SIP trunking is a method of delivering telephone and other unified communications services over the Internet to customers that have SIP enabled private branch exchange (IP-PBX) solutions. SIP utilizes both Voice over Internet Protocol (VoIP) and Session Initiation Protocol (SIP), and it replaces traditional telephone lines or PRIs (Primary Rate Interface).
Traditional business phone systems consist of two key components. The PBX, which provides call management and features such as Auto Attendants and voicemail, and the PRI lines which connect calls to the PSTN (Public Switched Telephone Network) where they are routed to the destination telephone. When SIP trunks are utilized, the IP-enabled PBX connects to the data network instead of the PRI lines and the voice traffic travels the Internet to connect to the PSTN. This method is typically much more cost effective than PRI lines.
These days, most new PBX systems are SIP-capable. There are several ways to check and see if your PBX is SIP-enabled. First, if your PBX has a data jack or Ethernet jack on the back, there is a good chance that it is SIP-capable. Older PBX or key systems just have analog lines to connect to the PSTN, so if your system does not have a data jack or Ethernet jack, it is probably not SIP-capable. However, in those cases, you can still use what is called an ATA (analog telephone adapter) that will convert SIP over to analog. The ATA will front-end your legacy PBX and allow you to use SIP.US trunks. ATAs come in a variety of sizes, from single port all the way up to 24 analog ports. ATA manufacturers include Cisco/Linksys, ObiHai, Grandstream, and others.
If your PBX has a data jack and you are still unsure if it's SIP-capable, you can check the user manual. You'll want to look for a section on 'configuring a SIP Trunk' or you might find it in the specifications section, typically located at the end of the manual. Look for words like SIP or SIP-enabled IP calling.
You can also contact us with the particular make and model of the PBX. We'll check it out for you and let you know for sure.
Yes. We provide 'Nomadic' e911 service on all of our US DIDs. Simply purchase our 'Enhanced' DID service (or upgrade from a standard DID), and you will have the ability to set e911. Our 'Nomadic' e911 service allows you to set ANY physical address in the United States as your address to be transmitted on 911 calls. This means your calls to 911 will route to the closest PSAP (Public Safety Answering Point) to your registered e911 address on our system. That address will also appear on the emergency services operator's screen when you call. The address can be updated at anytime online via our control panel. Most people will set a single address upon registration of the number and never modify it, but you do have the ability to change the address associated with each e911 DID on our system making it an ideal solution for people who travel between multiple offices.
Configuration, Deployment and Use
When placing a SIP call with SIP.US you will want to make sure that your PBX or device is configured properly using Username / Password authentication or IP address authentication.
An easy way to test a SIP Call with SIP.US is to use a softphone, such as Xlite or Zoiper, and configure a SIP.US trunk directly in the softphone. When making your SIP call from the softphone, you'll want to be sure to dial the country code followed by the area code and then the number. For example, to dial the SIP.US main line, you'll want to dial 15612322200 or 18005669810. When calling other countries, simply enter the country code, followed by the city code and then the number. There's no need to dial 011 in front of the number.
Business Continuity and Disaster Recovery
We maintain redundant SIP proxies (and will be deploying additional proxies in the future). Currently, our SIP proxies are located in Atlanta, GA and Dallas, TX at: atl.sip.us and dal.sip.us. We front end these servers with DNS SRV capability via: srv.sip.us. Most IP-PBX systems will adhere to DNS SRV and if your system supports DNS SRV you will only have to register to srv.sip.us. If your system does not support DNS SRV, you will need to set up two SIP trunks on your device, one for atl.sip.us and one for dal.sip.us. You are able to send/receive calls from either server.
Working with SIP.US
You are welcome to ping our servers, but because our SIP trunking gateways (currently located in Atlanta, GA, and Dallas, TX) are SIP messaging engines only, latency is not an issue. Our engines only process a few thousand bytes of information per call. Unless we forcibly proxy media (at your request or the request of law enforcement), we release all media of the telephone call to your server and the closest media gateway of our underlying carrier. Thus, there is no need to be 'closer' to our gateways or worry about the latency of our servers. The messaging path consists of call setup and teardown with an occasional re-invite.
However, if you still wish to test the latency, you can ping gw1.sip.us and gw2.sip.us. For those wanting to use our SRV entry, please use gw.sip.us.
You can check your Call Detail Records (CDRs) on our Control Panel in the CDR section. Our CDRs are written in real-time to our database. You have the ability to pull 90 days worth of CDRs on the Control Panel. If you require CDRs that are over 90 days old, please open a support ticket.
Direct inward dialing (DID) is a feature offered by telephone service providers for use with their customers' private branch exchange (PBX) systems. Individual numbers are provided to specific subscribers. This makes it possible for a 10 digit phone number to reach a specific telephone within a company, rather than reaching a main line.
We accept all major forms of credit card including MasterCard, Visa, Discover, and American Express. We also allow ACH processing once the initial account deposit is made. We do not store any credit card information on our systems. Instead we utilize the Authorize.net Customer Information Manager (CIM) service which tokenizes and stores our customers' sensitive payment information on the Authorize.net secure servers.
Yes. We offer International DIDs in 63 countries around the world. By default, they are 2-channel max unlimited inbound numbers for a fixed flat rate per month. If additional channels are required beyond the 2-channel maximum, please open a support ticket to get a quote on boosting the number of channels per International phone number.
Ready to see it in action?
Get Your Free SIP TRUNK in 60 Seconds.