They have great out going rates great if you want to just have a few extra outgoing lines… Min required is 100$ USD and you can use paypal!

everything looks wonderful except…
they only give me this information to setup the freepbx asterisk setup

exten => _123NXXNXXXXXX,1,Dial,SIP/${EXTEN:2}

Please note that will authenticate inbound calls based on IP

address or
SIP domain/realm only. will not perform user authentication
based on MD5.

I am unable to add this provider to my box

here is the email with name removed

[code:1]Dear USER,

Welcome to! Our goal is to become your preferred vendor for wholesale termination.


  • Your account has been loaded with $100.0 and is now ready to go.
  • The username you have chosen is: [email protected]
  • The password we have assigned to you is: *****. You will be asked to enter a new password the first time you sign-in to your account.

This username and password combination is the login to your Management Console. You can access the Management Console at From the console you can check the remaining balance of your account in real-time, as well as have access to rate and CDR information.

TOLL FREE TERMINATION: will pay you $0.001 for each minute of U.S. toll free traffic that you send. More information is available here


To start sending traffic, your gateways should be configured as follows:

  1. Your originating gateways should send traffic to The tech prefix assigned to you is 123.
  2. will accept both SIP and H.323 traffic and is a signalling proxy only.
  3. All interconnected gateways should be configured to use RFC2833 for DTMF relay.
  4. For H.323 Gateways, FASTSTART and H.245 tunneling must be enabled.
  5. performs routing based on the tech prefix assigned to you plus the e164 address. For example, if the tech prefix assigned to you is 123 and the destination number you want to reach is +6328120642, then the expected dialstring will be 1236328120642.


For more information, please browse to

We want you to be 100% satisfied with our product.Please send any comments or questions to [email protected].

In case of any inquiries, please include the following information in all correspondence:


Your Truly,

The Team[/code:1]

In there FAQ they have this !!

[code:1]How do I configure my Asterisk server to send traffic to
From extensions.conf (putting 01 before numbers) exten => _011NXXNXXXXXX,1,Dial,SIP/${EXTEN:2}

Please note that will authenticate inbound calls based on IP address or SIP domain/realm only.
will not perform user authentication based on MD5. [/code:1]

I tryed to set them up with sip using no name no registr string no password

– Got SIP response 480 “Temporarily Unavailable” back from

The support from the company tell me they do not support asterisk and they regurgitate the FAQ line mentioned above

I also tryed to enter using custom and it treats it like a iax2 connection… ugh!

I would really like this to work they sound like a nice company!

has anyone here any experience with IP authentication with no MD5 or anything simular to give me some pointers

Well that and you filling in the blanks on the website (your IP address)
is all that is needed.

BUT that is a joke service OK… get a real provider, the rates are not that great.

go get account with vitelity or someone who “Supports the productr they target”

I mean if they do not support asterisk (there target users) then WTF…

I tried out GRNVoip all I got to say is that it is the worse ripoff artists around.
They show one rate, yet charge you another rate. (Basic/Premium rates)

The Basic service fails to establish audio 80-90% of the time however since their service does not support call control, they end up charging up up to 30 minutes of “DEAD AIR”

Reverted to the premium service yet their is no where on their website that mentioned of a premium service price. I ended up getting screwed of about 15$ of overcharges before I finally said screw this and left.

One good note was that their premium service worked 90% of the time.

Oh and their 1-800 rebate of 0.001/min I never saw a credit for it even after 60 minutes of toll free which should have given me 0.06 credit.

I would avoid like the plague.

The early warning signs just that they make them selves very hard to reach and there response times in email defiantly not acceptable…

I will look into this vitelity company

try there rates seem to be better, and the service too. seems to be just a outbound provider I will look into them thanks

Thanks p_lindheimer

We are very sorry that you had trouble when using GRNVoIP. And we regret that we do not have detailed knowledge of the Asterisk platform or configurations. The fact of the matter is that we do not use Asterisk nor do the majority of our customers. That said, we do have many happy customers that use Asterisk, so we know that we can satisfy your needs. Accordingly, we are offering a $100 bounty in free termination credit to the first person to provide the necessary configuration (including screenshots) on this forum.

It is true that we do not support authentication by username/password. However, you will find that this is the case with all tier-1 carriers. And it is a benefit to you because MD5 authentication can increase the PDD and eat up resources on your system.

We terminate over 3 million minutes of SIP traffic daily to more than 50 carriers and we have many satisfied customers. We take great pains to respond to all tickets and requests for assistance in a timely way. And we do provide refunds with no questions asked - even if it sometimes takes a day or two to process the paperwork. If you happened to catch us on a bad day, then please accept my apology, and my promise that we will try to do better in the future.

In response to the premium rates post, Premium rates are NOT available on the website.
However, we readily provide them to anyone that asks, and they are
available from inside the management console to any customer. It is
true that sometimes calls don’t complete to a particular destination.
This can be caused by many things, such as unmatched codecs. However,
if you have trouble with termination to a particular destination our
customer support group will try to find you a route that works, all you
need to do is ask. If for any reason, we can not find a route that
works, then we will tell you that, so that you can route those calls to
another carrier.

Yours Truly,
Anthony Hicks
GRNVoIP Sales Manager

The following configuration should work fine for using this service. I tested the North American configuration, have not tested the international yet as I’m off to Phoenix to Astricon. There is nothing significantly different with this service from that of many others with the exception of their required prefix, in the case of this example, 123.

The following highlights for the SIP trunk configuration should have you up and running:

Outbound CallerID: <425NXXXXXX> (Use your CID Here)

Dial Rules


The “123+” was specified by grnvoip as a required pre-fix, these rules would be adequate for North American 11 digit numbers and International dialed in standard 011XX. format

Trunk Name: grnvoip

PEER Details:


Outbound Route:
For you outbound route in this example, you would have something like:


That is it, then just use the trunk.

Philippe Lindheimer - FreePBX Project Lead
http// - IRC #freepbx

Hi there. I have not been successful connecting to GRNVoIP. I tried with the settings p_lindheimer recommends and also the settings GRNVoIP recommends, but can’t make calls through that trunk:

PEER Details:


I left the user name blank when I used their example, because I think it’s optional. Here is what they say about it: GRNVoIP
I used the same pattern for the dial rules, replacing the 123 with the numbers I was assigned:


My outbound routes work fine with my other providers, so I just added this trunk.

Any ideas?

I am replying to a very old post on grnVoIP, but I am wondering if anyone has ever been successful at getting them to respond to an email or telephone call? I have tried to reach them for close to two weeks for more information on their toll free termination and I do not even get a canned reply.


To DeLaFe:

If you’re a current customer, you should use the Live Chat you’ll see in the left hand column after you’ve logged in. I’ve always gotten a quick response that way. If you’re not a customer, I know they’ve got a [email protected] address but I’ve never tried contacting them that way.

I also found this:

Good Luck.