[asterisk-users] help with sip 401 unauthorized

Tarek Sawah tareksawah at hotmail.com
Wed Jun 23 11:10:57 CDT 2010


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.that was my situation.. so your problem is in the firewall settings.. just try to look at it and see what is missing.. 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.. 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 ;)regards

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993  



> Date: Wed, 23 Jun 2010 08:44:21 -0400
> From: geisj at pagestation.com
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] help with sip 401 unauthorized
> 
> I am getting a SIP 401 unauthorized message.
> 
> My public IP or PIP is being pre-routed with iptables to goto an 
> internal IP or IIP
> All the polycom phones in the office point to the IIP. they work fine.
> I have 2 external phones that are registering to the PIP. I see the 
> register attempt
> as I am getting the 401 unauthorized message.  For the 2 external phones 
> both have nat=1 enabled.
> 
> remote phone (192.X.X.X) ----> GW ----> internet ----> PIP (prerouted) 
> (74.X.X.X) ----> internal server (192.X.X.X)
> 
> This used to work before I moved the server inside the firewall. What 
> special setting do I need to
> enable to get this working.
> 
> Thanks,
> 
> Jerry
> 
> <--- Transmitting (NAT) to X.X.X.X:1024 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6ea01bc7;received=X.X.X.X
> From: <sip:xxx at X.X.X.X.;user=phone>
> To: <sip:xxx at X.X.X.X;user=phone>;tag=as21ab1732
> Call-ID: 000ff78d-ebb20007-22675f66-5da7e6b7 at X.X.X.X
> CSeq: 1196 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c6a6002"
> Content-Length: 0
> 
> [XXX]
> type=friend
> username=XXX
> secret=
> dtmfmode=RFC2833
> host=dynamic
> context=external
> rtptimeout=60
> qualify=no
> canreinvite=yes
> nat=yes
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> 
> 
> 
> -- 
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