Frequently Asked Questions

General / Billing

We are a carrier-grade wholesale VoIP provider offering high-quality SIP trunking and DIDs (phone numbers) at the most competitive rates. Our service is very flexible and scales down from the VoIP hobbyist up to the largest call centers.

No. There's no volume commitment and you can spend your credits at the pace you want. Your balance never expires.

The minimum deposit is $50 (USD).

We accept PayPal, Credit Card, ACH, Wire Transfers and Western Union.

Yes. Several of our customers are call centers and marketing companies. We are proud to be the #1 retail call center VoIP provider in the country.

We do not charge absolutely anything other than what you actually use. You get full credit what you pay, we bear credit card processing fees.

Yes, we have DID numbers available for USA and Canada with instant activation. We also have USA/Canada toll free numbers. We have international DIDs available in 30 countries.

Yes, we allow ports both ways. Port orders into Flowroute are $25 for single-number orders. Port orders with more than 10 numbers are $15 for each number. Once logged in, customers can submit their port requests using our automated LNP submission form.

Yes, we support calls to all toll free destinations in the US and Canada. Unlike some other carriers, calls to toll free numbers over our network are absolutely free of cost.

We support all SIP compatible equipment including Cisco, Avaya, Grandstream, Polycom etc. We also support most VoIP software including Asterisk, Trixbox, Elastix, Vicidial, etc. And soft phones like X-lite, Eyebeam, Twinkle, etc.

You can view completed call records, associated costs for calls and purchases, and remaining balance in real time once you've logged into your account. You can also setup automatic email and SMS notifications to be sent our from our servers when your balance goes below your custom threshold.

Yes, you can access your Call Details Report online in real time. You can also download your CDR in MS Office friendly CSV format.

USA and Canada 12 seconds initial, 6 seconds increment Mexico 60 seconds initial, 60 seconds increment World 6 seconds initial, 6 seconds increment

It's the way we calculate our rates in order to bill your calls. For example, if you call USA for 10 seconds, you will be charged for 12 seconds. If your call is 15 seconds long, you will be charged to 18 seconds. We do not bill you for the whole minute.

No, we take our customers' privacy very seriously and do not release customer information unless we are legally required to do so. Please see our Privacy Policy for more details.

The main office is located in Sunnyvale, California. Credit Card charges are made in US Dollars (USD) by Rapid Eagle Inc via our secure USAePay merchant gateway.

Technical

We primarily support SIP. We can support IAX2 on special request.

We support G.711-ulaw and G.729. Most free or open-source PBXs are not packaged with the G.729 codec due to licensing issues. We recommend that you install it for more efficient bandwidth usage. However, if bandwidth is not a concern we always recommend using G.711-ulaw for best quality.

For best sound quality, if you have the bandwidth available, we recommend G.711u. However, you can still maintain an excellent voice quality and lower bandwidth usage with codecs like G.729. You can use this link to calculate bandwidth requirements www.bandcalc.com

We support the universally accepted E.164 format for all outbound calls. For example, to dial a US/Canada number you would call 1 followed by the 10 digit number. For UK, dial 44 followed by the number.

We support 100% caller id pass through for all our domestic as well as international routes.

We transmit caller ID based on the presence of one of the following header fields: "P-Asserted-Identity" or "Remote-Party-ID". If your device is incapable of sending these headers, you can set the global caller ID for all calls placed from your account on the Outbound Setup page.

For most destinations we can handle burst traffic in excess of 1000 simultaneous sessions. Simultaneous sessions may be restricted for accounts with low remaining balances. If your channels usage is expected to be over 100 simultaneous calls, please contact Support so we can reserve channels for you and make arrangements for future traffic requirements.

Yes we do provide termination in every country. We are an A-Z VoIP termination provider. We always do our best to find and keep quality working routes. If you were to experience some problems with a particular destination, let us know and we'll make everything possible to fix the problem.

Yes, here are some measures we use to protect your accounts:

1. We block international calls on all accounts by default. You can enable specific countries on your account by contacting Support.
2. The SIP authentication credentials for your account are 10-digit random numbers for high level of security.
3. We have intelligent firewalls on our servers which blocks IP addresses based on multiple failed registration attempts.
4. We optionally support and recommend IP authentication as opposed to the default registration based authentication. IP authentication protects your account by allowing calls only from your static IP address.
5. Our website has SSL support which ensures that your authentication and credit card details are sent over secure encrypted channels over the Internet when you use our website.

We also advise that you follow some general security practices in order to minimize fraud on your account:

- Ensure your password for your VoIP Essential account is very secure
- Use non-standard ports for system services
- Restrict web access to your PBX/VoIP system
- Protect your PBX/VoIP system with software-based firewalls such as IPTables
- Protect your network with a hardware-based firewall
- Use strong passwords for all phone extensions on your PBX
- Implement a VPN for your PBX/VoIP system
- Review access logs on a regular basis
- Keep up to date on security patches and practices for your network services

VoIP fraud patterns are constantly evolving today and we are doing our part to develop increasingly sophisticated detection and prevention measures. The best line of defense is always the security of your own systems.

We do support calls to emergency and informational services however these features are only available on our hosted PBX and SIP trunking products. Contact Us for more information.

Most accounts on VoIP Essential require a $5 minimum balance. This is to prevent accounts from going in the negative when you have several concurrent calls. Please add funds to your account to resume normal operation or contact Support if you need more information.

Our VoIP servers consist mainly of OpenSIPS and Asterisk running on the Ubuntu Server x64 operating system.

Support

Our official support hours are Monday through Friday, 9am to 7pm EST (Eastern Standard Time). However, we do have staff attending your email requests beyond these support hours on weekdays as well as weekends.

You can reach us on +1.855.455.VOIP (+1.855.455.8647) from Monday-Friday, 9am to 7pm EST. Beyond these hours, you can get in touch sending an email to support@voipessential.com.

Currently, we have an English speaking staff but we can arrange for professional translators in case you are an international wholesale customer.

We do have configuration samples for Asterisk, FreePBX/Trixbox/Elastix, Vicidial/Goautodial. Click here to view these samples in our Wiki. We'll keep adding more configuration examples in the future. Configuration to use our service is very straightforward with most softwares/devices. If your equipment is not included in our Wiki and you have difficulties setting up, contact us and we'll be glad to assist you.

We always do our possible to have you up and running. Customer support is an important part of our philosophy. We'll do the best to help you no matter the type of equipment you are using. Wherever possible, we can have a technician enter your server and configure the basics for you.