As part of setting up outbound calling, you will need to program a legitimate Caller ID (CID) to show in your FROM header of the SIP INVITE. This is not only convenient for the recipients of your calls, but also necessary to comply with STIR/SHAKEN regulation mandated by the FCC. Without a proper CID in place, SIP.US and receiving carriers are not required to pass your call traffic. This article covers the basics of how to provide a legitimate CID to guarantee we pass your calls, options for troubleshooting CID issues, and supplemental information.
What is a Proper Caller ID?
SIP.US requires that you send a legitimate 10 or 11-digit CID in your FROM header of the SIP INVITE for North American calls. You may use international numbers as your CID as needed for international calling.
Note: CID formatting must adhere to the North American standard format NPANXXXXXX. This consists of the area code, prefix, and line number of the DID. You may optionally include the 1 at the beginning for the country code. Only include numbers in your CID. Do not include any letters, dashes, stars, parentheses, or other symbols.
- Industry format: 1NXXNXXXXXX (1-NPA-NXX-XXXX)
- CID examples: 18002506510, 8002506510, 14706102875, 4706102875
You can review your call traffic on the CDRs page to ensure your configuration sends us what you intend and to rule out basic issues. To view CDRs, navigate to SIP Trunking and click Call Detail Records in the submenu.
How to set Caller ID
Caller ID is set in your PBX or phone system. The configuration steps vary depending on what phone system you're using, but many share similar terminology in which you can enter the number into a field within the GUI. Look for terms such as From CID, Outbound Caller ID, etc. You may need to configure the CID per user or within the outbound routes. SIP.US offers some Device Setup Guides for common phone systems, but is not responsible for the configuration of the CID in your phone system.
If you are unable to configure the CID in your phone system, SIP.US additionally offers a way to implement a "CallerID Override" via our Customer Portal. This feature will rewrite whatever you send us in your FROM header with the input in the Customer Portal. To implement the CallerID Override, navigate to the "SIP Trunking" tab and then "SIP Trunks". Choose "Modify Trunk" and then enter the number you are calling from in the "CallerID Override" field and submit the changes.
The disadvantage with using the CallerID Override function is the lack of control over CID per user (or if you have multiple phone numbers), as it will rewrite the CID of ALL outbound traffic through the SIP trunk. For this reason, we recommend setting up caller ID in your phone system for additional control over your traffic.
NOTE: SIP.US can read CID off of the PAI or RPID header instead of the FROM header if preferred. This setting can be updated from the user end by modifying the SIP trunk setting within the customer portal. Hover over the "SIP Trunking" tab and choose the "SIP Trunks" submenu option. Click on "Modify Trunk" for the relevant SIP trunk and then select the preferred option from the "Outbound CID Header" dropdown and click "Submit" to save the change.
Caller ID Versus CNAM
A common point of confusion is the difference between caller ID and CNAM. While both of these concepts apply to identifying a calling party, they are not interchangeable. Caller ID strictly applies to the numerical phone number string identified in the FROM header of the SIP INVITE. Caller ID must be a numerical phone number. CNAM is a feature of the North American Public Phone System which allows for phone numbers to be associated with a name.
If you attempt to transmit letters as an outbound caller ID, your calls will fail STIR/SHAKEN. Refer to this article for more info about CNAM.
What is STIR/SHAKEN?
STIR/SHAKEN is an industry initiative [external] introduced by the FCC to reduce the number of robocalls and fraudulent phone calls through authentication checks on originating caller IDs. SIP.US implemented STIR/SHAKEN in 2021 and has since added stricter requirements for what can pass as a caller ID from our network. If you send an outbound caller ID that does not comply with the NPANXXXXXX format and STIR/SHAKEN, we may FAIL your call outright or it may fail to complete at the receiving end. Please ensure you are sending a legitimate outbound caller ID if troubleshooting failed call issues.
How SIP.US Signs Calls
Call attestation is a way carriers grade the authenticity of a call. When you purchase a number from SIP.US and use that number as your outbound caller ID, we will sign that call with an A. If you choose to use an external number we cannot verify, we will sign that call with a B. We will also sign forwards with a B.
In an effort to further align with evolving industry standards, SIP.US also now signs inbound calls that comply with STIR/SHAKEN attestation. SIP.US inserts a [v], for verified, into the username section of the FROM and PAI headers in the SIP INVITE. Note the following example:
From: “[v]SIP.US INC” <sip:+18002506510@gw.sip.us>
P-Asserted-Identity: “[v]SIP.US INC” <sip:+18002506510@gw.sip.us>
However, attestation of STIR/SHAKEN on inbound is effectively a best effort measure, because call routing information is carried by SS7 protocol when origination is not from VoIP. Due to the way SS7 is handled by public telephone lines, SIP headers cannot be sent to VoIP end users and will cause an unreliable checks on STIR/SHAKEN events. This makes the usability of the SS7 protocol specifically over the PTSN to VoIP problematic and often incorrect when analyzing for STIR/SHAKEN. As a result, not all calls can be reliably verified. For all other calls, CNAM will appear as normal.