Thanks — I’m responding to your question about routing a SIP call to two different numbers using Telnyx, LiveKit Cloud, the JS SDK, and Python agents. Short answer • Create an outbound SIP participant for each phone number you want to reach using the agent’s access to LiveKit’s SIP API (i.e., call

create_sip_participant

once per number). Each SIP participant joins the same LiveKit room so both numbers are connected into that room (either sequentially or in parallel). What you can do (high-level) • Sequential dialing (dial number A, then dial number B after A answers or fails). • Parallel ringing / simultaneous dialing (dial A and B at the same time and accept the first to answer). • Put both answered SIP participants into the same LiveKit room to enable mixed audio, conferencing, or bridging logic. Concrete steps and examples 1) Ensure you have an outbound SIP trunk configured (Telnyx) and the

sip_trunk_id

ready. 2) From your Python agent, call

create_sip_participant

once per target number. Example Python agent pattern (conceptual — adapt to your agent scaffolding):

# inside your agent function
from livekit import api  # pseudo-import, adapt to actual client

# sequential example
try:
    await api_client.sip.create_sip_participant(
        api.CreateSIPParticipantRequest(
            room_name=ctx.room.name,
            sip_trunk_id=MY_SIP_TRUNK_ID,
            sip_call_to=number1,
            participant_identity=f"sip_{number1}",
            krisp_enabled=True,
            wait_until_answered=True,
        )
    )
    # number1 answered (or connected)
except Exception:
    # handle failure

# then dial second number
try:
    await api_client.sip.create_sip_participant(
        api.CreateSIPParticipantRequest(
            room_name=ctx.room.name,
            sip_trunk_id=MY_SIP_TRUNK_ID,
            sip_call_to=number2,
            participant_identity=f"sip_{number2}",
            krisp_enabled=True,
            wait_until_answered=True,
        )
    )
except Exception:
    # handle failure

Parallel (simultaneous) ringing example using asyncio.gather:

await asyncio.gather(
    api_client.sip.create_sip_participant( api.CreateSIPParticipantRequest(..., sip_call_to=number1) ),
    api_client.sip.create_sip_participant( api.CreateSIPParticipantRequest(..., sip_call_to=number2) ),
)

• If you want “first to answer” behavior, call both in parallel and implement logic to keep the first answered participant and hang up the other. • If you want a conference between the two numbers, keep both SIP participants in the same LiveKit room after they answer. 3) Initiate agent invocation from your JS front end (or server) — pass target numbers via metadata or parameters so the Python agent knows which numbers to dial. Notes, limitations, and helpful tips • All routing/bridging must occur through a LiveKit room; there’s no native SIP-only bridging outside the room. • Caller ID: the trunk’s number is normally used. If you need to override caller ID, use the trunk/number override options supported by your Telnyx trunk (subject to Telnyx policies). • Use

wait_until_answered=True

if you need the agent to wait for answer before continuing; omit it to fire-and-forget or to handle answer events asynchronously. • Use

participant_identity

to label SIP participants so you can find/control them later in room participant lists. If you’d like, I can: • Provide a ready-to-run Python agent code sample tailored to your agent framework (please tell me which agent template you’re using). • Show how to implement “first-to-answer” logic (race handling) and cleanly hang up the losing leg. • Walk through how to pass numbers from your JS SDK client into the agent (showing example metadata/dispatch command). Which of those would help you next, and do you have a preferred dialing pattern: sequential, parallel (simultaneous), or conference? Sources: Agents telephony integration | LiveKit Documentation | Outbound Calling Script | LiveKit Documentation | sip | livekit/livekit_composite | DeepWiki | Preserving Caller ID During SIP Transfers in LiveKit Agents | Preserve caller ID during SIP transfers with Twilio | LiveKit