VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
sa at hktelecoms.com
Guest
|
 Posted: Fri Mar 28, 2008 11:37 pm Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
 |
|
Hi
Can anyone suggest how I can simulate a 911 call with a ALI (Automatic Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive this information?
Thanks in advance...
Si |
|
| Back to top |
|
 |
asterisk_help at iwish...
Guest
|
| Posted: Sat Mar 29, 2008 6:19 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
On Sat, 29 Mar 2008, Si Tai Fan wrote:
| Quote: | Can anyone suggest how I can simulate a 911 call with a ALI (Automatic
Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive
this information?
|
OK, first... I think of asterisk as "the telephoney toolkit" rather than a
PBX. The answer to "can asterisk do that" is always yes. But somethings
are easier to do. Recall you have source code so you can do anything.
ALI != Automatic NUMBERING Information
What you are asking about is huge! I think standards are still emerging
so you really need to define exactly what you want.
Try reviewing these publications first:
http://www.its.dot.gov/ng911/ng911_pubs.htm
Esp look over: NG9-1-1 System Description and Requirements Document
Then use google and the following terms: "LoST sip 911 i3 ecrit"
-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
!DSPAM:47edc8f1273931640078321!
_______________________________________________
--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 |
|
| Back to top |
|
|
sa at hktelecoms.com
Guest
|
| Posted: Mon Mar 31, 2008 9:21 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
Thanks for your response.
I have a project... that why. 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.
Since I have to prove that asterisk can do the job, I need to show that it will work by simulating it. Any suggestions?
asterisk_help@iwishi.nu (asterisk_help@iwishi.nu) wrote: | Quote: |
On Sat, 29 Mar 2008, Si Tai Fan wrote:
| Quote: | Can anyone suggest how I can simulate a 911 call with a ALI (Automatic Numbering Information) over SIP to Asterisk? Will Asterisk be able to receive this information?
|
OK, first... I think of asterisk as "the telephoney toolkit" rather than a PBX. The answer to "can asterisk do that" is always yes. But somethings are easier to do. Recall you have source code so you can do anything.
ALI != Automatic NUMBERING Information
What you are asking about is huge! I think standards are still emerging so you really need to define exactly what you want.
Try reviewing these publications first: http://www.its.dot.gov/ng911/ng911_pubs.htm
Esp look over: NG9-1-1 System Description and Requirements Document
Then use google and the following terms: "LoST sip 911 i3 ecrit"
-Eric Osterberg
-Sound Choice Communications LLC
Minnesota, US
|
|
|
| Back to top |
|
|
asterisk_help at iwish...
Guest
|
| Posted: Mon Mar 31, 2008 10:26 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
| Quote: | ... 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 |
|
| Back to top |
|
|
sa at hktelecoms.com
Guest
|
| Posted: Mon Mar 31, 2008 11:26 pm Post subject: [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 (asterisk_help@iwishi.nu) wrote: | Quote: | | Quote: | | Quote: | ... 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
|
|
|
|
| Back to top |
|
|
david.cantera at iacne...
Guest
|
| Posted: Wed Apr 02, 2008 12:33 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
Si,
if you can't make double the cost+ of the development on the application, drop it or pass it off to the provider and take a commission.
you have to make the investment with the provider (or some deal where he/she will do the prototype for free based on you and him getting the final deal) to get the prototype done since you can not do it yourself. in any case, you will, most likely, have to put the provider and the customer together for the project to be successful.
get a non-compete, non-circumvent, or agency contract between and with both you and the customer and the provider. meaning a contract between; a) you and the customer, and b) you and the provider....
specifically, that the customer is yours and the provider can not circumvent you on this or future deals.
daveC
Si Tai Fan wrote: | Quote: | 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 (asterisk_help@iwishi.nu) wrote: | Quote: | | Quote: | | Quote: | ... 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
|
_______________________________________________
--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
No virus found in this incoming message.
Checked by AVG.
Version: 7.5.519 / Virus Database: 269.22.2/1353 - Release Date: 03/31/2008 06:21 PM
|
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
|
|
|
| Back to top |
|
|
sa at hktelecoms.com
Guest
|
| Posted: Wed Apr 02, 2008 8:42 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
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.
Si
| dave cantera | wrote: | Quote: | Si,
if you can't make double the cost+ of the development on the application, drop it or pass it off to the provider and take a commission.
you have to make the investment with the provider (or some deal where he/she will do the prototype for free based on you and him getting the final deal) to get the prototype done since you can not do it yourself. in any case, you will, most likely, have to put the provider and the customer together for the project to be successful.
get a non-compete, non-circumvent, or agency contract between and with both you and the customer and the provider. meaning a contract between; a) you and the customer, and b) you and the provider....
specifically, that the customer is yours and the provider can not circumvent you on this or future deals.
daveC
Si Tai Fan wrote: | Quote: | 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 (asterisk_help@iwishi.nu) wrote: | Quote: | | Quote: | | Quote: | ... 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
|
_______________________________________________
--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
No virus found in this incoming message.
Checked by AVG.
Version: 7.5.519 / Virus Database: 269.22.2/1353 - Release Date: 03/31/2008 06:21 PM
|
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
_______________________________________________
--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 |
|
|
|
| Back to top |
|
|
trixter at 0xdecafbad.com
Guest
|
| Posted: Wed Apr 02, 2008 9:48 am Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
On Wed, 2008-04-02 at 21:33 +0800, Si Tai Fan wrote:
| Quote: | 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.
|
| Quote: | 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!
_______________________________________________
--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 |
|
| Back to top |
|
|
sa at hktelecoms.com
Guest
|
| Posted: Tue Apr 08, 2008 10:11 pm Post subject: [asterisk-biz] Simulating 911 ANI/ALI |
|
|
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: | Quote: | | Quote: | On Wed, 2008-04-02 at 21:33 +0800, Si Tai Fan wrote:
| Quote: | 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.
|
| Quote: | 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
|
|
|
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
