Developer Platform, APIs, & Integrations

## SetupWe use `ringcentral-softphone` to answer inbound calls that are routed through a single RingCentral **call queue** to one of several **agent softphones**.- **4 softphones** sit behind one queue, each a distinct RC user/extension with its own SIP credentials.## What we're trying to doFor every call we need to capture two kinds of data and store them under **one phone call record**, then display them together for a single call in our own UI:1. **Internal logs** (from the Node softphone process): a small set of details we write to our database when the call starts, against a unique identifier.2. **Telephony lifecycle**: the full call history — transfer → answered/rejected → transferred again → ended — which we obtain from an **account-level `telephony/sessions` webhook subscription** (or any better approach you'd recommend).**The core question: how can we tie these two records together with the same identifier?**## The problemFrom the SIP connection in `ringcentral-softphone(-ts)`

I have tried to access the API to no avail and this is getting extremely frustrating now.We have RingSense licenses, user’s are set to record conversations and produce transcripts.I can successfully authenticate to the /insights API:https://platform.ringcentral.com/ai/ringsense/v1/public/accounts/~/domains/pbx/records/3129356…./insightsERROR: {    "errorCode": "InsufficientPermissions",    "message": "In order to call this API endpoint, application needs to have [RingSense] permission",    "errors": [        {            "errorCode": "CMN-401",            "message": "In order to call this API endpoint, application needs to have [RingSense] permission",            "permissionName": "RingSense"        }    ],    "permissionName": "RingSense"} The Developer playground “Try It” … DOES NOT WORK.The API documentation is vague.We’ve tried using Postman to connect, producing the above error.We’ve tried using Zapier to get the “ProcessedRecording” event, which does not work.Searching this commu

I already tried submitting a ticket for this, but got told to contact dev support. I'm not a dev, just an admin. So I'm going to try here before jumping into that mess. For some reason there is little online regarding my issue.I have recently migrated to the new RC SSO webapp that included SCIM integration for auto provisioning. So far it is working well, but it is having some strange issues with cell phone numbers and countries. I have not been able to nail down exactly what is going wrong, and I'm starting to wonder if the default mappings logic is not sufficient. I figured I would loop you guys in since I'm having trouble finding much documentation online so far.To be clear, our integration is fully configured and is running in update/create mode. We are syncing 42 users currently without issue. The rest(mostly outside the USA) are failing due to country field issues:SystemForCrossDomainIdentityManagementServiceIncompatibleStatusCode: BadRequest Message: Processing o

I have an application where I’d like to register a webhook for every logged in user (by extensionId). The most sensible way to do this is to have a subscription for each extensionId. However, I’m worried by this piece of documentation “Subscriptions are currently limited to 30 subscriptions per user/extension (for particular application).”Does this mean I can have up to 30 subscriptions per extensionId, and thus my current use case is no problem? Or is it 30 subscriptions per “user” aka per JWT? I notice that the ListSubscription documentation explicitly states “maxItems:30”, which would suggest the limit is not per extension.If I have 85 call center representatives who are signed in, how do I make sure I receive webhook call backs for all of them? Thanks for your help! 

Hello,I’ve been using the Multi-Site IVR Settings API to fetch our sites' IVR settings, and it has been working great.However, the API documentation states:Internal identifier of a site. Please note that siteId cannot take the main-site value.Is there any way to retrieve this information through another API call?Our Main Site IVR settings are used as the defaults for many of our IVR menus, but there currently doesn't seem to be a way for us to retrieve the default values for No Input, #, or * from the Main Site.Thank you. The data exists under Company > Sites > Main Site > Company Call Handling > Default IVR settings page 

We have an application in the Developer Portal which was originally created by one of our users.How does ownership of Developer Portal applications works?Specifically:Is the application owned by the RingCentral organisation or by the user who created it? If the creator’s RingCentral account is deleted in the future, will the application continue to function and remain accessible to other developers within the organisation?Main concern is if in 18 months or so the user who created the app leaves, we delete their account not realising we’re also deleting access to the app they made.  Thanks.  

I am trying to mass download all call recordings for my company. I am using this endpoint:https://developer.niceincontact.com/API/MediaPlaybackAPI#/RecordingMediaPlayback/getACDContactRecordingDataIf I provide an acd-call-id/master contact id I get from the admin portal I am able to download the recording. I need to do this for all our contacts. Is there an endpoint to list all contacts with the acd-call-id/master contact id, or a way to export a list from the admin panel? How do I get all contact ids?Thanks!

Hello,I am able to find the all of the DeviceID's assigned to an extension by calling this endpoint:https://platform.devtest.ringcentral.com/restapi/v1.0/account/accountId/extension/extensionId/deviceI can then use these DeviceID's to place a call using my chosen device by calling this endpoint and supplying the DeviceID:https://platform.devtest.ringcentral.com/restapi/v1.0/account/accountId/telephony/call-outMy question:How do I find the DeviceID of the RingCentral App?https://www.ringcentral.co.uk/desktop-app.htmlThe /extension/~/extensionId/device EndPoint only returns the ID's for Softphones and HardphonesThank you

I'm building an application where an operator may need to place a 911 call on behalf of someone at a location that:Is different from where the operator is physically located Changes dynamically at runtime (i.e. it is not known in advance)I understand that E911 is designed to address dispatchable location requirements for VoIP calls, but I'm struggling to find a clear, reliable API approach for this scenario. My key concern is timing: even if I update the registered E911 location via API immediately before the call is placed, I cannot guarantee that the updated location will be propagated to the PSAP before the call is answered. Can you please point me to the best practice and API set to use in this scenario ? thank you in advance. 

Hi everyone,I have a website integrated with RingCentral and would like to implement an AI note-taker feature within the pop up in our website.My goal is to:- Add a toggle button (Start/Stop AI Notetaker) during a call we opened that pop up-I am storing the call detail along with I need to store the notes as wellIs this feasible? if so kindly guide me to implement the featuretell me how to implement in step by step Thank you

Hi RingCentral community,I’m integrating the RingCentral embeddable/web phone widget into our website and using widget postMessage events to save call history.Issue:For an inbound call, if the call popup is open and the user answers from the mobile, I usually receive rc-active-call-notify with:data.call.telephonyStatus === "OnHold" But if the same inbound call is answered from the RingCentral mobile app, the browser widget sometimes receives rc-call-end-notify while:data.call.telephonyStatus is blank/missing or not reliable, and my stored value from rc-active-call-notify is also empty. Because of this, our service can incorrectly classify the call as missed even though it was answered on mobile.Questions:Is rc-active-call-notify.data.call.telephonyStatus expected to be reliable for calls answered on another device, such as RingCentral mobile? Which widget event should be used to know the final disposition of an inbound call? Does rc-call-end-notify.data.call include a stable ID, like t

Current approach:We were relying on rc-active-call-notify and storing:case 'rc-active-call-notify': ringCentralPhoneWidget.latestActiveCallTelephonyStatus = data.call && data.call.telephonyStatus ? data.call.telephonyStatus : "";break;Then during rc-call-end-notify, we determine whether the call should be saved as missed/incoming.Problem observed:In several debugging scenarios:Incoming call arrives rc-active-call-notify fires with: telephonyStatus = "Ringing" User answers the call on mobile or RC desktop Instead of receiving another rc-active-call-notify with a connected state, we directly receive: rc-call-end-notify At this point: latestActiveCallTelephonyStatus is still "Ringing" or sometimes null UI answer button may still be visible CRM incorrectly logs the call as missed We also noticed behavior changes while debugging:With breakpoints/debugger enabled, sometimes the correct event arrives later Without debugger, the issue reproduces mo

I have a TAPI application that makes my phone accessible to someone with low to no vision. This TAPI application allows me to control the phone via keyboard. A requirement of the TAPI application is that it needs a TAPI driver. Does ring central have a TAPI driver that I can use so this TAPI application - accessaphone - can control my ring central phone?

Hi, Completely new to webhooks but had hit a issue with my code. I have set up a webhook that sends a notification that a new call is presented (This works fine)I then have another Webhook that will tell me the status, the callers number and Handled by, The handled by is where I’m having an issue. I have successfully had #agent_username# work but only want to show agent full name.Ive tried #agent_full_name# and #agent_first_name# #agent_last_name# Does anyone have any suggest or a list of values that can be used. Code below: {  "cardsV2": [    {      "cardId": "alert-001",      "card": {        "sections": [          {            "widgets": [              {                "decoratedText": {                  "topLabel": "Alert Status",                  "text": "<b>Call Answered</b>",                  "startIcon": { "knownIcon": "PHONE" }                }              },              {                "decoratedText": {                  "topLabel": "Caller Number",           

I am needing assistance trying to setup a 3rd party SIP on UnifiTalk - for our Ring Central Account. UnifiTalk has the ability to add custom SIP provider but I am not sure what the specific key are.I have the details but not sure what fields to create on the Unifi side. SIP Domain              sip.ringcentral.com:5060Remote SIP port         5060Local SIP port          5060Outbound Proxy          Please select outbound proxy according to the location of your deviceOutbound Proxy Port     5096User Name               1########### (phone)Password                CPwdeeeeAuthorization ID        123425566 I am needing these converted to I guess what is expected by RingCentral. Is this documented anywhere????<SIP_domain><Remote_SIP_port><VoIP_Registrations><Protocol>SIP</Protocol><RegServer>sip.ringcentral.com</RegServer><RegClient>19546031388@sip.ringcentral.com</RegClient><LocalAlias>19546031388@99.27.55.122</LocalAlias> 

RingCentral APIs are designed to be used and tested directly in the production environment. There is no technical requirement for a separate sandbox or test environment in order to explore or integrate with the APIs.If you'd prefer to isolate development activity, using a separate developer account is an option. Whether or not to do this is entirely up to your personal preference or your organization's policies. Some developers choose to create separate accounts for development and production use cases, while others use a single account for both.This streamlined approach allows for faster iteration and eliminates the complexity and requirement of managing different environments for testing and production, unless you want to. Historical NoteIn December of 2024, RingCentral retired its developer sandbox environment. This change greatly simplified the process of producing an app. As a result of this change, the legacy concept of "app graduation" was retired.

Welcome to the home of the RingCentral developer community, where developers and users come together to ask questions, seek support, and work together to get the most out of RingCentral's unified communication platform.This article is provided to developers going through the getting started experience accessible from within the RingCentral Developer Console and needing help specifically with the "Initiate a phone call" code sample.I am receiving an error when I run my code sampleThe code sample provided for you is 100% complete and ready to go. You should not need to edit it in any way.However, if you are still receiving an error, check the following: Make sure you installed the SDK. If the error message relates to a missing library or a file that cannot be found, return to Step 2 on the Making a Simple API Call page and follow the instructions for your preferred programming language to properly install the RingCentral SDK. I am experiencing a different problemIf the above content di

Hi all, I've been working on a fairly standard "forward inbound SMS to an external system" integration and have hit a wall I can't find documented anywhere. Looking for guidance.SetupRingCentral account with 3 extensions total (2 user extensions (x101, x106) and 1 call queue, main account/auto-receptionist (x111)) Business SMS Booster license active, assigned to x101 A TCR-registered, SMS-enabled phone number is assigned as the direct number of a call queue extension x111 (company number, no user assigned) The call queue's SMS Recipients is set to a user extension (x101) x101 is also a call queue manager Auto-Receptionist > Call/Fax/SMS Recipients also lists User (x101) as the SMS Recipient, x111 as the operator, x111 as the Fax recipient JWT-authenticated as a super admin with x101’s account (super admin) App permissions: Read Accounts, Read Messages, SMS, Webhook SubscriptionsWhat I builtA webhook subscription using the standard event filter, with a small middleware to validate th

Hi RingCentral Team,I am implementing SMS functionality using RingCentral REST API with JWT authentication (server-to-server app).Environment:JWT authentication flow Existing number available under Admin → Phone System → Phone Numbers Number assigned to a Softphone extensionIssue:When sending SMS via API, I receive:“Phone number doesn't belong to extension”This happens even though the phone number exists in the account and is active.Questions:Is SMS sending restricted to phone numbers assigned only to the authenticated JWT extension? Can existing numbers be reused across multiple extensions for SMS sending? What is the correct way to bind or select an existing phone number as a valid SMS sender for JWT-based API usage? Is there any configuration required to enable this use case without purchasing a new number?Goal:Use an existing company phone number as the SMS sender for OTP via API without creating a new number or extension.Thanks in advance for the clarification.

Hi everyone,I’m working on a custom integration using RingCentral and would like to implement an AI note-taker feature within the RingCentral dialer.My goal is to:- Add a toggle button (Start/Stop AI Notetaker) during a call- Capture the call recording and/or transcript- Generate AI-based notes (summary, key points, action items)- Automatically save these notes into a CRM history recordI would like guidance on the best approach for this:1. Does RingCentral provide APIs for real-time or post-call transcription and AI summaries?2. Is it recommended to use RingCentral’s native AI capabilities, or integrate external services for transcription and summarization?3. What is the best way to hook into call events (start/end) to trigger this workflow?4. Are there any sample implementations or best practices for syncing AI-generated notes to a CRM?Any insights, documentation, or examples would be greatly appreciated.Thanks in advance!

Hi everyone,I’m implementing merge in a Laravel app and need confirmation on expected behavior.StackPHP SDK: ringcentral/ringcentral-php 3.0.4 WebPhone SDK (ringcentral-web-phone-2) Using call-control + telephony conference APIsFlowCreate conference POST /restapi/v1.0/account/~/telephony/conference → 200 Bring in active calls into conference POST /restapi/v1.0/account/~/telephony/sessions/{conferenceSessionId}/parties/bring-in → 403 TAS-106Observed app result/api/r/call-control/merge returns 409 Conflict Message: Bring-in was rejected (TAS-106). Automatic transfer fallback to conference target is disabled.Sample payload that still fails (even with -1 parties)primary_party_id: ...-1 secondary_party_id: ...-1 Still gets TAS-106: Operation is not allowedScopesCallControl, ReadPresence, VoipCalling, and other standard scopes are present.QuestionHas anyone seen TAS-106 for bring-in/merge while conference creation works?Is this usually account/role/call-leg restriction, and what exact settin

Using the “get Company Call Records” and not seeing recipientextensionNumber.  We are attempting to filter calls that route through the AutoReceptionist, then through a user with a specific firstName description.  We want to then “marry” all call legs associated with that single call and then take that session ID and pull that call log into a separate bucket.   We have many different sites in our RingCentral tenant that all have different work hours, so we want to “tag” a call by using a User that has AFTERHOURS in its firstName. Very new to the API world... 

We are using `ringcentral-softphone-ts` as the monitoring softphone in a RingCentral call supervision setup.Our flow is:1. Receive telephony session notifications for monitored agents2. When party status becomes `Answered`, call:   `POST /restapi/v1.0/account/~/telephony/sessions/{sessionId}/parties/{partyId}/supervise`3. Wait for the monitoring SIP invite in `softphone.on('invite', ...)`4. Answer the invite5. Start consuming `audioPacket` and stream audio to transcriptionThis works, but we consistently lose the first ~3-4 seconds of the call transcript.Observed behavior:- The first transcript segment always starts a few seconds after the actual human conversation begins- The delay appears to happen before audio packets begin arriving, i.e. between `Answered` -> `/supervise` -> monitoring invite -> audio startQuestions:1. Is `Answered` the earliest supported moment to call `/supervise` for call monitoring?2. Can `/supervise` be called earlier, such as during `Setup` or `Procee

looking for a developer to allow us to bulk load numbers into the Caller ID field when creating a rule?

Does anyone have any knowledge/experience with integrating RingCentral with ITSM tools?We use Hornbill ITSM which doesn’t have any out-of-the-box integration connectors, but I was hoping we could maybe use APIs to have RingCentral record a call and pass parameters into Hornbill via an API call and have it log an incident/service request in the system?