[Asterisk-biz] E911 VoIP Solution - Dash911

[Michael Giagnocavo]

>-----Original Message-----
>From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz->bounces at lists.digium.com] On Behalf Of Rusty
Shackleford
>Sent: Friday, June 03, 2005 10:06 PM
>To: 'Commercial and Business-Oriented Asterisk Discussion'
>Subject: RE: [Asterisk-biz] E911 VoIP Solution - Dash911
>
>Please clarify...

My pleasure.

>> 	When a call comes in,
>
>"...Comes in...", WHERE?

When a subscriber dials 911 on the CPE and then goes to your softswitch.

>> a simple HTTP request is sent to 
>> our servers,
>
>Sent by...?

The softswitch (possibly Asterisk)

>> and the reply will contain the 10-digit number + 
>> caller ID to set. 
>
>WHAT "10-digit number"?
>
>What do you mean by " + caller ID to set"?

We will reply with a 10 digit PSTN number for the softswitch to dial to
complete the 911 call, including the Caller ID to set when dialing that
number. The caller ID acts as a key. 

>> From there, you simply use your current 
>> PSTN connection.
>
>Simply use it... how? In what manner? I don't HAVE a PSTN connection. I
>use VOIP. Now what?

If you (VoIP Service Provider) do not have a PSTN connection, then you
aren't an Interconnected VoIP provider. If you can't place or receive calls
to the PSTN, no, 911 service will not work via our system (nor is it
required). 

If "don't have a PSTN connection" meaning you don't have a hardware
telephony card, that's unimportant. So long as your softswitch can get to
the PSTN (via SIP, H.323, IAX2, whatever), place a call, and set caller ID,
things will work.

In the future, we'll offer service to customers who cannot set caller ID
(possibly at an additional fee).

>> We chose this path because it has "less 
>> moving parts", and is far easier for our customers to 
>> implement than having to SIP to different endpoints. As well, 
>
>If you say so. 

Well, assuming that a VSP has figured out how to connect to the PSTN (a
prerequisite) allowing them to simply dial a PSTN number like any other call
instead of having to negotiate SIP + RTP could be seen as easier. 

>Pretend that we don't have the slightest idea of the topology that you
>are envsioning your service operating over, and explain it again,
>please.

Sure, I'll even get some diagrams up on the website in the next few days and
reply here.

Thanks for your interest!

-Michael