similar to: Number portability

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

Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.

Hi all, I just saw this article about this new offer from Lingo.com: http://www.techweb.com/wire/story/TWB20040607S0008 $20 monthly plan with unlimited local and long-distance calling in North America (US & Canada) and Western Europe. Plus first three months free and free equipment. It doesn't say what hardware they send you. Sounds like a very good deal. I searched the list and

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? >From what I've read, the Fed's seem to say "no", they cannot refuse, or impede this. joe a.

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

Hello, i'm looking for the procedure to generate a random number between 0-1. Try rnorm() but can't get to do generate between 0-1 thanks(i'm sur it's easy) -- Nicolas Baurin Doctorant Institut de Chimie Organique et Analytique, UPRES-A 6005 Universit? d'Orl?ans, BP 6759 45067 ORLEANS Cedex 2, France Tel: (33+) 2 38 49 45 77

First I have to say I am sorry because I have not been so clear in my previous e-mails. I will try to explain clearer what it is my problem. I have the following model: lnP=Sc+Ag+Ag2+Var+R+D In this model the variable Sc is endogenous and the rest are all objective exogenous variables. I verified that Sc is endogenous through a standard Hausman test. To determine this I defined before a new

Is anyone else having really bad problems with Voicepulse over the past few days - terrible call quality, dropped calls etc.? Or is it a problem our end? Arrgggh. Makes you want to go back to POTS :-( Simon -- Simon J. Coles President & COO, Amphora Research Systems http://www.amphora-research.com/ EMail: simonc@amphora-research.com Phone: (513) 697-4764

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

I'm presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Hello Mates, I am facing a really weird issue with my mail server, somehow I cannot connect using port 993. It works with 143 but not 993 nor 995. Here is a little bit of more information: # telnet localhost 143 Trying ::1... telnet: connect to address ::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. * OK [CAPABILITY IMAP4rev1 LITERAL+

Voicepulse Connect provides callerid number, but not name. If the caller isn't already in my database I would like to do a reverse number lookup on the net to get a name. There are several web sites that allow you to do exactly that, for free. To do that from * would require that I write scripts to run the web query and get the name off the results page. Some of the sites I looked at

To whom it may concern, When dialing out an 800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This is not due to a problem on your Asterisk or your line- the bad sound effect occurs in VoicePulse. (just spend lots of time finding that out) Assaf Benharoosh MCP, MCSA, MCSE ab@franticllc.com <mailto:ab@franticllc.com> Frantic, LLC. -------------- next part

Hi, I am trying to obtain the heteroskedasticity consitent standard errors (HCSE) after apply 2SLS. I obtain 2SLS through tsls from package sem or systemfit: #### tsls #### library (sem) Reg2SLS <-tsls(LnP~Sc+Ag+Ag2+Var+R+D,~I2+Ag+Ag2+Var+R+D) summary (Reg2SLS) #### systemfit #### library (systemfit) RS <- LnP~Sc+Ag+Ag2+Var+R+D Inst <- ~I2+Ag+Ag2+Var+R+D labels

R versions 2.1.0 to present. Examples shown were computed under Windows R-devel, current SVN, but ix86 Linux shows similar behaviour (sometimes NaN or -Inf rather than Inf, depending on the compiler and optimization level used). The replacement pgamma algorithm used from R 2.1.0 has an inadequate design and no supporting documentation whatsoever. There is no reference given to support the

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

Are there other providers similiar to VoicePulse-Connect? I signed up and added a $30 credit with them, only to later find out they don't have coverage in the 757 NPA. So hopefully they will refund the $$ since otherwise I won't ever use them since my goal is incoming only, and #2 I'm still without a 757 dialup. I would actually like to find a service that will let me really have the