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

Tue Nov 11 01:58:47 PST 2008

Actually, I am just trying to build a minimal client/UA for Freeswitch. Do I
need to go into 3pcc or something. I remember I did play with 3pcc but at
that time I was implementing a sip stack and trying to come up with a
minimal softswitch. Am I misunderstanding something?

It sounds like, I need to include codecs. I have tried the similar INVITE
with a Voip provider directly, and send it with 0 content-length, and no sdp
at all. In response I got whole lot of things like codec, and all.
Therefore, I did the same for the Freeswtich too. However, I will try
including codecs.

Thanks.

On Tue, Nov 11, 2008 at 5:38 PM, Ken Rice <krice at suspicious.org> wrote:

>
>    1. you need to include an SDP or enable 3pcc support on the sip
>    profile...
>    2. the 3rd invite looks like it doesn't 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 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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 <sip%3A1001 at 192.168.253.101>
> <mailto:sip%3A1001 at 192.168.253.101> <sip%3A1001 at 192.168.253.101>>;tag=5919
> To: <sip:1002 at 192.168.253.101 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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 <sip%3A1001 at 192.168.253.101>
> <mailto:sip%3A1001 at 192.168.253.101> <sip%3A1001 at 192.168.253.101>>;tag=7387
> To: <sip:1002 at 192.168.253.101 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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> <http://192.168.253.101> ",uri="sip:
> 192.168.253.101:5060 <http://192.168.253.101:5060><http://192.168.253.101:5060>",algorithm=MD5,opaque="",nonce="b1f5e8d8-afd0-11dd-ad17-f7c4d6a988fd",response="2c70f260de2292d9e01406bdd1f90e28"
> Content-Length: 0
> -------
>
> and then I tried,
>
> ------
> INVITE sip:1002 at 192.168.253.101 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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 <sip%3A1001 at 192.168.253.101>
> <mailto:sip%3A1001 at 192.168.253.101> <sip%3A1001 at 192.168.253.101>>;tag=6309
> To: <sip:1002 at 192.168.253.101 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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> <http://192.168.253.101> ",uri="sip:
> 192.168.253.101:5060 <http://192.168.253.101:5060><http://192.168.253.101:5060>",algorithm=MD5,opaque="",nonce="85c12b22-afd2-11dd-ad17-f7c4d6a988fd",response="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><http://192.168.253.101>
> s=-
> c=IN IP4 192.168.253.101 <http://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 <sip%3A1001 at 192.168.253.101>
> <mailto:sip%3A1001 at 192.168.253.101> <sip%3A1001 at 192.168.253.101>>;tag=6309
> To: <sip:1002 at 192.168.253.101 <sip%3A1002 at 192.168.253.101>
> <mailto:sip%3A1002 at 192.168.253.101> <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,REGISTER,INFO,PUBLISH
> Supported: 100rel,timer,precondition,path,replaces
> Allow-Events:
> talk,presence,dialog,call-info,sla,include-session-description,presence.winfo,message-summary
> Content-Length: 0
> -------
>
> Thanks.
>
> --
> Best,
> Adeel Ansari
>
> http://www.linkedin.com/in/adeelansari
>
> ------------------------------
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>

-- 
Best,
Adeel Ansari

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