[asterisk-users] invite authentication error !?

Fri Sep 30 02:54:42 CDT 2011

hi, alex
thanks for your kindly reply.
the remote proxy do not use domain as realm, the following  is part of the message

210.83.80.xxx is the ip of asterisk
202.104.188.xx is the ip of remote proxy sip server ( not asterisk）

--------------------------------------------------------------
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP 210.83.80.xxx:8090;branch=z9hG4bK15264427
From: "18971416745" <sip:555 at 210.83.80.xxx:8090>;tag=as04f73506
Call-ID: 0672dae66b88e08038259ee07f2ecd6d at 210.83.80.xxx:8090
To: <sip:01897141xxxx at 202.104.188.xx:5060>;tag=28094711162141020641417313
Contact: <sip:202.104.188.xx:5060;transport=udp>
Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="131719964128160427109472139339"
Content-Length: 0

INVITE sip:01897141xxxx at 202.104.188.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 210.83.80.xxx:8090;branch=z9hG4bK16d8cbb0
Max-Forwards: 70
From: "1897141xxxx" <sip:555 at 210.83.80.xxx:8090>;tag=as04f73506
To: <sip:01897141xxxx at 202.104.188.5:5060>
Contact: <sip:555 at 210.83.80.xxx:8090>
Call-ID: 0672dae66b88e08038259ee07f2ecd6d at 210.83.80.xxx:8090
CSeq: 103 INVITE
User-Agent: wsvoip2.0.1
Proxy-Authorization: Digest username="555", realm="VoipSwitch", algorithm=MD5, uri="sip:01897141xxxx at 202.104.188.5:5060", nonce="131719964128160427109472139339", response="a7794045c0248461332e9393332699ef"
Date: Wed, 28 Sep 2011 08:47:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 293

2011-09-30 

cnasterisk 

>发件人： Alex Balashov 
>发送时间： 2011-09-30  15:36:29 
>收件人： Asterisk Users Mailing List - Non-Commercial Discussion 
>抄送： asterisk-users 
>主题： Re: [asterisk-users] invite authentication error !? 
>
This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication "realm" (domain) is part of the authentication credentials.  So, what you have in your 'fromdomain' and 'host' settings on the peer does matter.

--
This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness.

Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Sep 30, 2011, at 3:16 AM, "cnasterisk" <cnasterisk at 163.com> wrote:

hi,
   Dear all.
   I setted a sip account on a sip trunk. when a  client call via this sip trunk, asterisk call failed on this trunk.
I have captured the sip messages on the host where asterisk located, and found that:

1. asterisk send a INVITE message to remote sip proxy without "proxy-authorization" field.
2. the remote sip proxy send back a " SIP/2.0 407 Proxy Authentication Required" message.
3. asterisk send a INVITE message with  "proxy-authorization" field.
4. remote proxy send back a "403(Forbidden)" message， that is mean "wrong password"

I also tested the sip account on a softphone, it works normal!

why this happed？ and how can i solve it?

2011-09-30 

kevin
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