In my configuration the Asterisk Box is a user, it sends calls to the
Sipxpbx.
Sipxpbx is doing the Trunk Termination, not the Origination.
I have had excellent results with Sipxpbx in our simulations and
I am beginning a test run in a practical environment. My tests show
that Sipxpbx can handle the large load that will be generated
by our Termination Services, where Asterisk Servers fail to handle
the throughput. But Sipxpbx has to be able to accept an Asterisk
box as a user, the same way it accepts the FXS and E1 SIP
gateways that we tested.
----- Original Message -----
From: "Picher, Michael" <[email protected]>
To: <[email protected]>; "Scott Lawrence" <[email protected]>
Cc: <[email protected]>
Sent: Tuesday, December 16, 2008 9:39 PM
Subject: RE: [sipx-users] Help: Asterisk Registration in to Sipx - 401 Error
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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