similar to: RE: Asterisk-Users Digest, Vol 5, Issue 187

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

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 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

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

I've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server

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.

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 =>

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

I have working SIP numbers with broadvoice, and just added a DID from http://connect.voicepulse.com/ . The calls answer, but DTMF is not recognized. With "iax2 debug" active pressing DTMF does nothing. Zilch. Zero. A friend tried a different IAX2 connection, and got the same results. I see the following in the archives: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: > Hey

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

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

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

I was following docs from http://ftp.easynet.be/samba/docs/man/Samba-HOWTO-Collection/passdb.html#mysq lsam But I?m stumped I?m running FreeBSD 4.8, and need to create a mapped relationship between SOME windows users and SOME freebsd users I got Samba 3.0.8 running based on : ./configure --with-krb5=/usr/heimdal And everything works if I manually add matching Windows usernames to my etc/passwd,

I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: 1. An inbound PSTN caller calls our number 2. Asterisk answers and provides greeting 3. PSTN user dials extension of internal SIP phone 4. No ringback is heard from PSTN callers perspective 5. SIP user picks up or the

I noticed while reading some posts that people were looking for a complete NPA-NXX list for all area codes and prefixes. We happen to have the entire database. So I am making it available to the public. Help is available at: http://download.sixtel.net/npa/help.txt (Caution, 20meg files) Mysql Insert for this is available at: http://download.sixtel.net/npa/npainsert.txt CSV data for

How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm

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