[asterisk-biz] Simulating 911 ANI/ALI

[Si Tai Fan]

A simple question? If the provider send me the information over the SIP 
trunk... would I be able to see it on the manager output?

Trixter aka Bret McDanel wrote:
> 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
>
>   
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