similar to: E911 from Remote Office via PRI

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

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

Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice

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

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

OK, that "VOIP dialtone?" thread was getting really out of hand, so I'll condense my answers into one big ugly message: 1) 911 service. Yes, that is one of three reasons to keep your PSTN line. The other two reasons are: Inbound calls from local callers still should work on a POTS line, for now. You can't find VOIP providers in most area codes, so you'll most

How exactly does Asterisk provide E911 service??

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

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

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the

I have posted this to the Asterisk Forums, but got no response yet. Sorry if you are reading this for the second time. What fax hardware do I need for a T1? Ideally, I will switch my T1 to a digital PRI (not CAS I'm told, which is not as good) coming into the building. My CLEC said I can do this switch no problem. I have an analog T1 coming in now. >From the Asterisk box, I will connect

Hello everyone, We have recently turned up a new T1 from McLeod (Midwestern CLEC). It is configured like so: /etc/zaptel.conf: loadzone=us defaultzone=us span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs) bchan=13-23 nethdlc=1-12 dchan=24 /etc/zapata.conf: switchtype=national context=pri-in signalling=pri_cpe group=1 channel => 13-23 I can get hdlc0 (and pvc0) up just fine after the

I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest

Having just returned from four days at the VON show in Santa Clara, I thought I would submit a "highlights" message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. [VoIP IS BIG] First, I have to say that VoIP is BIG. It is the buzz technology of the day. The show was packed, and everybody there was there

OK.. I'm up to my eyes in LD BS! I can't for the life of me understand how any carrier, either VoIP or traditional service provider can make heads or tails of how to hand off an * based call to an LD provider. Every provider I talk to, says I have to have a traditional T1 put in to their respective networks. I don't want to do this. I want a LD provider that can take a IP, SIP, IAX

Hi, I'm trying to set up a remote office with its own Asterisk Server they'll have a dedicated land line, but we'll still want them connected to the main office via VOIP (IAX2 via VPN). I've tested using IAX2 to bridge between the offices based on extensions, since the extensions we want to share are in isolated blocks of numbers. I'm not sure how to handle voicemail though.