7f8e7bc0c4ac0483cca54a48489
Max-Forwards: 2
Contact: <sip:1001 at 192.168.253.101:6453;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Authorization: Digest username="1001",realm="192.168.253.101
<http://192.168.253.101> ",uri="sip:192.168.253.101:5060
<http://192.168.253.101:5060>
",algorithm=MD5,opaque="",nonce="b1f5e8d8-afd0-11dd-ad17-f7c4d6a988fd",respo
nse="2c70f260de2292d9e01406bdd1f90e28"
Content-Length: 0
-------

and then I tried,

------
INVITE sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101>
;transport=udp SIP/2.0
Call-ID: 12e7c3af8d94ba048789f75e7036ea74 at 192.168.253.101
CSeq: 1 INVITE
From: <sip:1001 at 192.168.253.101 <mailto:sip%3A1001 at 192.168.253.101>
>;tag=6309
To: <sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101> >
Via: SIP/2.0/UDP 
192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd
Max-Forwards: 2
Contact: <sip:1001 at 192.168.253.101:4487;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Authorization: Digest username="1001",realm="192.168.253.101
<http://192.168.253.101> ",uri="sip:192.168.253.101:5060
<http://192.168.253.101:5060>
",algorithm=MD5,opaque="",nonce="85c12b22-afd2-11dd-ad17-f7c4d6a988fd",respo
nse="c8d6083112fc68b6e53d7e5b38e990eb"
Content-Type: application/sdp
Content-Length: 111

v=0
o=1001 279445 280814 IN IP4 192.168.253.101 <http://192.168.253.101>
s=-
c=IN IP4 192.168.253.101 <http://192.168.253.101>
t=0 0
m=audio 1436 RTP/AVP
------


But I always receive the same 406. Any idea. The response I am receiving
from Freeswitch is,

-------
received response : SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 
192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd
From: <sip:1001 at 192.168.253.101 <mailto:sip%3A1001 at 192.168.253.101>
>;tag=6309
To: <sip:1002 at 192.168.253.101 <mailto:sip%3A1002 at 192.168.253.101>
>;tag=H1DH6S41c0U2N
Call-ID: 12e7c3af8d94ba048789f75e7036ea74 at 192.168.253.101
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M
Accept: application/sdp
Accept-Encoding: 
Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE,RE
GISTER,INFO,PUBLISH
Supported: 100rel,timer,precondition,path,replaces
Allow-Events: 
talk,presence,dialog,call-info,sla,include-session-description,presence.winf
o,message-summary
Content-Length: 0
-------

Thanks.

-- 
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari


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