asterisk users - Feb 2005 - 407 Proxy Authentication Required

Hi everybody:

I configured my Asterisk to register to my VoIP provider, and I can make
outgoing calls, but I can't receive any calls with it.
I used Ethereal to sniff the activity of it, and I found something that
might be causing the problem:
When my provider's gateway does the "Request: INVITE
mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407
Proxy
Authentication Required", (log line 10) but my provider's gateway never
sends this info back, so my Asterisk keeps on asking for the Authentication,
and it never comes back... so it gives a "time-out" (I guess).

What I need to know is how to configure my Asterisk for not to ask for
Authentication.

Here's the log if you would like to see what's going on:
192.168.1.116 = ATA from which I'm calling
"mynumber@my-voip-provider.tld"
192.168.1.48  = My Asterisk server

Thank you ;)

No.     Time        Source                Destination           Protocol
Info
      1 0.000000    192.168.1.116         VoIP Prov IP          SIP/SDP
Request: INVITE sip:mynumber@my-voip-provider.tld, with session description
      2 0.369430    VoIP Prov IP          192.168.1.116         SIP
Status: 100 Trying
      3 0.401052    VoIP Prov IP          192.168.1.116         SIP
Status: 407 Proxy Authentication Required
      4 0.407666    192.168.1.116         VoIP Prov IP          SIP
Request: ACK sip:mynumber@my-voip-provider.tld
      5 0.414146    192.168.1.116         VoIP Prov IP          SIP/SDP
Request: INVITE sip:mynumber@my-voip-provider.tld, with session description
      6 0.907932    192.168.1.116         VoIP Prov IP          SIP/SDP
Request: INVITE sip:mynumber@my-voip-provider.tld, with session description
      7 1.541468    VoIP Prov IP          192.168.1.116         SIP
Status: 100 Trying
      8 1.563302    VoIP Prov IP          192.168.1.116         SIP
Status: 180 Ringing
      9 1.635021    VoIP Prov IP          192.168.1.48          SIP/SDP
Request: INVITE sip:mynumber@192.168.1.48:5060;maddr=192.168.1.48, with
session description
     10 1.636719    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     11 1.653490    VoIP Prov IP          192.168.1.116         SIP
Status: 100 Trying
     12 1.686395    VoIP Prov IP          192.168.1.48          SIP
Request: OPTIONS sip:ceincasa02@192.168.1.116:5061
     13 2.637223    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     14 3.647291    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     15 3.887926    VoIP Prov IP          192.168.1.116         SIP
Request: OPTIONS sip:ceincasa01@192.168.1.116
     16 3.897185    192.168.1.116         VoIP Prov IP          SIP
Status: 200 OK
     17 4.119698    VoIP Prov IP          192.168.1.48          SIP
Request: OPTIONS sip:mynumber@192.168.1.48
     18 4.120788    192.168.1.48          VoIP Prov IP          SIP
Status: 200 OK
     19 4.647336    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     20 5.647409    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     21 6.647465    192.168.1.48          VoIP Prov IP          SIP
Status: 407 Proxy Authentication Required
     22 7.657954    VoIP Prov IP          192.168.1.116         SIP
Status: 180 Ringing

I am getting error: Call rejected: 407 Proxy Authentication Required - if a
user is trying to call using * over a long latency network using sjphone &
snom.
How to overcome this..!!
Pls advice..!
Shahan




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There is a problem between 2 asterisk servers in message 407.  In the
normal flow, it should something like diagram1.  However, in my case,
I got the situation as diagram 2 and the call dropped finally.  Does
anyone have the same problem with me?  How to solve the problem?
Anyone can help?

UA1	AST1	AST2	UA2
----INVITE-->
<---407----
---ACK------>
----INVITE-->
	-----INVITE-->
		-----INVITE-->
-----------continue as normal----------
diagram1


UA1	AST1	AST2	UA2
----INVITE-->
<---407----
---ACK------>
----INVITE-->
	-----INVITE-->
	<-----407-----
	-----ACK------>
----------call drop after timeout---------
diagram2