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i faced a similar situation with my ISP .. they block INBOUND UDP port 5060 which means if i try to register.. the server would receive my registration message.. but when it sends the acknowledgement .. the ISP Firewall rejects the message so the server responds with Unauthorized.. i simply changed the port on the server to 5070 and set my dialer to listen to port 5070 as well (for inbound messages) and this solved my issue.<div>that was my situation.. so your problem is in the firewall settings.. just try to look at it and see what is missing.. </div><div>and by the way when you send all of your IP sections XXX no one will assist you as no one will know who is talking to whom.. </div><div>just like if you go to a doctor with a prostate problem.. you can't tell him that you won't remove your clothes off ;)</div><div>regards<br><br>-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 <br><br><br><br>> Date: Wed, 23 Jun 2010 08:44:21 -0400<br>> From: geisj@pagestation.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: [asterisk-users] help with sip 401 unauthorized<br>> <br>> I am getting a SIP 401 unauthorized message.<br>> <br>> My public IP or PIP is being pre-routed with iptables to goto an <br>> internal IP or IIP<br>> All the polycom phones in the office point to the IIP. they work fine.<br>> I have 2 external phones that are registering to the PIP. I see the <br>> register attempt<br>> as I am getting the 401 unauthorized message. For the 2 external phones <br>> both have nat=1 enabled.<br>> <br>> remote phone (192.X.X.X) ----> GW ----> internet ----> PIP (prerouted) <br>> (74.X.X.X) ----> internal server (192.X.X.X)<br>> <br>> This used to work before I moved the server inside the firewall. What <br>> special setting do I need to<br>> enable to get this working.<br>> <br>> Thanks,<br>> <br>> Jerry<br>> <br>> <--- Transmitting (NAT) to X.X.X.X:1024 ---><br>> SIP/2.0 401 Unauthorized<br>> Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6ea01bc7;received=X.X.X.X<br>> From: <sip:xxx@X.X.X.X.;user=phone><br>> To: <sip:xxx@X.X.X.X;user=phone>;tag=as21ab1732<br>> Call-ID: 000ff78d-ebb20007-22675f66-5da7e6b7@X.X.X.X<br>> CSeq: 1196 REGISTER<br>> User-Agent: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>> Supported: replaces<br>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c6a6002"<br>> Content-Length: 0<br>> <br>> [XXX]<br>> type=friend<br>> username=XXX<br>> secret=<br>> dtmfmode=RFC2833<br>> host=dynamic<br>> context=external<br>> rtptimeout=60<br>> qualify=no<br>> canreinvite=yes<br>> nat=yes<br>> disallow=all<br>> allow=ulaw<br>> allow=alaw<br>> allow=gsm<br>> <br>> <br>> <br>> -- <br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> <br /><hr />The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. <a href='http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4' target='_new'>Get busy.</a></body>
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