hi

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

import inspect from typing import Dict, Any, Optional from google.protobuf.duration_pb2 import Duration from livekit import api from livekit.agents import function_tool from livekit.agents.llm import ToolError from .utils import get_sip_number def build_warm_transfer_tool( agent_ref, transfer_config: Dict[str, Any], transfer_context: Dict[str, Any], call_metadata: Optional[Dict[str, Any]] = None, ): """Build a warm transfer tool - AI speaks to agent first before connecting caller.""" routing_type = transfer_config.get("routing_type", "static") if routing_type == "static": return _build_static_warm_transfer(agent_ref, transfer_config, transfer_context, call_metadata) return _build_dynamic_warm_transfer(agent_ref, transfer_config, transfer_context, call_metadata) def _get_caller_id(config: Dict[str, Any], call_metadata: Optional[Dict[str, Any]]) -> Optional[str]: """Determine caller ID - use agent's number or preserve caller's number.""" if config.get("caller_id") == "agent": return get_sip_number(config, call_metadata) return call_metadata.get("from_number") if call_metadata else None def _get_handoff_message(config: Dict[str, Any]) -> str: """Get message to tell caller before transferring.""" message = config.get("handoff_message", "I'm connecting you to a human agent now.") if config.get("handoff_message_type") == "prompt": return f"I'll transfer you to a human agent. {message}" return message def _build_context_summary(call_metadata: Optional[Dict[str, Any]]) -> str: """Build context summary for human agent.""" if not call_metadata: return "" parts = [] if call_metadata.get("from_number"): parts.append(f"Caller: {call_metadata['from_number']}") if call_metadata.get("direction"): parts.append(f"Call: {call_metadata['direction']}") for key, value in call_metadata.get("metadata", {}).items(): if value and str(value).strip(): parts.append(f"{key}: {value}") return "Caller info: " + ", ".join(parts) if parts else "" def _build_static_warm_transfer(agent_ref, config, transfer_context, call_metadata): """Build warm transfer tool for static routing.""" destination = config.get("transfer_to", "").strip() base_desc = config.get("description", "Transfer the call to a human agent") description = f"{base_desc}. This is a WARM transfer - you will speak to the agent first. Destination: {destination}" async def transfer_call(): ctx = transfer_context.get("ctx") room_name = transfer_context.get("room_name", "") sip_trunk_id = transfer_context.get("sip_trunk_id", "") dest = destination.removeprefix("tel:").removeprefix("sip:") if not dest: raise ToolError("No transfer destination configured") if not sip_trunk_id: raise ToolError("Transfer not available: no outbound SIP trunk configured") caller_id = _get_caller_id(config, call_metadata) handoff_msg = _get_handoff_message(config) # Step 1: Tell caller we're transferring await agent_ref.session.say(handoff_msg, allow_interruptions=False) # Step 2: Dial the human agent and WAIT for them to answer try: await ctx.api.sip.create_sip_participant( api.CreateSIPParticipantRequest( sip_trunk_id=sip_trunk_id, sip_call_to=dest, room_name=room_name, participant_identity="human-agent", wait_until_answered=True, # This waits for agent to pick up play_dialtone=True, ringing_timeout=Duration(seconds=30), sip_number=caller_id, ) ) except Exception as e: raise ToolError(f"Failed to reach human agent: {e}") from e # Step 3: Agent has answered - introduce yourself and explain await agent_ref.session.say( "Hello! I'm an AI assistant handling a call. Let me quickly brief you on this caller.", allow_interruptions=False ) # Step 4: Provide context context = _build_context_summary(call_metadata) if context: await agent_ref.session.say(context, allow_interruptions=True) # Step 5: Hand off and exit await agent_ref.session.say( "Connecting the caller to you now. They'll be on the line shortly.", allow_interruptions=False ) agent_ref.session.shutdown() transfer_call.name = "transfer_call" transfer_call.doc = description transfer_call.signature = inspect.Signature([]) transfer_call.annotations = {} return function_tool(transfer_call) def _build_dynamic_warm_transfer(agent_ref, config, transfer_context, call_metadata): """Build warm transfer tool for dynamic routing.""" base_desc = config.get("description", "Transfer the call to a human agent") rules = config.get("dynamic_routing_rules", "") description = f"{base_desc}. This is a WARM transfer - speak to agent first. {rules}" parameters = [ inspect.Parameter( name="transfer_to", kind=inspect.Parameter.KEYWORD_ONLY, annotation=str, default=inspect.Parameter.empty, ) ] async def transfer_call(*, transfer_to: str): ctx = transfer_context.get("ctx") room_name = transfer_context.get("room_name", "") sip_trunk_id = transfer_context.get("sip_trunk_id", "") dest = transfer_to.strip().removeprefix("tel:").removeprefix("sip:") if not dest: raise ToolError("No transfer destination provided") if not sip_trunk_id: raise ToolError("Transfer not available: no outbound SIP trunk configured") caller_id = _get_caller_id(config, call_metadata) handoff_msg = _get_handoff_message(config) # Step 1: Tell caller we're transferring await agent_ref.session.say(handoff_msg, allow_interruptions=False) # Step 2: Dial the human agent and WAIT for them to answer try: await ctx.api.sip.create_sip_participant( api.CreateSIPParticipantRequest( sip_trunk_id=sip_trunk_id, sip_call_to=dest, room_name=room_name, participant_identity="human-agent", wait_until_answered=True, # This waits for agent to pick up play_dialtone=True, ringing_timeout=Duration(seconds=30), sip_number=caller_id, ) ) except Exception as e: raise ToolError(f"Failed to reach human agent: {e}") from e # Step 3: Agent has answered - introduce yourself and explain await agent_ref.session.say( "Hello! I'm an AI assistant handling a call. Let me quickly brief you on this caller.", allow_interruptions=False ) # Step 4: Provide context context = _build_context_summary(call_metadata) if context: await agent_ref.session.say(context, allow_interruptions=True) # Step 5: Hand off and exit await agent_ref.session.say( "Connecting the caller to you now. They'll be on the line shortly.", allow_interruptions=False ) agent_ref.session.shutdown() transfer_call.name = "transfer_call" transfer_call.doc = description transfer_call.signature = inspect.Signature(parameters) transfer_call.annotations = {"transfer_to": str} return function_tool(transfer_call) this wamt trnafer is not wrkingagain same thing agent callaed to the name for eaan trnad and sain i have conned to to huma then godby

Thanks — I see you pasted your warm-transfer implementation and that it’s not working: the AI dials the agent, says it has connected the caller to a human, then immediately says goodbye (so the handoff never actually completes). I can help troubleshoot this — a few targeted questions and checks will quickly narrow the cause. Please answer or check the following items: 1. SIP trunk / provider behavior • Do you have an outbound SIP trunk configured in

transfer_context["sip_trunk_id"]

? You already check this in code, but confirm it’s set and valid. • Does your SIP provider allow caller-ID passthrough for the

sip_number

you send? If the trunk rejects the

sip_number

, the SIP leg may fail even though the agent’s phone rings. 1. Observe SIP participant events and logs • Can you share logs or events for the SIP participant created by

ctx.api.sip.create_sip_participant

? Look for

sip.callStatus

changes (dialing → ringing → active → hangup). If it goes to

hangup

right after

active

, that indicates the SIP leg dropped quickly. • Any exception trace from the try/except around

create_sip_participant

? Your code wraps it and raises ToolError; include the original exception text if available. 1. Confirm wait-until-answered behavior • You use

wait_until_answered=True

. Does the human agent actually press accept on their phone and stay on the line (not auto-reject)? Some phones/softclients auto-answer or immediately send a transfer response—confirm the agent’s endpoint behavior. • If possible, test with

wait_until_answered=False

to see if behavior differs (this can reveal whether waiting is causing a race). 1. Verify agent session lifecycle • After the SIP participant answers, you call

agent_ref.session.say(...)

and then

agent_ref.session.shutdown()

. Confirm that

agent_ref.session.shutdown()

is not triggered prematurely (for example, if the session thinks the agent handoff was already completed). Do you see any session shutdown events right after the introduction phrases? • Try temporarily removing or delaying

session.shutdown()

to observe whether the agent connection remains. 1. Caller audio / hold-state • Is the caller put on hold or hearing hold audio while you dial the agent? If the caller I/O is disabled or muted, agent may think caller disconnected. Confirm the caller remains connected to the room during the warm transfer steps. 1. Destination format and stripping • You strip prefixes with

.removeprefix("tel:").removeprefix("sip:")

. Confirm

dest

is a valid e164 or SIP URI after stripping. Print/log

dest

before calling

create_sip_participant

to verify. 1. Quick tests you can run now • Enable more verbose logging around the SIP create call (print the request and the resulting participant ID/status). • Try calling a test number you control and watch SIP traces on the provider side to see if INVITE/200/ACK occur and if the provider sends a BYE. • Temporarily replace the SIP dial with a non-SIP participant (if supported) to isolate whether the issue is SIP-specific or your session tooling. If you can, please paste: • Any exception text or stack trace from the ToolError when the transfer fails. • Recent SIP participant events showing

sip.callStatus

. • The values for

sip_trunk_id

,

dest

(masked if sensitive), and whether

caller_id

was set or left None. What I can help with next • Interpret SIP participant events and the exception trace you share. • Suggest small code changes (examples) to add logging or to adjust when

shutdown()

runs. • If the trunk is rejecting caller ID, suggest alternatives (use trunk-verified caller or notify human agent and then use SIP REFER transfer). Send the logs/events and answers to the checks above and I’ll walk you through the next debugging steps. Sources: WarmTransferTask | LiveKit Documentation | Preserving Caller ID During SIP Transfers in LiveKit Agents | Agents telephony integration | LiveKit Documentation | Company directory phone assistant | LiveKit Documentation | Job lifecycle | LiveKit Documentation, with interactive elements

{ "message": "Agent functions retrieved successfully", "data": [ { "id": 36, "name": "transfer_call", "description": "Transfer the call to a human agent", "endpoint": "internal://transfer", "method": "POST", "headers": null, "query_params": null, "transfer_config": { "caller_id": "agent", "transfer_to": "+16193892389", "routing_type": "static", "transfer_type": "warm", "handoff_message": "The AI will determine the correct number based on these rules.\n\n", "transfer_method": "invite", "handoff_message_type": "prompt" }, "is_active": true, "timeout_seconds": 30, "speech_during_config": null, "speech_after_enabled": true, "end_instructions": null, "agent_id": 32, "client_id": 2, "created_at": "2026-03-22T114549.255Z", "updated_at": "2026-03-22T121548.965Z", "created_by": 2, "updated_by": 2 } ] }

Thanks — this is helpful. I can see your active transfer configuration (static warm transfer to +1 619‑389‑2389, caller_id set to "agent", 30s timeout). You're seeing the AI dial the human, announce the handoff, then immediately say goodbye instead of completing the handoff. Next steps — please check/confirm the following so I can pinpoint the cause and give a concrete fix: 1) SIP trunk acceptance and provider behavior • Confirm

transfer_context["sip_trunk_id"]

is set and points to the outbound trunk used for this transfer. • Does your SIP provider allow the trunk to present the agent’s number as

sip_number

? Some providers block unverified caller IDs and will fail the outbound call even though it rings briefly. If you’re not sure, try a test where

sip_number

is left blank or set to a verified number. 2) SIP participant events / status • Please share the SIP participant events or logs around the transfer attempt (especially

sip.callStatus

values). Look for the sequence: dialing → ringing → active → hangup. If you see

hangup

immediately after

active

, the agent leg is likely dropping. • If you have an exception stack trace from the try/except that raises

ToolError("Failed to reach human agent: ...")

, paste it here. 3) Verify dest and caller_id values used at runtime • Log the

dest

(after removeprefix) and

caller_id

immediately before calling

create_sip_participant

. Confirm

dest

is a valid e164 number (e.g.,

+1619...

) or SIP URI and not empty. • From your config,

transfer_to

is "tel:+1619...", so dest should be correct, but please confirm the actual runtime string. 4) Confirm agent endpoint behavior • Does the human agent actually accept the call and stay connected? Some endpoints auto-answer and then immediately close or send a transfer response that your flow treats as completion. • Try calling a phone you control to reproduce and watch if you can stay on the line. 5) session.shutdown() timing • You call

agent_ref.session.shutdown()

after announcing "Connecting the caller to you now." If the session believes the transfer already completed (or if there’s a race), shutdown may run too early. • As a quick test, comment out or delay

session.shutdown()

(e.g., sleep 10–15s) to see if the caller actually connects and the agent remains on the line. 6) Additional quick tests • Temporarily set

wait_until_answered=False

to see whether behavior changes (this can reveal if waiting is causing a timing issue). • Try using the LiveKit

WarmTransferTask

(if available in your stack) which handles hold music, separate human agent room, and connectors — it implements a proven handoff flow and can help isolate whether the problem is your orchestration or SIP leg behavior. What to paste here for fast diagnosis • Any exception text from the raised ToolError. • SIP participant event timeline showing

sip.callStatus

. • The runtime values (or logs) for

sip_trunk_id

,

dest

, and

caller_id

(mask sensitive digits if needed). If you prefer, I can suggest small code changes to add logging and to change the shutdown flow so we can observe where the handoff drops. Which logs can you share now? , with interactive elements