But what you describe is a user privacy feature. Internal calls provide full
username/line. The callee can hit dnd, reject or ignore the call. That is
internal to the proxy. If I am called from a site and a number in their
huntgroup shows up instead of their mail pilot number, how will I get the
information from them (what is every number you might ever call me from) and
maintain it?

I'd also worry about scalability.

The "9999999" example is a simple case of the proxy checking dialplan and
permissions.

CEO's are so out of touch in some companies. I guess if we allow them to
screen all calls automatically it keeps them insulated so they can say "I
don't recall that conversation senator...". Ha! So much for an open door
policy and transparency!




============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: [email protected]

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: [email protected]
<[email protected]>
Cc: [email protected] <[email protected]>
Sent: Sat Aug 07 11:10:53 2010
Subject: Re: [sipx-users] Blocking SIP URI Calls from the innternet

I don't think any RFC changes are required at all.  Invites are challenged
all the time.

Example,

Ext 345 sends invite to sipxproxy for [email protected]

sipxproxy responds "404 not authorized"

Ext 345 resends invite with authorization header

sipxproxy checks authorized.xml and process the call.

The only difference is that sipxproxy does not require any authorization
when the destination is an internal domain.  But its a feature that could be
implemented and would not only help with the scenario of out unwanted
inbound calls not from the itsp, but also for the CEO that has a phone and
doesn't want all employees to be able to call him.

-M

>>> Tony Graziano <[email protected]> 08/07/10 10:27 AM >>>
Then you would have to invent an authorization rfc for an simple invite,
which kind of breaks the intent of sip in a way. Invites from the internet
to the proxy (port 5060) can only reach your system (AA, conferernce, media,
users), not place calls.

Itsp's require auth to send calls to them. Sipx proxy does the same for
outbound calls. Sipx does not need permissions or registration internally to
dial internal calls.

The only thing lacking in sip overall is a monitor or auto "clamp" feature
when excessive failed attempts occur (sipvicious), which also has a script
to combat it.

Email: hello I have an email for user 200. I don't have a user 200.
Sip: I have a call for user 200, ok hold on I'll ring that for you. If there
isn't a user 200, its a failed call.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: [email protected]

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: [email protected]
<[email protected]>
Cc: [email protected] <[email protected]>
Sent: Sat Aug 07 10:04:02 2010
Subject: Re: [sipx-users] Blocking SIP URI Calls from the innternet

Yeah, I knew that it would for remote workers by determine the SIPX-NONAT
status , I guess I didn't realize it would for non registered sip UA like an
inbound sip call.

Makes me wonder how hard it would be to get SipXproxy to work with ITSP that
must send to port 5060 instead of port 5080.  If the ITSP doesn't use
registration and can handle REFERS, then it would likely work even while
bypassing sipxbridge.

-M

>>> "Martin Steinmann" <[email protected]> 08/07/10 9:58 AM >>>

 <o:shapedefaults v:ext="edit" spidmax="1026" />
<![endif]-->
 <o:shapelayout v:ext="edit">
  <o:idmap v:ext="edit" data="1" />
 </o:shapelayout><![endif]-->The proxy handles NAT and anchors media.  This
is used forremote workers and happens automatically<o:p></o:p>
--martin<o:p></o:p>
<o:p></o:p>

</[email protected]>

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