wide-flower-24692
02/25/2026, 8:04 AMtall-airline-16756
02/25/2026, 8:04 AMtall-airline-16756
02/25/2026, 8:05 AMtransfer_sip_participant, and the agent keeps getting` generate_reply timed out waiting for generation_created event`. I’ll walk through the most likely causes and concrete next steps to help narrow and fix each issue.
SIP 403 Forbidden during transfer
• What this indicates: The SIP provider or LiveKit’s SIP handling is rejecting the transfer request with a 403 SIP/permission error. This usually means a credentials, trunk configuration, or permission mismatch rather than a bug in your transfer call code.
• Things to check:
◦ SIP trunk / provider credentials: Verify the credentials the LiveKit SIP gateway uses to authenticate to your SIP provider are correct and active.
◦ Allowed operations/permissions: Confirm the SIP trunk/account is permitted to initiate outbound transfers to the dialed destination (some carriers block transfers or require additional permissions or routing).
◦ Caller ID & UUI behavior: Some trunks require a valid caller ID or specific UUI/headers when performing transfers; ensure any required headers or caller ID settings are set correctly.
◦ Formatting of transfer target: In your code you log the transfer target as "+918956521667". Confirm the actual transfer_to string passed to the API is plain E.164 (e.g., tel:+918956521667) without markdown or extra characters.
◦ Trunk routing / dialplan: Check whether your SIP provider requires the transfer to go through a particular route or gateway—403 can happen when the destination is not routable for that account.
• Immediate diagnostic steps:
a. Capture the SIP INVITE/REFER exchange (SIP logs or pcap) to see the SIP response body and any provider error details.
b. Reproduce a simple outbound call (not a transfer) to the same number from the same trunk to confirm outbound calling works to that destination.
c. Try transferring to a different destination (or internal extension) to see whether only certain destinations are blocked.
d. Confirm the exact transfer_to value being sent in TransferSIPParticipantRequest (print it right before call) to rule out formatting issues.
generate_reply timeout (Realtime LLM generation)
• What this indicates: The realtime reply call is waiting for a generation_created event from the LLM/real-time model and times out. That can happen if the real-time model/service is unreachable, slow to respond, or there’s a messaging/eventing problem between components.
• Things to check:
◦ Model connectivity: Ensure the realtime LLM endpoint is reachable from the agent host and the key/credentials are valid.
◦ Latency / service availability: Confirm whether model requests sometimes succeed; if intermittent, it may be transient latency or rate limits.
◦ Timeouts and retries: Check the configured timeouts in your realtime session code. If the default is too short for the model or network, increase it and add retry/backoff.
◦ Logs around the generation request: Look for earlier logs showing the request to the LLM service and any network or gRPC/Twirp errors.
• Immediate diagnostic steps:
a. Reproduce generate_reply in isolation (if possible) and inspect whether the request is sent and whether any response/error arrives from the LLM service.
b. Increase the reply-generation timeout temporarily to see if longer waits succeed.
c. Check for resource constraints on the agent host (CPU, memory) which could delay handling events.
d. Collect logs of the realtime session component (including any gRPC/Twirp traces) around the timeout.
Recommended next actions (concise)
1. Confirm the exact string passed as transfer_to in TransferSIPParticipantRequest — it must be plain tel:+918956521667 (no markdown or brackets).
2. Capture SIP debug logs/pcap during a transfer attempt and share the SIP responses if you want me to help interpret them.
3. Verify SIP trunk account permissions and outbound dialing rules with your provider; test a direct outbound call to the same destination from the same trunk.
4. For the LLM timeout: increase the generate_reply timeout, reproduce the failure and share the realtime model request/response logs (or confirm whether the model endpoint is reachable).
5. If you want, share the minimal snippets of:
◦ The exact transfer_to value you pass
◦ Any SIP provider error details from SIP logs
◦ The realtime/generation request log lines (before the timeout)
I can then help interpret SIP responses and advise changes to your code/timeouts and headers.
If you'd like, tell me which one you want to tackle first (SIP 403 or generate_reply timeout) and share the transfer_to value plus any SIP provider logs you have — I’ll guide you step‑by‑step.
Sources: Pass UUI data to Genesys Cloud during SIP transfers | LiveKit | Preserving Caller ID During SIP Transfers in LiveKit Agentswide-flower-24692
02/25/2026, 8:46 AMtall-airline-16756
02/25/2026, 8:46 AMgenerate_reply timeout error. I’ll give you a safe, practical version that:
• Ensures the transfer proceeds even if reply generation times out or fails.
• Uses explicit timeout handling for generate_reply.
• Logs and reports errors clearly so you can debug later.
• Sends a clean E.164 transfer_to value to the SIP API.
Before you paste this in: confirm the correct participant identity you should use (in your earlier snippet you used self.participant.identity — the example below uses self.participant.identity but I can adjust if your code stores it elsewhere).
Example function (with timeout handling and robust error flow):
import asyncio
from asyncio import TimeoutError
@function_tool()
async def transfer_call(self, ctx: RunContext):
"""
Inform the user (best-effort) then transfer the SIP participant.
This separates generate_reply from the transfer so timeouts in reply generation
won't block the transfer, and it uses an explicit timeout for generate_reply.
"""
# Use a plain E.164 string for transfer target
transfer_to = "tel:+918956521667"
if not transfer_to:
return "Cannot transfer call: no transfer target specified."
# 1) Try to generate the reply but don't let it block indefinitely
REPLY_TIMEOUT_SECONDS = 8 # adjust higher/lower depending on your model latency
try:
# run generate_reply with an explicit timeout
await asyncio.wait_for(
ctx.session.generate_reply(
instructions="Let the user know you'll be transferring them in the current language."
),
timeout=REPLY_TIMEOUT_SECONDS,
)
<http://logger.info|logger.info>("Spoken notification generated before transfer.")
except TimeoutError:
logger.warning("generate_reply timed out; continuing with transfer.")
except Exception as e:
logger.error(f"Non-critical error generating reply: {e}; continuing with transfer.")
# 2) Proceed with the transfer (critical path)
job_ctx = get_job_context()
try:
await job_ctx.api.sip.transfer_sip_participant(
api.TransferSIPParticipantRequest(
room_name=job_ctx.room.name,
participant_identity=self.participant.identity,
transfer_to=transfer_to,
)
)
<http://logger.info|logger.info>(f"Transferred call to {transfer_to}")
return "Call transferred"
except Exception as e:
logger.error(f"Error transferring call: {e}")
# return or take additional actions (hangup, retry, etc.)
try:
# attempt to inform user of failure (best-effort, with timeout)
await asyncio.wait_for(
ctx.session.generate_reply(instructions="There was an error transferring the call in the current language."),
timeout=REPLY_TIMEOUT_SECONDS,
)
except Exception:
logger.warning("Could not generate failure announcement.")
# perform cleanup if required
await self.hangup()
return "Could not transfer call"
Notes and recommended adjustments
• Timeout value: Increase REPLY_TIMEOUT_SECONDS if your model normally needs more time. If your LLM is unreliable, keep the timeout so transfers are not blocked.
• Pre-generate audio: If you repeatedly use identical transfer phrasing, consider pre-generating the audio (TTS) and playing that instead of calling generate_reply live — this removes LLM variability.
• Logging: Keep detailed logs around the transfer call and the exact transfer_to value. Your earlier logs showed markdown-like brackets — ensure the value you pass is plain tel:+918956521667.
• SIP 403: If transfers still fail with SIP 403, the cause is likely permissions/config on the SIP trunk. The function above prevents the LLM timeout from blocking transfers but won’t change carrier permissions.
• Retries: If you want automatic retries for transient SIP errors, I can add a retry/backoff wrapper to the transfer call.
If you want, I can:
• Add exponential retries around transfer_sip_participant (with limits).
• Show an alternate flow that always returns the spoken reply to the user before transferring (i.e., wait but with longer timeout).
• Help interpret SIP logs if you still see 403 Forbidden errors.
Which of those would you like next?
Sources: Amazon Nova Sonic integration guide | LiveKit Documentation | Agent speech and audio | LiveKit Documentation