Frequently Asked Questions - 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

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.