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Calling Name showing as SPAM? Here's what you can do   You’re trying to reach your customers, and instead your calls are getting ignored or blocked because they are being labeled as Spam Likely, Scam Risk, Robo or similar..… you’re not alone.    This is a very common industry-wide issue and we’re on your side in helping you work through it.   Why This Happens Call labeling is handled by Wireless carriers (like AT&T, Verizon, T-Mobile) and Third-party analytics companies (like TNS, Hiya, First Orion, Neustar).   These systems are designed to stop spam and fraud, however sometimes legitimate businesses can get caught in the mix and end up getting mislabeled.   When the analytics companies determine if calls are related to spam, they look at the following: How your numbers are being used Calling patterns and volume Customer complaints or blocking activity Historical reputation of the number Calling Name (CNAM)   What It Looks Like Different carriers and analytics providers label calls differently.   Call Label Carrier / Provider Fraudulent Call Altice, Fidelity Communications, Telzio, Windstream Spam Risk,  Fraud Risk AT&T Blocked – High Spam Risk Comcast (Xfinity) Potential Spam C-Spire, US Cellular, Verizon Robo Caller Frontier, Lumen Spam Likely Spectrum (Charter) Scam Likely T-Mobile Spam Truecaller Robo? Or Spam? (Prepend) Altice, Comcast (Xfinity), Fidelity Communications, Windstream   How You Can Fix It There isn’t a single instant fix, but the following steps will give you the best chance of improving the issue overall.   1. Configure Calling Name Value (CNAM) Make sure your Calling Name is set with an accurate name customers will recognize. Missing, generic, or incorrect CNAM can negatively impact how your calls are scored, making them more likely to be labeled as spam. 2. Register your numbers (most important step) Use the Free Caller Registry website to register your numbers. This site will register your numbers with the major analytics providers (e.g. First Orion, Hiya, TNS) that support multiple phone providers to help establish that your calls are legitimate. Analytics Company Carrier / Provider Registration Link Free Caller Registry  Multiple http://www.freecallerregistry.com/  First Orion  Tmobile https://firstorion.com/register/ HIYA  ATT https://hiyahelp.zendesk.com/hc/en-us/requests/new TNS  Verizon https://tnsi.com/robocall-mitigation/ Neustar  Multiple http://www.home.neustar/support     Note - After registering, analytics systems will still evaluate your traffic based on behavior.   3. Follow Calling Best Practices Call labeling is influenced by how calls are placed and how recipients interact with them. For a full list of recommended calling practices, see Calling Best Practices section below.   What If the Issue Persists? If your calls are still being labeled as spam after updating CNAM and registering with the Free Caller Registry, the next step is to submit your number(s) directly to the provider responsible for labeling on the affected carrier’s network. Each provider has its own process and review timeline.   Note - If you are unsure which provider applies to a specific number, Voyant Support may be able to assist in identifying the correct carrier.   Provider and Carrier Labels Carrier / Provider Labeling Altice Fraudulent Call or CNAM prepend of Spam? or Robo? ATT Spam Risk or Fraud Risk Comcast (Xfinity) Blocked – High Spam Risk or CNAM prepend of Spam? C-Spire Potential Spam Fidelity Fraudulent Call or CNAM prepend of Spam? Frontier Robocaller Lumen Robocaller Spectrum  (Charter) Spam Likely Telzio Fraudulent Call T-Mobile Scam Likely Truecaller Spam US Cellular Potential Spam Verizon Potential Spam Windstream Fraudulent Call or CNAM prepend of Spam?   What if the Issue Persists After Working with Provider? If you’ve engaged the problem provider and you’re still stuck, bring it to us.   We can help with the following: Help you understand what might be triggering the labeling Guide you through next steps Assist where escalation is possible At the same time, we want to be transparent: Carriers and analytics companies control the labeling systems They’ve intentionally pushed registration and reputation management to end users That means we don’t have a direct remove spam label button But that doesn’t mean you’re on your own. We’ll work through it with you.   Calling Best Practices   Build Trust with Consistent, Legitimate Traffic Maintain consistent, natural calling patterns Avoid sudden spikes in volume or very short duration calls at scale. Predictable usage builds trust over time. Ensure strong customer validation (KYC) Confirm that traffic on your account is legitimate. This helps protect overall number reputation.   What Can Accidentally Make It Worse The first few calls from a number are more likely to be labeled due to no reputation Testing new numbers repeatedly This is one of the most common issues. While it feels logical to test a new number: Numbers with very low call volume are often flagged as suspicious Cycling through numbers reinforces this pattern This can create a loop where every new number continues to be flagged.   Day-to-Day Calling Practices Avoid high-volume call bursts Spread outbound calls over time to avoid looking like automated traffic. Limit repeat calls to the same number Frequent attempts in a short window can increase spam risk. Keep calling patterns consistent Sudden changes in volume, timing, or location can trigger filtering. Use consistent outbound numbers Frequently rotating numbers can reduce trust and reputation. Watch answer rates and rejections Low answer rates or frequent declines can impact how calls are labeled. ... View more

SIP REFER is a method defined in the Session Initiation Protocol (SIP) that allows one endpoint in a call to instruct another endpoint to initiate a new SIP request to a third party. In practice, it is most commonly used to perform call transfers. Instead of the transferring device directly establishing the new call itself, it sends a REFER request that tells the receiving endpoint where to send a new INVITE. The destination is specified in the Refer-To header, which contains the SIP URI of the party to be contacted. Once the REFER request is accepted— with a 202 Accepted response—the receiving endpoint initiates the new call to the referred destination. In a typical call transfer scenario, two parties are already connected. When one endpoint wants to transfer the call, it sends a REFER request to its SIP server or call control platform. The server acknowledges the request and then initiates a new INVITE to the target identified in the Refer-To header. This creates a new call leg between the platform and the referred endpoint. Because the REFER method triggers another SIP transaction rather than carrying media itself, it operates strictly at the signaling layer of the SIP protocol.     SIP REFER Support in Sinch Elastic SIP Trunking   Sinch Elastic SIP Trunking supports SIP REFER requests initiated by customers for both origination and termination traffic. When a REFER request is sent to the Sinch Elastic SIP Trunking platform, the handling of the transfer depends on the URI contained in the Refer-To header. Same Domain with New Phone Number: If the Refer-To URI contains a new phone number but retains the same domain, the call will be forwarded to the PSTN. Calls forwarded to the PSTN are treated as new outbound calls on the trunk and are therefore subject to the same routing rules, policies, and restrictions that apply to any standard outbound call. The new call leg to the PSTN  will be billed separately as if it were a regular call from your platform. Different Domain: If the Refer-To URI contains a different domain, the call will instead be forwarded to the specified SIP destination. In this case, the destination does not need to be included in your Access Control List (ACL), and any routing rules or restrictions configured for the trunk will not be applied to this transfer. Calls transferred in this way do not incur any new fees.  Referred calls can be found in your portal on the Call Detail Records page.  The new calls legs will have a second call record showing the referred call. ... View more

Sinch actively maintains a knowledge base available here at https://community.sinch.com/ - you do not need to register unless you would like to post or comment on an article. More specifically, you can find our Elastic SIP Trunking articles under the Voice & Video section. Be sure to subscribe to our product release notes, also, in the Sinch Community and let us know if you have questions.   If you can't find the answer you're looking for in our knowledge base, you can contact support. To raise a support request, you have a couple options: You can create a ticket directly into our ticketing system at support.sinch.com.  Technically, you don’t need to create a login on the site in order to create a ticket.  It is recommended, though, so that you can manage your current tickets and review past tickets.   To create a login, click Sign in in the top right corner and then click on the Sign up link on the next page.    To submit a ticket,  click on the Submit a Request button, where you can report technical issues, request number porting, ask questions, and more. And, there’s even the ability to attach files, screen shots and PCAPs.  For EST issues please include the product information and Voice services (Elastic SIP Trunking):   You can also email support@sinch.com. Please include "EST" or "Elastic" in your subject line.  This will help our system properly route your ticket request. ... View more

Sinch’s Elastic SIP Trunking product will forward any CNAM (Caller Name) data that you include in your SIP INVITE. However, due to the nature of PSTN call routing across multiple carrier networks, this CNAM information may not always be preserved end-to-end. To help ensure that your company’s caller ID name is displayed to call recipients, Sinch offers an enhanced option: registering your desired caller name to each telephone number in a national CNAM database.   How CNAM Display Works   Database-Driven Display: When you make a call, the recipient’s carrier performs a CNAM database lookupusing the caller’s phone number to retrieve and display the registered name. LIDB Repositories: These lookups occur through CNAM repositories, which are part of what’s known as the Line Information Database (LIDB) system. Dependent on Recipient’s Carrier & Plan: Successful CNAM display depends on whether the recipient’s phone service supports CNAM lookups. Some carriers require a paid feature for CNAM to be shown on inbound calls. Best Practices for CNAM Delivery   Include CNAM in the SIP INVITE when placing outbound calls. For best success, Sinch recommends placing the CNAM data into SIP privacy headers such as Remote Party ID (RPID) or P-Asserted- Identity (PAI). Consider registering CNAM values with Sinch for each number to improve reliability across networks. Ensure that your use of CNAM complies with all applicable telecom regulations and anti-spoofing requirements. ... View more

SUBSCRIPTIONS: If you want to subscribe to updates for Elastic SIP Trunking, click the Options menu button in the right-hand-corner of this article and select Subscribe. You will then receive email notifications when this article is updated with new release notes.   Check in here to see information about new releases from Elastic SIP Trunking.   3/20/2026 - SIP REFER Support   Sinch Elastic SIP Trunking customers may now send SIP Refer messages from their endpoints after a call has been established. More information on how to use SIP Refer can be found here. https://community.sinch.com/t5/Elastic-SIP-Trunking/How-do-I-use-SIP-REFER/ta-p/19346   1/14/2026 - PIDF-LO Support   Sinch Elastic SIP Trunking customers may now send Presence Information Data Format – Location Object (PIDF-LO) information when making a call to 911 Emergency Services. This information is passed along to the 911 operator to provide more precise information as to the callers location. ... View more

Bring your voice systems into the future with Elastic SIP Trunking. Explore the flexible set up, pay-as-you-go pricing, and seamless scalability in our latest demo video with Sinch Director of Product, Steve Greenberg:   ... View more

Do Not Originate (DNO) is a feature that allows a Sinch Elastic SIP Trunk customer to designate numbers to be one way inbound only. This applies to DID and Toll-Free numbers. This prevents the selected numbers from being able to make outbound calls.   Note: DNO is only available for North American phone numbers.   A Do Not Originate (DNO) list is essentially a blocklist of telephone numbers that should never be used to make outbound calls. If a call appears to come from a number on this list, it is almost certainly fraudulent or "spoofed" and should be blocked.   By maintaining a DNO list, service providers can automatically flag and block calls from these numbers, preventing them from ever reaching a consumer.   Prior to any setup or routing of call, the network will first verify if the originating number is allowed to make a call. If not, call will be immediately rejected with: Recording:  “Sorry, but your call cannot be completed” SIP 603 (ISUP 21)   This can/should result in: Elimination of illegal robocall activity through the network if it is coming from numbers that have subscribed to DNO through any participating entity or an invalid number and Elimination of invalid formatting of calls through the network. By subscribing numbers to the Sinch DNO list, the numbers will be entered into multiple DNO databases as follows:   Somos RealNumber DNO US Telecom Industry Traceback Group Neustar YouMail Sinch Voice Customers Sinch Voice Reserve Inventory   How to Add Numbers to the DNO List   Open a ticket with support in the Sinch Dashboard at https://support.sinch.com or send an email to: support@sinch.com. The authorized user on the account must be the individual who needs to make the request.   How to Remove Numbers From the DNO List   Open a ticket with support in the Sinch Dashboard at https://support.sinch.com. or send an email to: support@sinch.com. The authorized user on the account must be the individual who needs to make the request. ... View more

This guide will walk you through the simple process of activating your trial for Elastic SIP Trunking (EST), understanding its features and limitations, and how to request an upgrade to a full production account.   Activating your trial   Log in to your Sinch account. From the main dashboard, you'll see an overview of the products available to you. Note, if you’ve just signed up for an account, you will have some test credits you can use for Elastic SIP and other services.   Navigate to Elastic SIP Trunking. On the left-hand navigation menu, expand the Voice section and click on Elastic SIP Trunking. Start your journey. On the overview page, click the Activate trial button to begin.   Review the trial limitations. A modal window will appear outlining the limitations of the trial account. You’ll need to acknowledge these terms before proceeding. Your trial access to Elastic SIP Trunking comes with the following feature limitations: May only place calls to verified numbers. Calls are limited to 5 minutes in length. May only place calls with the following countries: Canada, Puerto Rico, United States, United States Virgin Islands. After you accept, a welcome message is displayed:      Navigating your trial account   Once your trial is active, you'll notice a banner at the top of the Elastic SIP Trunking pages indicating that you are using a trial version (and an option to request an upgrade).       Verified numbers are key   During the trial, you can only make calls to numbers that you have verified for your account. This is an important security measure. You can manage your verified numbers by navigating to Numbers > Verified numbers. For more information on how to add a verified number, please see our guide: How can I add a verified number to my account?     Build trunks   With your verified number you can create new SIP trunk(s) and start making calls. See more information on setting up Elastic SIP in the Getting Started with Elastic SIP community article. The cost of each call will be deducted from your test credits.     Obtaining your free test number   Navigate to the Numbers section, and you will be presented with a banner which will assist you in obtaining your free test number.   Once you have obtained your test number, you will need to move it over to your Elastic SIP Trunking Trial to use it with the service.  To do this, navigate to Your Virtual Numbers underneath the Numbers section, and change the Voice Configuration from “Voice” to “Elastic SIP Trunking”.       Transition to a Full Production Account   Ready to unlock the full potential of Elastic SIP Trunking? Here’s how to apply for a full production account. Our Elastic SIP Trunking service is currently a sales-contracted offering designed for business use. The application process is in place to ensure we can meet your needs and establish a smooth, long-term partnership. Begin Your Application: Click the "Apply for Production" button in the banner at the top of any Elastic SIP Trunking page. Provide Your Business Details: A form will appear asking for information about your intended use of the service. This information is the first step in our sales process, which will also include standard credit and KYC (Know Your Customer) checks. Connect with Sales: Once submitted, our sales team will review your application and reach out to discuss your needs and guide you through the contracting process. Be on the lookout for an email from our team.   A Note on Billing: For now, EST customers must be billed through a North American entity. While our sales contracts are written in USD, our billing team can often accommodate payments in other currencies.   We hope this guide helps you get started with your Elastic SIP Trunking trial. If you have any questions, feel free to explore the rest of our community articles or reach out to support. ... View more

Call Blocking enables you to configure project-specific rules that filter calls based on your specified criteria. This can be as granular as blocking a specific number from being able to call your numbers, to blocking all calls on a project. Here‘s how to configure it.         Rule name   You can configure multiple rules within your project. To keep them organized, assign each rule a unique identifying name.   Direction   The rule's direction determines whether it applies to calls originating from or terminating at your SIP infrastructure. This direction is always defined from your SIP infrastructure's perspective. For example, 'Inbound' refers to calls from the Elastic SIP Trunking Service or PSTN directed towards your infrastructure. Conversely, 'Outbound' refers to calls your infrastructure sends to the Elastic SIP Trunking Service.   Matching criteria   The final step in creating a rule is to specify the matching criteria.   All Numbers – This criterion should be used if you wish to completely shut down all calls in a specific direction. For example, an inbound call center may want to disallow outbound calling completely. Numbers beginning with – This option will allow you to create a criterion which will block numbers beginning with the match pattern. For example, you could disallow calling to a specific NPA (Numbering Plan Area). Additionally, you can specify a full phone number here to block a specific individual from calling your numbers. Country – This criterion will allow you to specify a country. If applied to outbound calls, this will completely disallow calls to be made to that country. This rule will take precedence over your Country Permissions entry for that country.   ... View more

Currently, Elastic SIP Trunking is only available for North American business entities and can only be used with North American phone numbers. ... View more

Robocalls and spoofed caller IDs have become a massive issue. To fight back, the FCC introduced the STIR/SHAKEN framework — a system that lets carriers verify that a call really comes from who it says it does. What do you need to do to ensure your traffic is being signed correctly? If you are an enterprise customer, meaning a customer who is using their service just for their own business communications, you may choose to do nothing, as Sinch will sign calls on your behalf.   However, if you are a service provider, meaning you provide telecom services to other companies, then you are required by the FCC and the TRACED act to sign your own calls.   Enterprise Customer Responsibilities   Enterprises can choose to sign their own calls. When doing so, Sinch will ensure the signing and attestation remains intact as it transits our network. When Enterprises send unsigned calls from Sinch provided numbers (or numbers you have ported to Sinch), Sinch will sign them with an “A” attestation, the highest level of trust possible. Enterprises sending unsigned calls from non-Sinch numbers, will be signed with a “B”. This indicates that while we know who the Enterprise is, we have no knowledge about the calling party number.   Service Providers - You’re Required to Sign Your Own Calls   The FCC requires all voice service providers — including gateway providers and intermediate providers — to sign all outbound IP-based calls using your own STIR/SHAKEN certificate. You should have your own SPC token (from the STI-PA), certificate, and ability to sign calls. Like the enterprises above, any calls that have been signed prior to reaching Sinch will remain as they transit through the Sinch network.   By the same FCC rules, Sinch, as an intermediate provider, is legally required to sign any unsigned calls we receive. Unsigned calls from service providers on EST will be signed with a “C” attestation , meaning we don’t know or can’t vouch for the caller ID, which may increase the likelihood of calls being labeled as “Likley SPAM”.   Have Questions?   We recommend discussing your STIR/SHAKEN strategy with a telecom regulatory advisor or one of the approved certification authorities — but if you have questions about how Sinch handles unsigned calls or would like to talk about Sinch Hosting your certificate on our infrastructure, we’re happy to help. ... View more

Elastic SIP Trunks can be configured with Forwarding / Fallback. Once set, if a call to one of the telephone numbers associated with a trunk cannot be completed over any of your SIP Endpoints (or in the event you have disabled/removed them) the call will immediately forward out to the number you have configured. To forward all calls simply create a trunk without any SIP endpoints, associate phone number(s) and turn on Forwarding. This option can be configured under the Inbound Call Settings on your trunk.     Additionally, once this is enabled the route will display on your SIP Endpoints section.  This provides a visible reminder that it is enabled and will be used as the route of last resort for your telephone numbers.   ... View more