similar to: asterisk-users Digest, Vol 66, Issue 75

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

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

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

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

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

I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a

Dear All I have captured a file in my centos showing logs captured from many modules concurrently. Please find attached a sample of the file. As you see, there are logs from individual modules that have been captured concurrently. For example, there are logs from IPTR,SNMP,HLR,TCAP,XAPP,and SCCP modules but they are coming concurrently.Each log has the header name of its accompanied module in the

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

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})

Hi there, I am stuck with the location issues. It would be easy if you have DID for each extension so that outgoing caller id would be DID of the respective extension and also physical address. Now if you are not able to get DID's for some reason. I am thinking of some situations and appreciate your thoughts. 1. If I have a PRI and map physical number (original numbers in hunt group not the

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.

Check out the proposed prices when this is approved. BELL CANADA REPORT ON THE ECONOMIC EVALUATION FOR THE TARIFF REVISION OF Bell Canada's Access Services Tariff Item 315 ? Zero-Dialed Emergency Call Routing Service (0-ECRS) *2 March 2006 TABLE OF CONTENTS Page 1.0 GENERAL 3 1.1 Purpose of the Study 3 2.0 SERVICE DESCRIPTION 3 2.1 Service Characteristics 3 2.2

I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [9 at local:1]

> Date: Sun, 15 May 2005 15:17:53 -0700 > From: "trixter http://www.0xdecafbad.com" <trixter@0xdecafbad.com> > Subject: Re: [Asterisk-Users] 911 Options > To: Ira Burton <ira.burton@gmail.com>, Asterisk Users Mailing List - > > On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote: > > I am curious if anybody has pointers on the best way to get the 7

It may have gone to sleep. Chris Cooper Systems/Network Administrator EFC International 1940 Craigshire Blvd St. Louis, MO 63146 US Phone - 314-439-4325 Fax - 314-439-4443 Mobile - 314-402-8912 - -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent:

Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ************************************** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149

OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)

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

= Reminder: TRUGhat is Saturday, 6 May 2006! The Toronto Ruby User Group''s first hackathon is this Saturday, 6 May 2006. If you are attending, please add your name to the TRUGhat wiki page. == When and Where? TRUGhat 2006:1 will be held on 6 May 2006, from 10:00 A.M. to 7:00 P.M. Tucows has generously provided the space and network availability for this event at its offices in Toronto,

If you are interested *and* live in the SF Bay area, please send your resume to samir.jain@us.fujitsu.com (not to me). Thanks! ---------------------------------------------------------------- Fujitsu Network Communications Inc. is an innovator and strategic partner with over 20 years of experience as a leading provider of wireline and wireless networking solutions that solve critical business