[asterisk-biz] Simulating 911 ANI/ALI

[Trixter aka Bret McDanel]

On Wed, 2008-04-02 at 21:33 +0800, Si Tai Fan wrote:
> Hi Dave
> 
> It's not that simple. All I need for now is to prove that the Asterisk
> manager interface can show the data messages that comes through the
> SIP trunk from the provider's end. Since the provider is not in the
> picture, for now I only need to somehow simulate this so that the
> customer could see it for themselves by perhaps sending those
> information from another say... asterisk to behave like a provider. As
> far as the provider is concerned, they will only sell their services
> and nothing more. Hope you can see what I mean.


>From a compatiblity standpoint its somewhat dangerous to use the same
product to test against, ie asterisk->asterisk only proves that asterisk
is compatible with itself, and not with anything else.  It doesnt even
prove that asterisk can adhere to the specification required.

As for 911 specifically there are a few standards although a few years
ago the 911 organization (NENA) did approve a VoIP based standard for
address information, although I am not sure what method they used
because well I didnt care to look into it further :)

Here are some links to get people started with regards to NENA, voip and
e911 should anyone be interested:
http://www.nena.org/pages/ContentList.asp?CTID=24
http://www.nena.org/pages/ContentList.asp?CTID=11


As for this particular provider, and their sip messages, what type of
sip message is it?  SIP Instant Messaging, or is it a header or is it
something else entirely?  I do not think at this time that the
information is available via the management interface, and it may be
better to not use that (unless its required for something else) rather
have whatever app answers the call do whatever it has to do to get that
information and then send it where it has to go (ie an operators
terminal for example).

I dont know off hand, but does asterisk even support SIP IM?  I dont
think it does, if it does its not a well documented or talked about
feature.  This may be one of the missing features of SIP in asterisk,
after all the RFC required stuff isnt all there, something optional like
this may not be there as well (I think SIP IM is optional anyway).

Nokia maintains a sip stack that does have all of this, and is rfc
compliant, and hey its even open source.  Of course this will never take
off with asterisk officially (and to add it unofficially requires you to
no longer call the product asterisk per the trademark TOS on
digium.com).  The biggest problem with using this is that it cant be
sold in the commercial versions of asterisk without providing code (its
LGPL so its otherwise compatible) and nokia is not likely to let its
employees sign a disclaimer to let digium sell it as its own.

http://opensource.nokia.com/projects/sofia-sip/index.html

-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
Belfast +44 28 9099 6461        US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!