SIP for FreePBX
Implementing SIP for FreePBX®
The SIP.US trunking service is completely compatible with the FreePBX® open source PBX solution. Businesses choose to use SIP.US along with FreePBX® because of the flexibility, reliability and cost savings that they enjoy.
FreePBX® is an open source business phone application that is available without charge. It was created and is maintained by a community of developers and others who are dedicated to making sophisticated phone system software that is easy to use and useful. The software has been downloaded more than 5 million times. FreePBX® is sponsored by individuals and companies who are dedicated to the vision of a free, easily implemented business phone solution. Schmooze Com Inc. is the Principal corporate sponsor and they offer a commercially supported version of FreePBX®. FreePBX® enjoys strong relationships with Digium, the sponsors of Asterisk.
FreePBX® has been built by a community of software developers and telco experts. Although it is designed to be easy-to-implement, customization of the solution requires a good knowledge of script programming, telephony and networking. Thorough documentation and various paid support and professional services packages are available. For those who prefer a pre-packaged solution, there are several to choose from so that you can take advantage of the ease and flexibility of the solution.
Why Choose SIP.US for FreePBX®
Every phone system needs PBX (private branch exchange) software that handles things like call management, voicemail, call distribution groups, automated attendants and the rest. FreePBX® provides this functionality. In addition a phone system needs at least one phone number and access to the network that routes calls to numbers outside of the system. This is called the Public Switched Telephone Network or "PSTN." Traditionally, businesses used carrier provided PRI (primary rate interface) lines to send calls to the PSTN. With SIP, calls reach the PSTN through a data network rather than traditional telephone lines. Customers choose to deploy SIP for FreePBX® using SIP.US for several compelling reasons:
- A significant cost savings can be gained by deploying SIP.US vs. traditional telephone lines
- Customers enjoy the fliexibility that SIP.US provides by allowing the purchase of one channel at a time. (Traditional PRI lines are sold in bundles of 23 channels.) With SIP.US customers pay only for what they need and can expand on-demand.
- SIP.US does not require customers to purchase Internet bandwidth from us, allowing them to choose the most cost effective and reliable network in their area.
- SIP.US offers additional features, including a powerful, yet simple control panel for administration, excellent International calling rates and real-time call data records.
- In order to ensure quality and reliability, SIP.US leverages a Tier-1 redundant network
- SIP.US is used along with FreePBX in deployments across the country
How To Set Up SIP for FreePBX®
The SIP.US Module makes it easy to configure your SIP trunks, outbound route and inbound routes for SIP.US DIDs within FreePBX®. Just visit our knowledge base for a step by step configuration guide.
Start your trial today and join thousands of companies that are enjoying reduced telco costs and complications with SIP for FreePBX®.
SIP.US works with any broadband Internet connection including; Metro Ethernet, DSL, cable or TI-data. Although some SIP providers require customers to get Internet connectivity from them, we do not and you are free to choose the least expensive, fastest and most reliable connection available to you.
Here is what our customers have to say about working with SIP.US:
“It’s easy to be low cost, but it’s a lot harder to be low cost AND good! In my opinion, SIP.US should be on anyone’s short list of primary providers.”
– Michael Dodds, Doddstech
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