similar to: Source for 9-911 Labels to attach to phones?

Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? Gene Kochanowsky Solution

This one is for the archives. I got a call today that the * at one of my clients was not working. The switchboard is set up to ring for a while and then the rest of the phones start up if the switchboard doesn't pick up. This was not happening. Instead the mobile phone of one of the people there was ringing and after the delay the internals started ringing. When I connected to the web

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I

Hi list, I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitation such as much few states that a call can originate? How do

I upgraded my server from Fedora Core 4 to Fedora Core 5. I was wondering if anybody else has run into the problem and know's the fix? I recompiled asterisk and if I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File

Hi all, While pondering my choices for local dial tone service via a bunch of POTS lines or a T1, I began to wonder if perhaps there is another way. Are there VoIP dialtone providers? That is, could I use only my internet connection for voice calls and not have a separate T1/POTS bank for that? I guess I am imagining a company that gateways between the PTSN and the internet backbone.

Is nufone having problems taking gsm calls today i had some issues dialing overseas to call my folks. here's snip of what the console displayed -- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1

Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten => 911,1,ChanIsAvail(SIP/potsoutbound) exten => 911,2,Dial(SIP/potsoutbound/911) exten => 911,3,Hangup() exten => 911,102,SoftHangup(SIP/potsoutbound) exten => 911,103,Wait(1) exten => 911,104,Goto(1) Now,

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi, Is anyone else having trouble with their numbers associated with Nufone.net? Our IAX registration with them started failing about two days ago and they have been unresponsive to both emails and phone calls. If one rings the number, one gets a "number is unreachable, please try again" recording... Michael Swan Neon Software, Inc.

hi I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ? roy -- Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even

Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an

Hi: Is DynExtebDB module still working?? -- JO

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk "..the most powerful tool for manipulating and accessing CPN data.." > http://www.theregister.co.uk/2004/07/07/hackers_gut_voip/ I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time.

All - NEED HELP BADLY.. Using Asterisk with H323 Channels.. Everything is installed but we cannot register w/ Lucent gatekeeper. We ran ethereal and found that it was making GRQ (Gatekeeper discovery requests).. We had provided the name of the Gatekeeper (it's IP) and cannot determine why it's trying to do a GRQ. We want it to go straight into RRQ or ARQ and skip the GRQ.. Netmeeting does

This is not directly * related, but could be. My company is using a VoIP conferencing solution that is suffering from developer neglect. I've considered trying to leverage *, and our internal developers can build the management interfaces. If that plan is not accepted by management, I need to find a commercial solution that works. Requirements: H.323 SIP G7.11/G7.29 Multiple unique