I think as Scott mentioned (and I think I did earlier...  or at least I
thought it), setup the asterisk box as an unmanaged gateway in the sipX
box.  Then the two should be able to talk.

> -----Original Message-----
> From: [email protected] [mailto:sipx-users-
> [email protected]] On Behalf Of [email protected]
> Sent: Tuesday, December 16, 2008 4:48 PM
> To: Scott Lawrence
> Cc: [email protected]
> Subject: Re: [sipx-users] Help: Asterisk Registration in to Sipx - 401
> Error
> 
> Quoting Scott Lawrence <[email protected]>:
> 
> >
> > On Tue, 2008-12-16 at 18:30 -0200, [email protected] wrote:
> >>
> >> Looking at the sipregistrar directly there seems to be an error
with
> >> invalid
> >> nonce. But I haven't a clue to what this means...
> >
> > It usually means that the user agent changed either the call-id
value
> or
> > the From tag parameter value in the request between the request that
> got
> > a 401 response and the request it sent with authentication.  This is
> an
> > error and won't work.
> 
> I noticed with the other hardware (non Asterisk) that registers
> correctly, that it sends 1st register gets a 401 and then sends a 2nd
> register (with different info) and gets registered. The Asterisk sends
> the same register over and over. So there is no second attempt, I
> think that's the problem.
> 
> 
> >
> > In your earlier description, you said:
> >
> >> Now I need to connect a client that has an Asterisk Server with 10
> >> phones. I created a user in Sipxpbx. I created a Sip Trunk Route in
> >> the
> >> Asterisk to the Sipxpbx,
> >
> > If you're requirement is to route calls _to_ that Asterisk server in
> > order to call those 10 lines, you could do that by configuring the
> > address of the Asterisk server as an Unmanaged Gateway and create a
> dial
> > plan that routes those numbers there....
> >
> 
> No my problem is to route call out of the asterisk in to the Sipxpbx
> so I can terminate them on my GSM Gateways.
> 
> 
> Thanks again for your advice Scott,
> Alberto
> 
> 
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