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Thanks for reaching out, Kevin. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

Each tenant has their own subaccount. They buy a phone number and its saved in their subaccounts. The flow is the following: User buys phone -> assigns ai agent to phone -> end user calls -> hits our webhook ->livekit takes over -> agent experience comence -> user asks ai agent to transfer -> ai agent transfer to another phone number

Thanks — I’m responding to your flow where each tenant has a Twilio subaccount + number → calls hit your webhook → LiveKit handles the call → AI agent requests a transfer to another phone number. This is doable without replacing your webhook or assigning a SIP trunk. A reliable approach is to keep your webhook returning TwiML that connects the inbound PSTN caller into a LiveKit room, then have the AI agent trigger LiveKit’s SIP transfer API to forward the PSTN participant to the target phone number. Quick checklist and questions so I can give exact implementation snippets: • Confirmations I need a. Are you currently returning TwiML with <connect><room> (or similar) from your webhook to put the caller into a LiveKit room? b. Do you have “PSTN Transfer” enabled for the Twilio trunk/settings in each tenant subaccount? (This is required to perform SIP REFER style transfers.) c. Do you need a cold transfer (caller is moved and agent drops) or a warm transfer (3-way/consult then release)? d. Do you need the transferred call to preserve the original caller ID, or is showing the tenant’s Twilio number acceptable? • Implementation sketch (high level) ◦ Keep webhook as-is returning TwiML to connect caller into LiveKit room. ◦ Ensure the incoming PSTN participant is represented as a SIP/PSTN participant in the LiveKit room. ◦ When the AI agent requests transfer, call LiveKit’s TransferSIPParticipant API with: ▪︎

room_name

▪︎ PSTN participant identity ▪︎

transfer_to

set to the E.164 phone number (e.g.,

tel:+15105550123

) ◦ Handle failures by falling back to initiating a new outbound call from Twilio and bridging (if needed). ◦ Include tenant ID/room names to enforce isolation across subaccounts. • Notes / limitations ◦ Warm transfers are not supported the same way as cold SIP REFER transfers; confirm if you need warm transfers. ◦ Preserving caller ID depends on Twilio/account settings and number verification—it may not always be possible. If you confirm items 1–4 above (or tell me which you want), I’ll provide concrete TwiML and LiveKit API request examples tailored to your language/stack and a recommended failure fallback flow. Sources: Accepting calls overview | LiveKit docs | Agents telephony integration | LiveKit docs | Preserving Caller ID During SIP Transfers in LiveKit Agents | sip | livekit/livekit_composite | DeepWiki

we have everything set up, the issue is when the transfer requests comes in, it uses the livekit outbound trunk which causes 403 errors...

Thanks — I’m responding to the 403 error that occurs because transfer requests use the LiveKit outbound trunk. To diagnose and fix this quickly, please answer these and share small details/logs where possible: 1. How is the transfer API being called? ◦ Are you calling LiveKit’s

TransferSIPParticipant

(server SDK) from your backend, or making another API call via Twilio? 1. Which credentials/token are used for that request? ◦ Is the request authenticated with your LiveKit API key/secret (server key) or a short-lived client token? 403 often means the token lacks server-side scope. 1. Outbound trunk config ◦ Is there an outbound trunk configured in LiveKit per-tenant, or a shared outbound trunk? Are Twilio credentials configured on that trunk and valid? 1. Exact 403 response ◦ Paste the full error message or response body (status code, error string). If there’s a request ID or timestamp include that. 1. Caller/participant identity used ◦ Which participant identity is passed into the transfer call (e.g.,

PSTN-<number>

)? Confirm it matches the participant currently in the room. 1. Tenant mapping ◦ Are you ensuring the transfer request uses the tenant’s trunk/credentials (subaccount) rather than a different tenant’s trunk? Quick checks you can run now • Ensure the transfer request uses a server-side LiveKit credential with the Transfer scope (not a client token). • Confirm the room and participant exist and participant kind is SIP/PSTN when calling transfer. • Verify the outbound trunk’s Twilio credentials are valid and allowed to dial the

transfer_to

number. • Check LiveKit server logs for a request ID near the 403 to see policy/auth failure details. Answer the questions and paste the 403 response (sanitized if needed). I’ll provide targeted steps and example server-side request code to correct the authentication/trunk routing.

we have only 1 shared outbound trunk on live kit and: [dotenv@17.2.3] injecting env (0) from secrets.env -- tip: ⚙️ override existing env vars with { override: true } 🔥 Prewarming LinkAI agent... ✅ VAD model loaded 🚀 Starting LinkAI Voice Agent (LiveKit Cloud)... 👤 Participant joined: sip_+17873825481 (964ms) 🔍 Telephony room detected by name pattern: tel-_+17873825481_5VgqKyotAQ2q 📞 Telephony call detected (empty metadata fallback), fetching config for phone: +17875925211 🔍 Room name: tel-_+17873825481_5VgqKyotAQ2q 🔍 Participant identity: sip_+17873825481 🔍 SIP participant attributes: { 'sip.callID': 'SCL_uB72RN5RnmHm', 'sip.callTag': '03923036_c3356d0b_69591eb7-5579-4fc5-ae51-e99a1db78860', 'sip.twilio.accountSid': 'AC63991c2db51a41638adaa40c2fa6f205', 'sip.hostname': 'sip.twilio.com', 'sip.trunkPhoneNumber': '+17875925211', 'sip.ruleID': 'SDR_qz8dwh6mbjgZ', 'sip.callStatus': 'ringing', 'sip.trunkID': 'ST_iq7LukpU7dAf', 'sip.callIDFull': 'ec557d78463c9e0750e0177a6f248a41@0.0.0.0', 'sip.phoneNumber': '+17873825481', 'sip.twilio.callSid': 'CA5ea054a4da894c7e3f90ad68c2196093' } 🔍 Fetching agent config for phone number: +17875925211 ✅ Found chatbot: PR Convention for phone: +17875925211 📚 Loading knowledge profiles for 1 sources... ✅ Loaded 1 knowledge profiles 📞 Loaded call forwarding configuration: { forwardingLinesCount: 1 } ✅ Agent config built successfully for telephony 🔧 Enabled tools: { webSearch: true, calendar: false, email: true, forms: true, callForwarding: true, hasKnowledge: true } ✅ Agent config loaded for phone: PR Convention 🤖 Chatbot ID: cme7jn41o0001d4q5q4gt76dn 🎵 Voice: Charon ⏱️ Config loaded in 4467ms ✅ Agent configuration loaded: PR Convention 🌍 Language Configuration: Primary: en Secondary: es Bilingual Mode: ENABLED ✅ Voice pipeline configured with bilingual support: 🎤 STT: Deepgram nova-3 (multi (Nova-3 native codeswitching: en + es)) 🎵 TTS: ElevenLabs eleven_multilingual_v2 (automatic accent detection) 🔄 Turn Detection: MultilingualModel - Context-aware endpointing ⏱️ STT Endpointing: 100ms (optimized for code-switching per Deepgram docs) ⏱️ Turn Detection Delays: min=200ms, max=600ms (optimized for speed) 🌍 Bilingual Mode: ENABLED 🔍 Tool detection: { hasKnowledge: true, hasWebSearch: true, hasForms: true, hasEmail: true, hasTools: true, webSearchEnabled: true, unrestrictedWebSearchEnabled: false } 🔧 Pre-loading unified tools for VoiceAgent... 📡 Fetching unified tools metadata... { url: 'https://linkai-chat-runtime.fly.dev/tools/voice-format', source: 'Fly.io' } ❌ Failed to load unified tools (513ms): Error: Failed to fetch tools: 500 Internal Server Error. {"error":"Failed to format tools for voice","details":"Failed to import tool modules: ToolRegistry not found"} at loadUnifiedToolsForVoice (file///Users/kevindiaz/LinkAI/voice server/agent.js179:13) at process.processTicksAndRejections (nodeinternal/process/task queues95:5) at async createAssistant (file///Users/kevindiaz/LinkAI/voice server/agent.js447:22) at async Promise.all (index 0) at async entry (file///Users/kevindiaz/LinkAI/voice server/agent.js635:38) at async file:///Users/kevindiaz/LinkAI/voice-server/node_modules/@livekit/agents/dist/telemetry/traces.js7316 at async DynamicTracer.startActiveSpan (file:///Users/kevindiaz/LinkAI/voice-server/node_modules/@livekit/agents/dist/telemetry/traces.js7112) at async file:///Users/kevindiaz/LinkAI/voice-server/node_modules/@livekit/agents/dist/ipc/job_proc_lazy_main.js725 Voice agent will continue with only native tools ⚠️ No tools loaded, falling back to base agent 🔧 Using base Agent without tools ⏱️ Assistant created in 516ms ✅ Background Voice Cancellation (BVC) enabled (agent-side) [203429.748] INFO (72240): Configuring session recording enableRecording: false not authenticated ⏱️ Session started in 18ms 📞 Initializing call termination handlers... ⏰ Setting call timeout: 480 seconds (warning at 470s) 👤 Enabling user presence check: 9 seconds delay 🔇 Enabling silence timeout: 22 seconds 🎧 Setting up unified conversation item listener... 🔍 Session object type: object 🔍 Session has on method: function 🔍 Session methods: [ '_onUserInputTranscribed' ] 🔍 [DEBUG] Available session events: { hasOn: true, sessionType: 'object', sessionKeys: [ '_events', '_eventsCount', '_maxListeners', 'vad', 'stt', 'llm', 'tts', 'turnDetection', 'options', 'agent' ] } 👋 Saying greeting: Hi! Thank you for calling the Puerto Rico Convention Center. How can I help? 🎉 LinkAI Voice Agent operational! (Total startup: 5973ms) [203429.756] INFO (72240): participantValue.trackPublications participantValue: "sip_+17873825481" trackPublications: [ { "info": { "sid": "TR_AMMtj8An6FXfir", "name": "sip_+17873825481", "kind": "KIND_AUDIO", "source": "SOURCE_MICROPHONE", "simulcasted": false, "width": 0, "height": 0, "mimeType": "audio/opus", "muted": false, speech_id: "speech_cd18716c-c8b" [203437.638] INFO (72240): EOU metrics endOfUtteranceDelayMs: 574 transcriptionDelayMs: 572 onUserTurnCompletedDelayMs: 0 💾 Saved user message to database: Can you transfer me to the main office?... [203438.263] INFO (72240): LLM metrics ttftMs: 608 inputTokens: 514 promptCachedTokens: 0 outputTokens: 18 tokensPerSecond: 28.8 ✅ Found outbound trunk for user: { userId: 'cmbcjvgsq0000j5kjb37t1bss', email: 'hello@getlinkai.com', trunkId: 'ST_pjA2faq7vmfn' } 📞 ===== COLD TRANSFER INITIATED ===== Forwarding to: Main Office Phone: +7876417722 Extension: None Trunk ID: ST_pjA2faq7vmfn ⚠️ Phone number appears to be missing country code, adding +1: +7876417722 -> +17876417722 Formatted target number: +17876417722 Found caller participant: { identity: 'sip_+17873825481', name: 'Phone +17873825481' } ℹ️ Trunk information: Inbound trunk (call received through): ST_iq7LukpU7dAf Outbound trunk (transfer will use): ST_pjA2faq7vmfn Using outbound trunk for REFER transfer Sending SIP REFER to transfer call... Transfer details: { roomName: 'tel-_+17873825481_5VgqKyotAQ2q', participantIdentity: 'sip_+17873825481', targetNumber: '+17876417722', outboundTrunkId: 'ST_pjA2faq7vmfn', inboundTrunkId: 'ST_iq7LukpU7dAf', trunkToUse: 'ST_pjA2faq7vmfn', note: 'Transfers require outbound trunk with termination credentials' } 📤 Final transfer request: { roomName: 'tel-_+17873825481_5VgqKyotAQ2q', participantIdentity: 'sip_+17873825481', transferTo: 'tel:+17876417722', options: { playDialtone: false } } ❌ Cold transfer failed: Forbidden: twirp error unknown: sip status: 403: Forbidden at TwirpRpc.request (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/TwirpRPC.js61:13) at process.processTicksAndRejections (nodeinternal/process/task queues95:5) at async SipClient.transferSipParticipant (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/SipClient.js519:5) at async initiateColdTransfer (file///Users/kevindiaz/LinkAI/voice server/src/transfer/coldTransfer.js171:5) { status: 403, code: 'permission_denied', metadata: { error_details: 'CAcSGnNpcCBzdGF0dXM6IDQwMzogRm9yYmlkZGVuGjcKJXR5cGUuZ29vZ2xlYXBpcy5jb20vbGl2ZWtpdC5TSVBTdGF0dXMSDgiTAxIJRm9yYmlkZGVu', sip_status: 'Forbidden', sip_status_code: '403' } } Error details: { message: 'twirp error unknown: sip status: 403: Forbidden', sipErrorCode: '403', sipErrorMessage: 'Forbidden', stack: 'Forbidden: twirp error unknown: sip status: 403: Forbidden\n' + ' at TwirpRpc.request (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/TwirpRPC.js61:13)\n' + ' at process.processTicksAndRejections (nodeinternal/process/task queues95:5)\n' + ' at async SipClient.transferSipParticipant (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/SipClient.js519:5)\n' + ' at async initiateColdTransfer (file///Users/kevindiaz/LinkAI/voice server/src/transfer/coldTransfer.js171:5)', forwardingLine: { id: 'line-1767132856231', name: 'Main Office', extension: '', phoneNumber: '+7876417722', forwardPrompt: 'When user asks to speak with a human.' }, trunkId: 'ST_pjA2faq7vmfn' } ❌ Cold transfer failed: Forbidden: twirp error unknown: sip status: 403: Forbidden at TwirpRpc.request (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/TwirpRPC.js61:13) at process.processTicksAndRejections (nodeinternal/process/task queues95:5) at async SipClient.transferSipParticipant (file///Users/kevindiaz/LinkAI/voice server/node modules/livekit server sdk/dist/SipClient.js519:5) at async initiateColdTransfer (file///Users/kevindiaz/LinkAI/voice server/src/transfer/coldTransfer.js171:5) { status: 403, code: 'permission_denied', metadata: { error_details: 'CAcSGnNpcCBzdGF0dXM6IDQwMzogRm9yYmlkZGVuGjcKJXR5cGUuZ29vZ2xlYXBpcy5jb20vbGl2ZWtpdC5TSVBTdGF0dXMSDgiTAxIJRm9yYmlkZGVu', sip_status: 'Forbidden', sip_status_code: '403' } } 🎙️ Agent speaking - State: CONVERSING 🤖 Agent activity detected - resetting timers [203439.902] INFO (72240): TTS metrics ttfbMs: 889 audioDurationMs: 4000 🔇 Silence detected - starting timers ⏰ Starting presence timer ⏰ Starting presence timer: 9 seconds (waiting for user response) ⏰ Starting silence timer ⏰ Starting silence timer: 22 seconds (absolute user silence) 🎙️ Agent speaking - State: CONVERSING 🤖 Agent activity detected - resetting timers 🎙️ Agent speaking - State: CONVERSING 🤖 Agent activity detected - resetting timers 🎙️ Agent speaking - State: CONVERSING 🤖 Agent activity detected - resetting timers 🎯 CONVERSATION ITEM EVENT FIRED! { role: 'assistant', textContent: 'Sure, I can transfer you to the main office. Pleas', fullLength: 77, hasText: true, timestamp: 1767141283135

Thanks — I’m responding to the 403 Forbidden you get when the agent calls LiveKit’s transfer API using your shared outbound trunk. Likely causes (from the logs) • LiveKit returns SIP 403 Forbidden when sending the REFER through your outbound trunk — this usually means the trunk cannot terminate to the target number (missing/invalid termination credentials or unsupported caller/trunk mapping) or the transfer request was made with a credential that lacks server-side transfer permission. • The logs show the inbound trunk is ST_iq7LukpU7dAf and the outbound trunk chosen is ST_pjA2faq7vmfn. LiveKit notes: “Transfers require outbound trunk with termination credentials.” Actionable checks and fixes — run these and tell me results: 1) Verify outbound trunk termination credentials • Ensure the outbound trunk ST_pjA2faq7vmfn has working termination (SIP provider/Twilio) credentials configured in LiveKit for dialing PSTN numbers. If using Twilio, confirm the trunk’s credential (auth token, account SID, messaging/voice credentials) are valid. 2) Confirm the trunk can terminate to that destination • From the trunk provider (Twilio subaccount), verify the trunk/termination allows calls to the destination country/number (+17876417722). Check any ACL, geo permissions, or balance limitations. 3) Ensure the transfer call uses server-side LiveKit credentials • The Transfer API must be invoked with a server API key/secret (not a short-lived client token). Confirm your backend call to

transferSipParticipant

uses a server credential that has transfer scope. 4) Confirm correct trunk selection mapping • Because you have a shared outbound trunk, ensure LiveKit is permitted to use that trunk for the inbound subaccount’s call. If policies restrict which outbound trunk a given inbound trunk/subaccount may use, configure or map the tenant/subaccount to allow ST_pjA2faq7vmfn for termination. 5) Inspect the full SIP error detail from LiveKit logs • The Twirp metadata contains base64-ish details. If you can paste the raw LiveKit server log around the transfer request (request id/timestamp), I can decode or interpret the exact SIP response body for a more precise cause. 6) Fallback plan (if trunk termination not possible) • If you cannot enable termination on the shared trunk for this destination, fallback to placing a new outbound call from Twilio (or the tenant subaccount) and bridging it into the room, or configure per-tenant outbound trunks with proper credentials. Please confirm: • Which credential your backend uses when calling LiveKit transfer (server API key or client token)? • Whether the outbound trunk ST_pjA2faq7vmfn has Twilio termination credentials configured and verified to dial +17876417722. Send the answers and any raw LiveKit transfer log lines (sanitized request id or timestamps) and I’ll give the exact next steps or the specific config commands/snippets you should apply.