similar to: Calling Card question

How do you set the expiration date in ASTCC? DO you have to customize the CGI script? A maintenance fee field would be nice as well. Anybody? -- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Whoa, talk about flying under the radar. I got a few DIDs *months* ago from Sixtel (or iax.cc). Initial respone was great, but then it seemed that the only tech-support person had fled the country. No responses, bad responses, poor call-quality. I had a few $s left in the balance and kind-a just forgot about them. Not worth 10 minutes of my time to get $10 of my money back, quite frankly.

Hello, I have some problem using ASTCC application. I've installed the application and everything works well. I've created card numbers, routes trunk and others. When I dial the desired number (77) in my case, I'm prompted to enter my card number. All goes well till I'm prompted to enter the destination number. When I enter a destination number, the system says it's not a

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Just setting up Asterisk. I'd like to be able to dial out through VOIP providers and have customers type in a code in response to a prompt. So far, I've been able to set things up to make the call and play the prompt. However, my problem now is the DTMF tones; they don't register when I call get_data. When I make a call in person (from the DIAX softphone, through a VOIP provider,

Dear List members- I am trying to configure ASTCC (Asterisk calling card application) but having a hard time to configure it properly. My project deadline is approaching and couldn't figure out how to make ASTCC functional. Here are some details what I have done so far. 1) I have installed ASTCC successfully. 2) I can access astcc-admin.cgi script without any problem. 3) I have created

Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person

Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

hello list, I subscribe to Sixtel.net for a DID just to see how it worked. they say DID active inmediatly , but after 4 days , I have no DID , I tried to call tech support office and it seems they all ways are in hollidays, a support ticket no good luck, I call back service also not good luck. so anyone here has experience with them? are they a SCAM? fortunally only put there $10 bucks but

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

Hi, I've tried to make toll-free DID work for the last 2-3 weeks. Apparently only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody else just gets a busy signal. It takes for them about 5-6 business days to respond to my request. It seems they are looking in the once a week to the High priority tickets. It also seems that they are permanently closed ! No matter when I

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

I tried google and sixtel.com - couldn't find a web page for Sixtel - would some kind soul point me in the proper direction? ============================================================================ ====== Date: Mon, 13 Dec 2004 16:51:47 -0800 (PST) From: Steve Edwards <asterisk.org@sedwards.com> Subject: Re: [Asterisk-Users] IAX.cc / Sixtel? To: Asterisk Users Mailing List -

How can I fulfill that? *Billing Increments* Continental USA: six (6) second increments. International: thirty (30) seconds minimum and six (6) seconds thereafter. Mexico: sixty (60) seconds minimum and six (6) seconds thereafter. bye Ronald Wiplinger

Hi, How is iax.cc / sixTel to be configured as a termination provider in asterisk@home? The iax.cc / sixTel instructions tell you to do this: iax.conf: ------------ [sixTel] type = friend host = iax2.sixtel.net context = inbound secret = mypassword allow = all extensions.conf: -------------------------------- exten =>

Hello This might not be a 'pure' * question, but it is relevant to general VOIP technology. I tried x-lite on my notebook with wireless connection(802.11). The software has been tested with the fixed line connection. It worked fine to call through *. When using wireless connection, it is clear on my side using notebook; however, there is loud noise on the other side of the call which uses

On by eth1 does nothing, I tried configuring Shorewall and just ignoring it, but Shorewall errors with "Unable to determine the routes through interface ''eth1''", searching high low seems to reveal I am either the first person ever to see this error, or it is so trivially overcome that it is not documented or commented upon anywhere on the internet. I tried all sorts