FAQ

About SIP

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.

PBX Support

How to Verify That Your PBX is SIP-Enabled

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.

Can you connect to a legacy analog PBX or key system?

Which PBX systems and devices do you support?

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.

Does your SIP trunking service work with Asterisk, Elastix, FreeSwitch, PiAF, or other popular Graphical User Interfaces to configure and control Asterisk?

What VoIP protocols and voice codecs do you support?

Supported Features

Do you provide e911 service on your DIDs?

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.

Do you provide 411 directory information service?

Do you provide CNAM (CallerID Name) service on inbound numbers?

Do you provide CNAM (CallerID Name) service on outbound calls?

Can I use SIP.US service for faxing?

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.

Do you support T.38 faxing for your DIDs?

Configuration, Deployment and Use

How to place a call using SIP.US?

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.

What kind of Internet connection do I need for your service?

How much bandwidth do telephone calls consume on your service?

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.

Where can I find configuration instructions for your SIP trunking service?

How many simultaneous calls are allowed on the unlimited SIP Trunks?

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.

What ports do I need to forward to my system if it is behind NAT?

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.

What methods of SIP trunk authentication do you offer?

Do I need a dedicated public IP address to use your service?

What is QoS (Quality of Service) and how do I enable it?

QoS or Quality of Service allows for the prioritization of traffic on your network and is usually configured in your router settings. It is imperative that QoS be properly enabled to ensure the successful transmission of VoIP phone calls. Without QoS, you run the risk of audio breakup and voice choppiness whenever your broadband Internet connection becomes saturated. For example, if you are downloading a large video file from a website and you do not have QoS enabled and a VoIP telephone call was initiated at the same time the voice packets would get lumped in along with the packets of the download. The audio would break up and in some cases become unintelligible until the download was complete. (To test QoS is properly enabled, you can initiate a large download on your computer while making a telephone call.) Fortunately, there are inexpensive routers with robust easy-to-configure QoS.

Business Continuity and Disaster Recovery

How does your redundant service architecture work?

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.

How does your inbound DID routing redundancy and failover logic work?

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.

Do you offer PSTN forwarding?

Working with SIP.US

How do I test the latency to your servers?

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.

What services does SIP.US provide?

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.

How can I try the service?

How quickly can I be up and running?

How do I sign up for service?

Do you offer hosted PBX service?

How do I get support?

Reporting

How do I check my CDRs (Call Detail Records)?

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.

Telephone Numbers

What is a DID?

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.

What calling area do the unlimited SIP trunks cover?

What is your US DID footprint?

Do you offer Toll-Free DIDs?

Do you offer vanity telephone numbers?

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.

Can I port my telephone numbers to SIP.US?

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.

What happens if I cannot find a DID in a particular area code?

Payment

What methods of payment do you accept?

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.

How can I change my method of payment on file?

Can I upgrade or downgrade my unlimited channel rate plan?

How do I cancel my service and can I cancel at anytime?

What are federal USF/TRS charges?

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.

How much are federal USF/TRS charges?

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.

Am I exempt from federal USF/TRS charges?

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.

International Calling

Do you offer International DIDs?

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.

Do you offer international service and what are the rates?

How does your International call fraud protection work?

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.

What is the international calling country blacklist and how can I disable it?

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.

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FAQ