Scott,

Thank you very much for the detailed clarification.
Sipx in my case uses it's own dns, which is private (not available from 
internet).
But my test phone is "remote worker" behind remote nat. 
And remote in this case also means distant :) - about 5500 km between me and 
sipx.
And there is no split dns (at least yet)...

So I configured 217.106.156.51 (externally visible nated sipx address) as 
domain alias.

At the moment I can not ask to call from internal phone (which uses sipx as dns 
server), time gap is too big... 
On Monday I'll verify if it'll work as expected.

And a question: may be it's worth thinking about extending "domain alias" 
feature to my case?

Thanks and regards,
Nikolay.

> -----Original Message-----
> From: Scott Lawrence [mailto:[email protected]] 
> Sent: Friday, April 09, 2010 5:29 PM
> To: Nikolay Kondratyev
> Cc: [email protected]
> Subject: RE: [sipx-users] how does caller id work
> 
> On Fri, 2010-04-09 at 16:54 +0400, Nikolay Kondratyev wrote:
> > 
> > 
> > > The simplest configuration to do what you want is to set 
> a Default 
> > > Caller ID on the gateway for your trunk, and then set individual 
> > > Caller IDs for the users who have DIDs.  You should not need the 
> > > group setting.
> 
> > That was the first configuration a tried... 
> > And I just tried it again.
> > Merged xml is attached.
> 
> The problem is that your phones are using the IP address of 
> your server as the domain part of their From addresses.  
> 
> The caller alias feature looks at the From address 
> u...@domain, and the domain of the request target uri, and 
> searches the caller alias database (which you can see in 
> /var/sipxdata/sipdb/caller-alias.xml ).  First it tries to 
> find a mapping of the full identity to the target domain, and 
> if found substitutes the alias; if that fails, it then looks 
> for an entry that has no identity but matches the target 
> domain and if found inserts the alias from that entry.
> 
> Your caller-alias database maps:
> 
>     <item>
>         <identity>[email protected]</identity>
>         <domain>78.31.218.3</domain>
>         <alias>sip:[email protected]</alias>
>     </item>
> 
> which should have matched the call from 240 in your trace, 
> but didn't because the phone sent
> 
>         From: <sip:[email protected]> 
> 
> instead of
> 
>         From: <sip:[email protected]> 
> 
> but the default alias from your database:
> 
>     <item>
>         <domain>78.31.218.3</domain>
>         <alias>sip:[email protected]</alias>
>     </item>
> 
> does match because it doesn't care what the From address is, 
> and so the default for the gateway is used.
> 
> 
> 

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