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.
Absolutely. This is done via a multi-port analog telephone adapter. We extensive experience and recommend the Grandstream multi-port ATA’s. They come in 4,8 and 24 port increments. These multi-port ATA’s are very easy to configure and have proven to be very reliable with our customer deployments. They can be purchased from Amazon.
We support any PBX system that is SIP-enabled. This includes popular VoIP IP-PBX’s like Asterisk, FreePBX®, Trixbox, Switchvox, PBX in a Flash, Elastix, Bluebox, FusionPBX, 3CX, sipXecs, Go Auto Dial, Vicidial, Thirdlane, and more. Many traditional PBX manufacturers also support SIP trunking with their latest software releases. These SIP-capable PBX vendors include Toshiba, Panasonic, NEC, Avaya, Cisco, Nortel, Intertel, and others. For those systems that do not support SIP trunking, we support SIP-T1 gateway devices where it’s SIP-in and T1-out to the PBX. Examples of these types of gateways are the Digium G100 and G200 gateways. In addition, we fully support analog telephone adapters (ATAs) which can interface to legacy analog PBX’s and key systems. We recommend the Cisco SPA-2102 for a single port ATA interface and the Grandstream GXW400X series multi-port ATAs for interfacing with analog systems that require more than one line.
Learn how to Setup SIP Trunk for FreePBX, Cisco SIP Trunk, Asterisk and others in our Knowledge Base.
Our SIP trunking service works perfectly with Asterisk, FreeSwitch and other open source telephony applications including popular Graphical User Interface applications used to configure and control Asterisk. We provide detailed configuration instructions for these systems and will even help you configure the trunks if provided remote login credentials for your server.
We currently only support the SIP protocol. We support both G.711 and G.729 voice codecs for calling.
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.
Calls to 411 will route on our system, we send those calls to 1-800-FREE-411. It is a free, ad-sponsored directory assistance application.
Yes. Inbound CNAM (Caller ID with Name) delivery is available on all incoming calls (assuming it was provided to us by our upstream carrier).
Yes. We offer this service on most all of our US48 DIDs. Simply visit your DID management page on the control panel to set this 15 character alphanumeric CNAM string. The first setting is no charge and included in the price of the DID setup fee. Subsequent CNAM changes to a DID are $1 per update.
While it is possible to use the SIP.US service for faxing by placing an ATA (analog telephone adapter) behind a standard fax machine, keep in mind that calls over the public Internet are not designed to deliver faxes reliably 100% of the time. Traditional fax machines have fax modems which rely on data being delivered to them predictably without jitter or latency. With VoIP calling (and faxes riding on that connection), we can’t guarantee that fax transmissions will appease the finicky fax modems contained in most fax machines. We have extensive experience running faxes over the Internet and can say with confidence that if you are a light duty user of fax, meaning 1-2 faxes per week, an ATA might be a good solution for you. You may have to re-transmit your fax once out of every ten times. If you can live with this level of reliability, then, by all means, try it out. If you are a heavy user of fax, or if fax is a critical part of your business, then we recommend our SimpleFAX Secure FAX ATA. Our secure AudioCodes HTTPS fax device eliminates problems associated with traditional Fax over IP solutions, including unreliable ATAs, T.38 and G.711 fallback. Our solution uses store-and-forward technology to securely send faxes over an IP connection with a 100% success rate. Faxes are sent directly to our device which sits behind the fax machine and captures the entire fax from the fax machine then transmits it over HTTPS to our PSTN-based Fax facility for reliable transmission. If you don’t use a fax machine but still need to send and receive faxes we recommend our Fax-to-Email and Email-to-Fax service. If our recommended solutions still don’t fit the bill, we recommend a dedicated analog fax line from the phone company.
We support T.38 faxing on most US DIDs. This applies to incoming T.38 fax calls only. We support T.38 outbound faxing on a ‘best-efforts’ basis as not all of our upstream carriers fully support T.38 across all of their media gateways.
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.
Any broadband Internet connection will work with our service. Cable, DSL, T1-data, or Metro Ethernet are all supported. Unlike some other SIP providers, do not require you to get connectivity from us.
Telephone calls on our system are by default configured to operate on the G.711 voice codec, which consumes 85kbps of Internet bandwidth up and down. For example, a small DSL connection of 512kbps up and 3M down will have a limiting factor of the upstream 512kbps limit. Take 512 and divide it by 85 to arrive at a total of 6 maximum simultaneous calls on that particular type of Internet connection. That’s the very low end. Most broadband Internet connections these days are much faster, to the point where dozens and dozens of calls can traverse the data connection.
Wondering how to connect your PBX to a SIP trunk? Our Knowledge Base, located at http://support.sip.us contains examples of SIP trunk configurations for a variety of systems and devices along with other useful information to help you get started.
Each unlimited 2-way inbound / outbound channel can handle one SIP call (inbound to your local numbers or outbound to US48 / Canada). For example, if you have a PBX with 8 trunk lines, you would select 8 Channels for your rate plan. Our SIP Trunks can handle multiple calls, meaning in this example you don’t need to buy 8 SIP Trunks, you can get 1 SIP Trunk with 8 unlimited channels. If you need to upgrade to 30 unlimited channels in the future, you can do that all on the same SIP Trunk.
We recommend forwarding ports UDP/5060 and UDP/10000-20000 for standard FreePBX/Asterisk based installs. It may be possible to get your service working without port forwarding, but optimal service will be obtained with these ports. You can lock down port UDP/5060 to atl.sip.us and dal.sip.us for additional security. You can not lock down UDP/10000-20000 to any specific IP address as we release the media on all calls to the closest carrier media gateway for optimal performance.
Learn how to Setup SIP Trunk for Asterisk PBX, FreePBX and others in our Knowledge Base.
We offer the ability to do username/password or IP address authentication for our SIP trunks. These settings are configurable on your Control Panel under the Trunks page.
No. You are not required to have a dedicated public IP address to use our service, although if you can obtain one it is certainly recommended. If you have an Asterisk/FreePBX behind NAT, you will need to make a small modification to the sip_nat.conf file on your system (exact instructions can be found in our Knowledge Base.)
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.
For each DID telephone number in your SIP.US account you are able to specify a Primary SIP trunk, Secondary SIP trunk, and PSTN failover number. This means that we will first attempt to send the incoming call to your Primary SIP trunk (presumably your Primary IP-PBX or ATA), if for some reason there is no response from that destination, we will attempt to send that call to your Secondary SIP trunk (presumably your Secondary IP-PBX or ATA), then if that still fails you have the option of finally routing the call to a PSTN failover number of your choice. You do not have to have a Secondary SIP trunk listed if you wish to use the PSTN failover feature, you can just have a single Primary SIP trunk and a PSTN failover number if you prefer.
Yes. For each DID telephone number in your SIP.US account, you have the option of turning on PSTN forwarding. When set, we will ignore the primary and secondary trunk assignments and send your call directly to the PSTN number you have set without exception.
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.
SIP.US offers business class SIP Trunking service consisting of Origination (inbound) and Termination (outbound) calling. We also offer DID telephone numbers in the US, Canada and around the world. All of our services can be instantly provisioned online allowing you to get up and running immediately. Everything about our service can be easily managed online using our robust Control Panel application.
Simply click on the get started button on any page. Your free, no obligation trial comes with 60 minutes of free calling to the 48 contiguous US states and Canada. There is no credit card or commitment required.
All of our services are provisioned instantly. Free trial accounts are available as soon as you confirm your email address. For paid services, verification of payment method may be required. If that is the case, it could take up to 24 hours to validate your payment information, but payment verification is typically done within 1 hour during normal business hours.
It's easy to get started, simply click on the Get Started button and register for a free account.
We do not offer hosted PBX service. Instead, we partner with a number of hosted PBX providers. We recommend Host Virtual, which is a leading provider of many hosted VoIP appliances. Specifically, we have found the FreePBX distribution to be easy to use and setup. Visit the Host Virtual website for more information. Learn more about FreePBX SIP trunk in our Knowledge Base.
The easiest way to get support is by opening a support ticket at: support.sip.us. You can also email email@example.com which will automatically open a support ticket assuming you have an account established with us. If you need to call us, you can reach us at 800-566-9810, although we recommend logging a support ticket or live chat for the fastest response times.
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.
Our unlimited SIP trunks cover outbound SIP calls to the 48 contiguous US states and Canada (excluding the Northwest Territories of Canada). Inbound local DIDs for the 48 contiguous US states are also included in the unlimited SIP Trunks.
We have one of the largest DID footprints of any SIP trunking provider with access to over 6,000 rate centers in the United States. Whereas most other SIP trunk providers will only have a handful of numbers in each rate center, you will often find hundreds of numbers in each area code across the country in our inventory.
Yes. We offer toll-free DIDs which can be instantly added in your Control Panel. Toll-free pricing can be found on our main pricing page.
We do not currently offer vanity telephone numbers. Vanity local numbers are nearly impossible to come across as the various local phone companies maintain their own blocks of these numbers. Typically, the only way to get a vanity local number is to request a single analog line to your location from the local phone company with the vanity telephone number you want. Then, once the service is officially established and you receive your first bill, you can then port the number over to SIP.US. It is a somewhat painful process, but unfortunately, there is no way around it. Toll-Free vanity numbers, on the other hand, are easier to come by. We partner with www.tollfreenumbers.com the leading provider of toll-free vanity numbers in the US. Simply browse and purchase a toll-free vanity number through their website, and port it over to us. They will provide you with the necessary documentation for porting, making the porting process smooth.
Yes. You can port US48 numbers to SIP.US. The pricing for porting is located on our main pricing page. To initiate a port request, simply visit the Control Panel and click on ‘Number Porting’. From there, you will have detailed instrucitons on how the porting process works, what the timeframes are, and have the ability to generate new online port requests. You can also check the status of your ports from the Control Panel as well.
We are able to backorder numbers in a particular area code if not available. It normally takes 5-7 business days to fulfill a backorder request. To initiate a backorder, please open a support ticket with the specifics on the area code and rate center you are looking for.
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.
To add or change a method of payment on file with us, please visit the billing section of the Control Panel.
Yes. You can upgrade your number of unlimited channels anytime in the Control Panel under in the ‘Trunks’ section. Simply click on ‘Modify Rate Plan’ and we will pro-rate the channel upgrades. Channel downgrades are effective on your next billing cycle.
The Federal Universal Service Fund (USF), created by the federal government, is designed to help ensure first-class, affordable telecommunications service for all consumers across the country, especially residents in high-cost rural communities and low-income customers. Additionally, the Federal USF provides for discounted telecommunications services for schools, libraries, and rural healthcare facilities. All telecommunications providers are required to pay into the Federal USF, and their contributions may be recovered from customers. The FUSF is administered by the Federal Communications Commission (FCC) and the Universal Service Administrative Company (USAC). Telecommunications Relay Service (TRS) is a telephone service that allows persons with hearing or speech disabilities to place and receive telephone calls. TRS is available in all 50 states, the District of Columbia, Puerto Rico and the U.S. territories for local and/or long-distance calls. TRS providers – generally telephone companies – are compensated for the costs of providing TRS from either a state or a federal fund. There is no cost to the TRS user.
Effective April 1, 2003, the Federal Communications Commission (FCC) has changed the way in which carriers can recover their USF obligations. Rather than a flat-rate monthly charge, which customers have seen in the past, SIP.US began recovering the USF based on a percentage of each customer’s telecommunications charges. The percentage is based on a contribution factor announced by the FCC and is subject to modification by the FCC. More information along with current and proposed rates is available on the FCC website.
All customers must pay the Federal Universal Service Fund and Telecommunications Relay Services Fund charges. Under FCC rules and policies, telecommunications service providers (including resellers) that pay FUSF contributions directly to USAC are typically exempt from paying FUSF charges to underlying providers such as SIP.US. Accordingly, SIP.US will not bill FUSF charges to a customer that is a telecommunications services reseller and demonstrates that the customer: (a) has registered with the FCC and/or USAC under the FUSF program; (b) has a valid Filer 499 ID as proof of such registration; and (c) is listed on the FCC’s website as a direct contributor to the FUSF program. To request a SIP.US USF exemption form, please contact the SIP.US billing department.
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.
Yes. We offer international outbound calling service. We offer competitive rates.
We attack the problem of international call fraud on a number of fronts. First, we turn off International calling by default for all newly provisioned SIP trunks. You are able to modify this setting in the Control Panel. Second, you have the option to set the type of international calling ability on each trunk on our system (North America-only, or all international). In addition, you are able to set the maximum threshold for a calling destination for each trunk (by default it is $.25 per minute). SIP.US also maintains proprietary real-time International call fraud protection systems that monitor for potentially abusive international dialing patterns. If we suspect your SIP Trunk has been compromised, we will automatically terminate calls in progress, disable international calling on your trunk and notify you of the incident.
For the protection of our customers, we maintain a default blacklist of international calling destinations that are known for high fraud potential. The list is enabled by default on all of our trunks. If you have a requirement to call these destinations, you may request that the blacklist be removed for your account. In order to do this you must agree in writing that you will be responsible for any and all calls through your system regardless if they are legitimate or fraudulent. If you wish to obtain a copy of this form, please contact support and request the Blacklist Removal form.