To make sure your agent starts talking only after the SIP call status is active, you can keep an eye on the

sip.callStatus

attribute of the participant. Here’s a friendly step-by-step on how to do this: 1. Create the SIP Participant: Use the

create_sip_participant

method to kick off the call. This will start dialing the user and return right away. 2. Watch the Call Status: After you create the participant, keep checking the

sip.callStatus

attribute. It’ll switch to

active

once the call is answered. 3. Let the Agent Speak: As soon as the call status hits

active

, use the agent's

generate_reply

method to get the conversation going. Here’s a handy Python snippet to guide you:

import asyncio

async def monitor_call_status(participant):
    start_time = asyncio.get_event_loop().time()
    while asyncio.get_event_loop().time() - start_time < 30:
        call_status = participant.attributes.get("sip.callStatus")
        if call_status == "active":
            print("Call is active, agent will start speaking.")
            await session.generate_reply(
                instructions="Greet the user and offer your assistance."
            )
            return
        elif call_status == "hangup":
            print("Call has been ended.")
            break
        await asyncio.sleep(0.1)
    print("Call monitoring timed out.")

This way, your agent will jump in right after the call connects. If you have any questions or need a hand with this, just let me know—I’m here to help! Sources: agents/src/pipeline/pipeline_agent.ts | Make outbound calls | LiveKit Docs | Making calls using SIP | LiveKit Docs | Making calls using SIP | LiveKit Docs | Agents telephony integration | LiveKit Docs