FreeSWITCH is an open source, free platform for the creation of voice and unified communications applications. It is capable of being embedded into other projects, or being used as a stand-alone application. FreeSWITCH enables many telephony applications, including call control and IVR functionality. FreeSWITCH is available for source compilation on Unix and Linux distributions as well as Windows.
FreeSWITCH™ has a modular design that allows it to be extended without impacting its function or requiring a large development effort. Each module does something specific and often related tasks are grouped into one module. You will need to enable the modules that you would like based on their function. FreeSWITCH comes with a set of modules loaded, to enable most basic functionality. Complete documentation is available to walk you through the implementation. For those less familiar with telephony and software implementation, paid consulting and support are available.
Why Choose SIP.US for FreeSWITCH?
Two primary components make up a business phone system. The first is the software that handles call control and other phone system features such as conferencing, voicemail, automated attendants and others. This is often referred to as the PBX. FreeSWITCH plays the role of the PBX. The second necessary component is a way to deliver calls to the network that routes them to their intended recipient. This network is called the PSTN (public switched telephone network). In a traditional phone system, carrier provided PRI (primary rate interface) lines connect the PBX to the PSTN. SIP is an alternative that uses data lines, rather than telephone lines to make the connection. Customers choose to deploy SIP for FreeSWITCH using SIP.US to gain a variety of benefits:
- Using SIP.US with FreeSWITCH is usually much less expensive than traditional telephone lines
- Traditional PRI lines are sold in groups of 23 channels. SIP.US allows customers to subscribe to as few as one channel at a time, giving customers the flexibility to pay for only what they need now
- Although some SIP trunking providers require clients to purchase Internet connectivity from them, SIP.US does not. Our clients choose the Internet bandwidth provider that makes the most sense for their needs and budget
- SIP.US comes with a powerful, easy-to-use online interface for administration of your SIP trunk
- We utilize only Tier-1 upstream providers to route our traffic, giving our customers the most reliable network available
- SIP.US has been thoroughly tested with FreeSWITCH. They are used together in deployments across the US.
How To Set Up SIP for FreeSWITCH
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SIP for FreeSWITCH FAQs
Any broadband Internet connection will work with SIP.US and FreeSWITCH. Choose from DSL, T1-data, cable or Metro Ethernet. SIP.US gives our clients the flexibility to choose the Internet connectivity option that is best for them and unlike many SIP providers, we do not require customers to purchase Internet access from us.
Here is what our customers have to say about working with SIP.US.
We were able to get our systems up and running with SIP.US in a matter of minutes using the simple online interface.
Elevate Audio Visual
Low cost and good!
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.
All the tools I need.
The SIP.US Control Panel gives me all the tools I need to add, change and delete phone numbers. I can even order all the DIDs I need and set them up instantly. I love having access to these self-service tools, but I know the support team is also there if I need additional help.
Up and running in minutes.
We are able to get our Switchvox systems up and running with SIP.US in a matter of minutes using the simple online interface.
Quality support, low price.
I wasn’t sure how to configure my Asterisk system to work with SIP.US. The SIP team was able to remotely access my system and set it up for me. I didn’t expect that level of support for such a low price.
The Galvin Law Group
Fast and easy set-up.
I don’t have any experience with SIP trunking, but connecting my Asterisk PBX to SIP.US was fast and easy.
Sangfroid Web Design
The reason we moved most of our VOIP to you is because of your excellent support. We moved from another provider but your support makes the extra money well worth it.
CEO BMI SmartCloud