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Bind an agent to a Retell-managed or imported number so it can answer inbound calls. If you route calls through your own carrier, see custom telephony.

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Bind voice agents

  • A number can receive and make calls only after you bind an agent to it.
  • You can assign different inbound and outbound agents to the number.
  • Leave an agent unset to disable that direction. For example, if you only make outbound calls and don’t want callbacks, leave inbound_agent_id unset.
Phone number configuration showing agent binding for inbound and outbound

Bind inbound and outbound agents to your number

After you bind an inbound agent, the number can receive calls.

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Handle inbound overflow

Inbound calls count toward your workspace concurrency limit. To keep inbound calls available during high outbound volume, you can reserve part of your concurrency for inbound calls. See Reserved Inbound Concurrency. If all concurrency slots are in use, Retell briefly waits for a slot to open. If no slot opens after about 40 seconds, Retell transfers the call to the phone number’s fallback_number when one is configured. Without a fallback number, the call ends with concurrency_limit_reached.

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A/B testing

See A/B Testing.

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Inbound call webhook

You may want to use different agents for inbound calls to the same number, or provide dynamic variables and other per-call fields. Read more at Inbound Call Webhook.

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Route calls to Retell’s SIP endpoint

If you’re routing inbound calls from your own carrier or PBX, point your SIP trunk or dial plan at Retell’s SIP server:
  • SIP server URI: sip:sip.retellai.com
  • Transports: TCP (recommended), UDP, TLS, mTLS. Append ;transport=tcp, ;transport=udp, or ;transport=tls to the URI to select one.
  • Media encryption: SRTP (requires TLS transport)
  • Audio codecs: PCMU, PCMA, G.722 (HD)
  • IP blocks to whitelist: 18.98.16.120/30, 3.42.144.0/23, 153.57.128.0/18, 143.223.88.0/21 (certain US traffic), 161.115.160.0/19 (certain US traffic)
For the full setup — elastic SIP trunking, dial-to-SIP-URI, mTLS, and provider-specific guides (Twilio, Telnyx, Vonage, Avaya, Genesys, Five9, Amazon Connect) — see custom telephony.

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Inbound custom SIP headers

  • You can use custom SIP headers to pre-set dynamic variables for a call.
  • Retell extracts any header starting with sip.h.x-, along with common SIP headers like Diversion, History-Info, User-To-User and P-Asserted-Identity, and converts each into a dynamic variable by stripping the sip.h. prefix.
    • E.g. sip.h.x-caller: abc -> x-caller: abc
    • E.g. sip.h.p-asserted-identity: +12345678910 -> p-asserted-identity: +12345678910
    • E.g. sip.h.diversion: <sip:2000@192.168.254.254>;privacy=off;reason=no-answer;counter=1;screen=no -> diversion: <sip:2000@192.168.254.254>;privacy=off;reason=no-answer;counter=1;screen=no

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Get call detail

  • API: Use the Get Call API to get information like transcript, recording, and latency tracking.
  • Webhook: Set up webhooks to receive real-time updates when a call is initiated, ends, and is analyzed. Read more at Call Webhook Guide.