similar to: Reverse number lookup

I am trying find a way to help the local Kerry campaign and it occurs to me that VOIP and Asterisk could be a big help. I have never worked on a presidential campaign before but I think they must be very telephone call intensive. Using VOIP to connect local, regional and national campaign offices and to keep volunteers in touch with each other would save lots of money that could be spent on

i just installed wine and want to use the "Merrian-Webster's Collegiate Dictionary". the installation of this on-line dictionary seemed okay, but when we went to launch it, it failed and reported following error: -------------------- Warning: the specified System directory L"c:\\windows\\system32" is not accessible. err:module:import_dll Library MSVCIRT.dll (which is

I need help! I installed Winebottler, and installed the first disc of a 4 disc windows program on my mac book pro. But when I went to install disc 2 when prompted, I had to quit Winebottler to put the disc in. But I need Winebottler running to do the install. How do I do it? David Murphey

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.

I've had some success setting up a crude "screen pops" using Jabber, and am pretty happy with it. But I'd love to have it pop up with the full name and address of the caller. The easiest way to do this is using ServiceObjects "GeoPhone" web service - although it's expensive for small shops (free trial, but $50+/mo after that). The interface would be pretty simple

I'm porting the most recent version of Neil Provos' systrace to FreeBSD 5.1. I'm sending him the diffs to integrate into his distribution. I'd also like to submit them to someone with FreeBSD for consideration, and hopefully inclusion as a port or whatever you prefer. Who could I send them to, or what would you prefer me to do with regard to FreeBSD? Thanks, Rich Murphey

On 08/09/2016 03:18 PM, Rowland Penny via samba wrote: > On Tue, 9 Aug 2016 16:04:04 -0300 > francis picabia via samba <samba at lists.samba.org> wrote: > >> On Tue, Aug 9, 2016 at 3:35 PM, Rowland Penny via samba < >> samba at lists.samba.org> wrote: >> >>> On Tue, 9 Aug 2016 15:21:53 -0300 >>> francis picabia via samba <samba at

On Tue, Aug 9, 2016 at 3:35 PM, Rowland Penny via samba < samba at lists.samba.org> wrote: > On Tue, 9 Aug 2016 15:21:53 -0300 > francis picabia via samba <samba at lists.samba.org> wrote: > > > I've been working with the third statement being true and exercising > > the rid option. > > Can I suggest you put it back, or add the RFC2307 attributes and use

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

On Mon, January 12, 2015 11:47, Warren Young wrote: > On Jan 10, 2015, at 7:42 PM, James B. Byrne <byrnejb at harte-lyne.ca> > wrote: >> On Fri, January 9, 2015 17:36, John R Pierce wrote: >>> >>> Enterprise to me implies large business >> >> Enterprise literally means 'undertaking?. > > Danger: We?re starting to get into dictionary flame

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb

I am looking for hints on how to estimate ratings for competitors in an ongoing pairwise competition using R... my particular area of interest being the game of Go, but the idea of identifying ratings (on a continuous scale) rather than relative rankings seems easily generalized to other competitions so I thought someone might be studying something related already. I presume the rating of a

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki. Keith