similar to: OT: How to "own" a telephone number?

Hello! I am in the middle of having a vanity toll-free DID set up. It's been 13 days now (9 business days). This is the first time I'm doing this, and I'm not sure of the process. There has been a very weird progression of changes on my number, from fast-busy, to a message saying that I'm calling from a phone with restrictions (no matter *what* line I call from), to a

Hello! Several of my customers would like to add a backup to their Internet connection. ISDN is a good solution: decently fast for a dial-up-type connection, yet still faily affordable. While I was at it, I decided to look at a couple of more creative telephone service options to possibly improve their service or lower costs at the same time. These customers range from having just a

--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson <asteriskuser@aktzero.com> wrote: > What you are seeing with these bargain providers is they have a clause in > their contract that says they own the number, not you. It is a lock, and > it ought to be illegal, but sadly, it's probably not. If you choose one > of these companies that doesn't allow you to

Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have

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

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

I have an opportunity to pick up a couple of NetFinity 5500's 4 way Xeon 550's w/ 2 gig RAM for very little $$$ I have seen this: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg00719.html In it, there is a passing remark to the Digium cards having problems with NetFinity's. Can anyone here comment on whether this is still an issue with * 1.0? It'd be a bummer if

e164.org is a public name service which provides ENUM.164, a method devised by the IETF and ITU to allow an ordinary telephone to be connected to an Internet type network and provided dialling service from other, regular telephones. Unlike many other "free" voice over IP systems, e164.org allows users who have a regular telephone line, to also hook themselves up to the Internet

I'm thinking of starting a partyline, where people call in and talk to other people. For record keeping and billing purposes, I'd like to go by the callers telephone number. This method works fine as long as the caller doesn't have callerid blocked, but what are my options if they do block their number? I know there must be a way to report it, because there is a service provider here

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to

Just curious... Is there anyone out there who has given this outfit money and actually received any service from them?

Hello! I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me

I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

All, Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's

I am brand new to Asterisk. My question is if I want to have multiple participants all listening, or listening and talking, do I need to have a separate telephone line for each, or can they all dial in using a single telephone number and a single line? Thanks, John Fistere -------------- next part -------------- An HTML attachment was scrubbed... URL:

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

Hello! I have finally ordered some Asterisk hardware: the TDM DevKit. However, I want to use VoIP phones (or possibly adapters) for remote users. I would like to get some suggestions on which phones to buy. I'm hoping that some of you with real experience might be able to help me out! Here are the features that are important to me: * While these phones are initially going to be

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia