similar to: Re: 911 and lawsuits and redundancy

Hi, Most companies we work with, have 'designated' crisis management teams. These vary from the insignificant crisis', through to life-threatening crisis'. There is always an assigned emergency services contact, whose job it is in an emergency, to maintain communication with the emergency services. One of our corporate functions is crisis management - so we have to consider

I have also noticed that sip.conf doesnt get updated without a restart..... was thinking I am doing something wrong, but maybe not now...... Chris -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, 8 January 2004 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Another concern I have on this

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of

Hardly finished building our phone system for our school district and I have an opportunity to sell and install a system for a local small business. We are competing against a bid for an integrated voicemail/switch that runs about $1300 (without phones and cabling) and will work with analog phones. Is there hardware configuration (either using analog or IP phones) that would meet these needs and

FYI there is a way to do 911 its called E-911 enhanced 911 the user has to set it up with the local emergency services to it and you setup your pbx to xmit the data. Here is the fcc rule about it http://www.fcc.gov/911/enhanced/

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten => s,1,Set(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,Set(EMERGENCY=1,g) exten => s,n,Set(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})

After over a month (well, ok, no more than an hour a day :) of planning, getting hardware, tinkering and testing, I'm about to my Ultimate Home Phone System (tm) online. Connectivity to the outside world is provided by: A. 1 POTS phone line connected through an X100P ($11/month, needed to carry DSL) B. 1 Vonage ATA186 connected through an X100P (needed for the rate center :( ) C. 3 Broadvoice

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

May be slightly off topic, but I was wondering what everyone thinks of this latest ruling against Vonage? Does anyone really know what Verizon hold patents for, and could those patents possible affect anything in Asterisk? Who knows who Verizon will go after next. Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

> Tilghman Lesher wrote: >> On Sunday 18 January 2004 12:01, Adthrawn wrote: >> >>> Spell out RAID - Redundant Array of Inexpensive Disks. Bingo. >> >> >> That's "Independent Disks". It's the independence of each spindle >> that >> is valued, not the cost proposition. If one spindle goes, it's not >> all >>

NENA i2 The NENA i2 architecture was designed to support the interconnection of Voice over Internet Protocol (VoIP) domains with the existing Emergency Services Network infrastructure. This overview will describe the functional elements and call flow of a VoIP 9-1-1 call over the i2 architecture. The NENA i2 architecture was also designed to utilize existing 9-1-1 voice and data links to all

This was given to me by a Telco guy in Canada. Talk about a great view of things to come. ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery in Canada. PREPARE FOR 911 Executive Summary Emergency Services Working Group (ESWG) recommends on a consensus basis the Commission order the deployment of NENA Interim-2 (i2) compliant

I understand that most VoIP providers allow for 911 calling but that 911 service is not the same as that available to PSTN.

Ok here is one for you. I know we all do the this for 911: exten => _911,1,Dial(Zap/1/911) exten => _9911,1,Dial(Zap/1/911) And this probably is more then acceptable for most of us. However I have a system setup that uses SIP for most calls and 1 POTS line. We use a "least cost" routing that uses the POTS line for local calls AND SIP when appropiate. What I want to do is

ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery in Canada Executive Summary Emergency Services Working Group (ESWG) recommends on a consensus basis the Commission order the deployment of NENA Interim-2 (i2) compliant emergency services components, systems and upgrades to result in the operation within 18 months of enhanced

ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating Impediments to 9-1-1/E9-1-1 Service Delivery in Canada DRAFT Executive Summary Emergency Services Working Group (ESWG) recommends on a consensus basis the Commission order the deployment of NENA Internet-2 (i2) compliant emergency services components, systems and upgrades to result in the operation within 18 months of

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's