similar to: e911 Signalling

Ok, So... we've been looking at Intrado as a solution for national E911. They claim to be able to offer FCC compliant E911 services for VoIP companies. However, as I look into things further, they don't seem to have links to all the PSAPs for E911. Now, I understand if the PSAP is not capable of receiving E911 information, the VoIP provider is under no obligation to provide it.

During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

http://www.lightreading.com/document.asp?doc_id=73943&site=lightreading I know e911 has been discussed on ths list before, but I just read this and it got me thinking that if you have a Wholesale VoIP carrier - wouldn't they have to pass e911 on to you as a VoIP provider to, in turn, pass on to your end-users? Of course there would be a fee - just wondering if this is how the

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

How exactly does Asterisk provide E911 service??

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

We were able to solve the below problem. I'll post it in case someone encounters the same issue. No need to respond or even read unless you see a better way. Thanks - John We have manually set callerID on our outbound lines to reflect the appropriate DID both for e911 and to be polite to folks we call, e.g.: exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=5197546340) exten =>

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did

Does anyone have any experience getting inbound ANI information from a CAMA/MF/E&M Wink trunk on Asterisk? Is this only do-able with a PRI interface? Any information would be helpful.

Central business location has a PRI with a CLEC. Remote offices access the PRI for all voice traffic via VoIP. How does one get the telco to report the address of a remote office to the 911 call center when the call is made from that respective location?

Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2]

I'd like to have my Asterisk system pick up a certain extension on an existing Panasonic PBX when it rings. (It's connected to some proprietary Panasonic doorphones that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With