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Actually I am biding for the project and I am in between the provider
and the customer. The customer wants me to do a demonstration first as
a <b>proof of concept</b> but the data will be subject to the final
confirmation by the provider. Until then I won't be able to talk to the
provider directly as it is masked by the customer. Any suggestions?<br>
<br>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk_help@iwishi.nu">asterisk_help@iwishi.nu</a> wrote:
<blockquote cite="mid:Pine.LNX.4.64.0803310957540.8313@omni.sccvoip.com"
type="cite">
<blockquote type="cite">
<pre wrap="">... I plan to use Asterisk as the front end to
connect to a provider who will connect via SIP trunk and pass all 911 calling
informations like...
1. ANI (Automatic Numbering Information)
2. ALI (Automatic Location Information)
a. Caller no
b. Building name / caller name
c. Address
d. Latitude and Longitude of the caller address
3. Incident Information
a. Incident code
b. Incident Description.
c. might have other information as well.
Then I wish to pass these through the manager interface where it can be
collected and processed into a database server to display it on a console...
perhaps like a crm pop-up.
</pre>
</blockquote>
<pre wrap=""><!---->
You will need to contact the provider that will send these details via SIP
and ask of the standard they will be following. I'm not aware of any
single standard that will address the information you are expecting.
You might want to review:
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SIPcommands">http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SIPcommands</a>
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header">http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header</a>
Synopsis - Gets the specified SIP header
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPGetHeader">http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPGetHeader</a>
With this app, you can pick any header from an incoming invite and
stuff it into a channel variable. It is a generic way of supporting any
header a vendor or service provider may add that you want to use in your
dialplan.
In the US, the PSAP (Public Safety Answering Provider/Point) is given the
ANI (an identification number, normally a billing phone number) with the
telephone call and they must then use a seperate communications circuit
connecting them to a database provider to query for the information needed
to dispatch the call.
Please let me know what standard or spec they are using in their SIP
calls. As a CLEC and VoIP service provider myself, I'm always interested
in learning of new developments in this area.
-Eric Osterberg
Sound Choice Communications LLC
Minnesota, US
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</pre>
</blockquote>
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