Your originating gateways should send traffic to 70.42.72.49
70.42.72.49 will accept both SIP and H.323 traffic and is a signaling proxy only.
We use the standard ports - 5060 for SIP and 1720 for H.323.
All interconnected gateways should be configured to use RFC2833 for DTMF relay.
For H.323 Gateways, FASTSTART must be enabled.
grnVoIP.com performs routing based on e164 addresses.
Therefore, please send the destination number in the format of COUNTRY CODE+CITY(or AREA) CODE+NUMBER.
1. Who can use grnVoIP?
2. Is there a minimum monthly commitment?
3. How do I start using grnVoIP?
4. Where can I find your rates?
5. Can I cancel at any time?
6. What payment methods do you accept?
7. Is there a minimum deposit requirement?
8. What payment methods do you accept?
9. Is there a minimum deposit requirement?
10. Does my balance expire over time?
11. What is the initial cost? Is there a monthly service charge?
12. Do I get a paper bill?
13. Do you provide configuration guidelines for Asterisk?
14. How many concurrent calls does grnVoIP support?
15. What is the difference between "Standard", "Premium" and "1/1" routing?
16. Does grnVoIP support 1/1 billing increments?
17. What codecs does grnVoIP support?
18. Does grnVoIP provide telephone support?
19. Can I ping a grnVoIP server to test for network latency?
Who can use grnVoIP?
Any enterprise or ITSP that wants to buy quality VoIP termination services to all worldwide destinations. We do require you to have a static, public IP address. As a result, you will not be able to use our service from a consumer premise device such as a softphone or ATA.
Is there a minimum monthly commitment?
No, there are no commitments whatsoever. Simply prepay a minimum of $50.00 and use it at your convenience.
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How do I start using grnVoIP?
Click on the Sign Up menu item on the left side of this page, or follow this link http://www.grnvoip.com/terms_accept.htm
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Where can I find your rates?
Click on Rates on the left-side of this page. You will find all of our Standard and Premium rates as well as future planned changes.
We are dedicated to providing you high quality A-Z termination at competitive prices. We will be glad to discuss deeper discounts for significant traffic after we see some termination history.
Can I cancel at any time?
Yes you can with our complete money back guarantee:
If you are not satisfied at any time for any reason,
we will refund all money in your account.
No questions asked!
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What payment methods do you accept?
We accept online credit card payments and Paypal.
Is there a minimum deposit requirement?
The minimum amount you can start with is $50.00. After you are a customer, you may reload your prepaid account with any amount greater then $15.00.
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Does my balance expire over time?
No, there is no expiration of your balance.
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What is the initial cost? Is there a monthly service charge?
There is no initial charge, no monthly service charge, or any other hidden costs.
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Do I get a paper bill?
No, we do not provide paper statements.
You can view your account balance, call history
and payment history online.
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Do you provide configuration guidelines for Asterisk?
The configuration below has been reported to work with our service. Just be sure to replace with the sample tech prefix below - 11332403 - with the actual tech prefix assigned to you when you signed-up.
in sip.conf
[grnvoip] allow=all ---- [depending on the codec you want to send to us.
we support G729, G723 and g711 codec variants]
canreinvite=no
context=from-trunk
disallow=all
host=70.42.72.104
insecure=very
type=peer
in extensions.conf
[from-trunk]
exten => _1XXXXXXXXXX,1,Dial(SIP/70.42.72.49/11332400${EXTEN}) This
example will route U.S. calls to grnVoIP and prepend 11332403, the
Standard routing prefix, to the SIP INVITE sent.
eg:
1. customer will dial 12127773456
2. asterisk will send to us as (sip:1133240312127773456@70.42.72.104)
3. If you would like for all destinations to be sent to grnVoIP, then
use this: exten => _Z.,1,Dial(SIP/70.42.72.104/11332403${EXTEN})
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How many concurrent calls does grnVoIP support?
There is no set limitation on the number of concurrent calls. Your access to the system will depend on usage by other customers, which varies from time to time. However, we monitor traffic closely, and if it appears that your traffic may exceed the available capacity, we will contact you and ask you to utilize dedicated facilities.
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What is the difference between "Standard", "Premium" and "1/1" routing?
Our Standard routes have been tuned to prioritize the best possible rates with ASR and ACD that are within acceptable parameters. Our Premium routes have been tuned to prioritize the best available ASR and ACD without regard to the rates. 1/1 routes use the same rates as our Standard routes, but include fewer underlying carriers, resulting to lower ASR.
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Does grnVoIP support 1/1 billing increments?
At your request, grnVoIP will provde an additional trunk group option with 1/1 billing increments. The difference between this routing and our Standard routes is in the depth of underlying carriers available. Whereas our the Standard service (offering 30/6 billing increments) has many carriers available for each route, the 1/1 option uses only a subset of those carriers.
We will assign you a different prefix for the 1/1 trunk group. Calls sent using the 1/1 prefix will be charged using 1/1 increments. Calls sent using the 30/6 prefix, will be charged using 30/6 increments.
In order to get the best of both worlds, we suggest that you configure your switch to try the 1/1 prefix initially and then failover to the 30/6 prefix in the event that the carrier responds with an error. As with all our routes, we will return an appropriate code to allow your switch to forward route to the next route choice.
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What codecs does grnVoIP support?
Our carrier selection has been optimized for use with the g.729 codec. If you set this as your first priority codec, you will maximize your chances for completing calls.
That said, we use many underlying carriers for termination, and each carrier has different capabilities. Many support g.711, but not all.
The same goes for g.723.1 and iLBC. As a result, if you set any of these codecs as your preferre codec, a call may or may not complete using it, depending on which of our underlying carriers receives the call. Our softswitch will allow your endpoint and the carriers to negotiate the mutually preferred codec.
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Does grnVoIP provide telephone support?
grnVoIP offers retail quality termination at wholesale rates with no monthly fees or commitments. We are able to provide this outstanding value because we work hard to keep our overhead low. As a result, we do not typically make support available via telephone.
However, 24 x 7 support is available through email and Live Chat. Further, we respond to all service-affecting issues within one hour. Other issues are handled within 24 hours. We believe that our approach affords you, our customer, the most flexible and convenient way to access top tier wholesale termination.
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Can I ping a grnVoIP server to test for network latency?
Although a ping test is not necessarily indicative of actual network conditions (since intervening networks may block or delay ping requests) you can give it a try by pinging 70.42.72.104
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grnVoIP strives to make service as worry free as possible. However, when
trouble occurs, you can access our 24x7 tech support staff by opening a
trouble ticket. It's as simple as sending an email to . Just be sure to include as many details about your
problem as possible. To help us to provide answers to you quickly and
efficiently, please include the following information:
- Your Brand/Company Name and ID
- Your Contact Information
- If the problem involves termination, include your gateway's IP address
and the relevant CDR ID, or if that is not available, the number dialed
and approximate time (be sure to indicate the timezone)
- If the problem involves a grnVoIP Wholesale Console, please indicate the URL you are using to access
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