Deploying SIP for Asterisk Open Source PBX
Our SIP trunking service supports the Asterisk’s open-source PBX solution. Selecting SIP.US as your Asterisk SIP trunk provider will help your business reduce costs while getting a flexible, reliable business phone solution.
Asterisk is a free open source platform for communications applications. No special hardware is required because Asterisk runs on an ordinary PC. Asterisk includes IP PBX systems, VoIP gateways, conference servers and other custom solutions. The solution is used by businesses of all sizes in both the private and public sectors worldwide. The solution is sponsored by Digium.
There are currently more than one million communications solutions based on Asterisk in over 170 countries. It can be used as a complete business phone system or can extend an existing one.
Asterisk IP PBX augments SIP trunking by allowing you to create fully customized communication applications. If you require a communication network that can accommodate a changing system, Asterisk can fulfill your wishes. More than that, they’ve made sure to make the building process as easy as possible, so you won’t spend too much time on constructing the application.
The solution was built for developers by developers is most often implemented by developers or system integrators. If you want to create your own applications and solutions with Asterisk it requires a working knowledge of Linux, script programming, networking and telephony. However, if you do not have that technical expertise available, you can still leverage the flexibility Asterisk offers by using pre-packaged solutions built on Asterisk, or by working with an Asterisk consultant.
Why Choose SIP.US for Asterisk
A phone system includes two high-level components. First there’s the PBX (private branch exchange), in this case Asterisk, which includes call management and additional features like voicemail, automated attendants, call distribution groups and so forth. The next component is a connection to the Public Switched Telephone Network (PSTN), which routes calls to their intended recipient. Traditional phone systems use carrier provided PRI (primary rate interface) lines to connect to the PSTN. SIP is an alternative that leverages a data network, rather than telephone lines, to deliver calls to the PSTN.
Customers choose to deploy SIP trunking with Asterisk for a variety of reasons including:
- Most companies recognize a cost savings from deploying SIP.US vs. traditional PRI lines
- SIP.US offers the flexibility to purchase as few as one channel at a time, giving you the ability to pay for only what you need now and grow at any time
- With SIP.US customers can choose the Internet bandwidth that is most reliable and inexpensive for them
- SIP.US offers additional features, including as real-time call data records and Nomatic e911
- We leverage a Tier-1 redundant network to ensure both quality and reliability
- SIP.US has been tested for use specifically with the Asterisk platform and is leveraged in Asterisk deployments across the country
Here is what our customers have to say about using SIP.US and Asterisk.
“I don’t have any experience with SIP trunking, but connecting my Asterisk PBX to SIP.US was fast and easy.” – Eddy Pareja, Sangfroid Web Design
“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.” – Colin Cook, Thimble River
Guide to Setting Up SIP for Asterisk
There are just a few simple steps required to configure SIP for Asterisk and we’ve got them detailed in our knowledge base. We’re also happy to help if you’d like.
SIP for Asterisk FAQs
What kind of Internet Connection do I need to use SIP for Asterisk?
Our service supports any broadband Internet connection. T1-data, Cable, DSL, or Metro Ethernet will all work. Although some SIP providers require you to get connectivity from them, we do not.
Will you help me configure SIP for Asterisk on my PBX?
Absolutely. You’ll need to provide us with remote access to your system, and we will make best efforts to configure your Asterisk PBX for your SIP.US trunks. To get this assistance, please open a new support ticket.
Can I test the service before signing up?
Yes. Just click on the get started button on any page. We offer a free, no obligation trial with 60 minutes of free calling to the lower 48 states and Canada. We will not ask for your credit card or require a commitment before you start your free trial.
Come to SIP.US if you’re looking for a dependable Asterisk provider.