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@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@gmail.com> *Reply-To: *<freeswitch-users@lists.freeswitch.org> *Date: *Tue, 11 Nov 2008 17:29:50 +0800 *To: *<freeswitch-users@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@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101>;transport=udp SIP/2.0 Call-ID: ef1e16976ae32f0f011de0db2ab5804b@192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001@192.168.253.101> <mailto:sip%3A1001@192.168.253.101> <sip%3A1001@192.168.253.101>>;tag=5919 To: <sip:1002@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101> > Via: SIP/2.0/UDP 192.168.253.101:7620 ;branch=z9hG4bKcfac6aaa7ee4a96457335bdad787cf31 Max-Forwards: 2 Contact: <sip:1001@192.168.253.101:7620;transport=udp> Accept: audio/gsm,audio/x-gsm,text/plain Content-Length: 0 ------
and I tried,
------- INVITE sip:1002@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101>;transport=udp SIP/2.0 Call-ID: 7a612541e08e583e0069c988c8323425@192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001@192.168.253.101> <mailto:sip%3A1001@192.168.253.101> <sip%3A1001@192.168.253.101>>;tag=7387 To: <sip:1002@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101> > Via: SIP/2.0/UDP 192.168.253.101:6453 ;branch=z9hG4bK7ccbc7f8e7bc0c4ac0483cca54a48489 Max-Forwards: 2 Contact: <sip:1001@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@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101>;transport=udp SIP/2.0 Call-ID: 12e7c3af8d94ba048789f75e7036ea74@192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001@192.168.253.101> <mailto:sip%3A1001@192.168.253.101> <sip%3A1001@192.168.253.101>>;tag=6309 To: <sip:1002@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101> > Via: SIP/2.0/UDP 192.168.253.101:4487 ;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd Max-Forwards: 2 Contact: <sip:1001@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@192.168.253.101 <sip%3A1001@192.168.253.101> <mailto:sip%3A1001@192.168.253.101> <sip%3A1001@192.168.253.101>>;tag=6309 To: <sip:1002@192.168.253.101 <sip%3A1002@192.168.253.101> <mailto:sip%3A1002@192.168.253.101> <sip%3A1002@192.168.253.101>>;tag=H1DH6S41c0U2N Call-ID: 12e7c3af8d94ba048789f75e7036ea74@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