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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.

Bind voice agents

  • Bind an agent for each direction where you want normal AI call handling.
  • You can assign different inbound and outbound agents to the number.
  • Leave an agent unset to disable normal AI handling for 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.

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.

Divert calls with Backup mode

If you need to pause AI handling on one number, use Backup mode to send new inbound calls directly to that number’s fallback destination. For example, route calls to your staffed support line while you investigate an agent issue.
  1. Open Phone Numbers and select the number.
  2. Under Fallback Number, enter and save the destination number. Backup mode stays disabled until a fallback number is configured.
  3. Turn on Backup mode and confirm the change.
New inbound calls go to the fallback number before Retell selects an inbound agent. They do not wait for the concurrency queue to time out. Turn Backup mode off to resume normal AI routing. If Retell cannot connect to the fallback destination, the call ends. Backup mode applies to the selected number. It is separate from automatic concurrency fallback, which uses the same fallback destination when no concurrency slot becomes available.

Fallback billing

Both Backup mode and automatic concurrency fallback route the caller to the fallback destination without an agent handling the call, so no agent fee applies. For Retell-managed numbers, telephony charges apply at the rate for the fallback number’s country, from when the destination answers until the call ends (as of November 2026). Fallback cost = connected seconds ÷ 60 × the fallback destination’s per-minute rate. At an example rate of $0.015/min, a 10-minute fallback conversation costs $0.15, with no agent fee. See rate assumptions and examples. A no_concurrency_fallback call record can show that the agent call has ended while the fallback conversation is still accruing telephony charges. Backup mode can also incur telephony charges without creating an agent call record. With custom telephony, your provider bills the phone connection directly; Retell does not add a telephony fee. See call duration and telephony charges.

A/B testing

See A/B Testing.

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.

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)
  • Re-INVITE support: Retell answers in-dialog re-INVITEs (media re-anchor, codec change, SRTP re-key, hold/unhold, offerless) — see the custom telephony FAQ
  • IP blocks to whitelist: 18.98.16.120/30 (signaling only), 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.

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

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.