I don't understand what you mean by "remote call comes in and is
forwarded..."

Is this a call from a remote user, who has call forwarding set to the pstn?

Is this a call from the PSTN and the internal user has call forwarding
enabled to go our over the pstn?

In either case, if "normal" inbound and outbound calls works, then it would
be helpful to know what is different here.

"When" you have a provider that uses registration, it is customary to have
"register on initialization" checked, as Todd points out.

On Fri, Aug 20, 2010 at 2:33 AM, Todd Hodgen <[email protected]> wrote:

>  Don’t you want to check mark the Register on Init?  I’ve always found it
> doesn’t work unless that is marked……………..
>
>
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Michael Scheidell
> *Sent:* Thursday, August 19, 2010 2:19 PM
> *To:* Hearty, John
> *Cc:* '[email protected]'
> *Subject:* Re: [sipx-users] SIP digest auth and caller ID
>
>
>
> On 8/19/10 5:07 PM, Hearty, John wrote:
>
> I was assisting in troubleshooting problems doing SIP Trunking to sipXecs.
> When accessing the service over public internet, we require digest
> authentication.  When a User name for SIP authentication is configured and
> the “Use default asserted identity” box is checked, the value in the “User
> name” field gets put into the user part of the >From URI and
> P-Asserted-Identity URI.  This looks wrong.  The “User name” field appears
> to be overloaded, being used for both SIP credentials and subscriber
> identification.  Authentication credentials have nothing to do with
> subscriber identity.  The value in the “User name” field should only be used
> in the Authorization header.  It is completely unrelated to the identity of
> the caller.  Is this a bug being tracked?
>
>
>
> Below is text from the
> http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking page that
> supports what was observed.
>
>  If the ITSP does not recognize this field and uses the >From header for
> account identification, these features may not work as expected. if you
> select to "use default asserted identity", you must specify a user name so
> that the default *Asserted Identity* may be computed.
>
>
>
> and as clarification, I think this is version 4.2.0 sipx.
> from a 'sipx' itsp account setup, we have the account number and password
> in the itsp account fields, the 'register on init' CLEARED.
> The issues come in when a remote call come in, and is forwarded.  the
> callid that shows up on the phone that the call was forwarded to is the
> level3 'user'.
>
>  --
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-- 
======================
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