similar to: Re: 911 and lawsuits and redundancy

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

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

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

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 just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? c. where can I find help for configuring it? d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd.

I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total

I am trying to find a manual for the Carrier Access Adit 600. Does anyone know where I might be able to find one? Thanks -Jon -- Jon J. Brandon jon@monsoonretail.com http://www.monsoonretail.com

All, What i'm trying to do is setup a windows DUN connection via my asterisk box and over PSTN or VOIP to my work. What I hoped i'd find was a vitual modem driver for windows 2000 that wouldtalk over sip to my asterisk box and then act like a normal modem so I can dial out from that to our RAS service at work. Any One got any ideas as VPN is not an option for security reasons the DUN

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/3efdeb9f/attachment.htm -------------- next part -------------- Hello Though my SNOM 200 phone receive a voice mail but it doesn't show MWI on the LCD panel, Instead it keeps displaying "DND SW-REG,Call-log". Though I can access my voicemail using exten

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

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

Hi.. I've got an asterisk setup with an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5 seconds before asterisk realizes the line has been hung up and drops the call.. this causes my SIP phone to continue ringing, and occassional phantom voice mail messages to be left.. I'm located in good old standard North America, with a regular Verizon residential

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

OK, I asked this about a week back and met with no repsonse at all. But perhaps its worth trying again. Does anyone on-list have * running BRI to their local telco? I'm considering this as an alternative to my TDM400p card. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

I recently got a Pulver Innovations WiSIP wireless SIP phone to determine if we want to use them in our organization. Since the WiSIP phone arrived last week, I have had nothing but headaches. I do think I now have the problem narrowed down. I have spent a bulk of my time trying to get the WiSIP to work with a couple of Linksys WAP11 Version 2.2. I have met with no success in getting the WiSIP

I just purchased 30 of these after testing one for a few months and would like to quickly purchase another 40. We really like these phones: good sound quality, good echo control (no echo in speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys. Unfortunately we missed on big problem with Call waiting in our testing. When using asterisk 0.9.1 or rc2 the phone will reboot

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be