Re: [asterisk-biz] Simulating 911 ANI/ALI

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 proof of concept 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?

asterisk_help@iwishi.nu wrote:

... 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.     

  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:  http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SIPcommands  http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header Synopsis - Gets the specified SIP header  http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPGetHeader 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  _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com--  asterisk-biz mailing list To UNSUBSCRIBE or update options visit:    http://lists.digium.com/mailman/listinfo/asterisk-biz