docs(agents): BYO SIP cold transfers need PSTN transfers enabled on the trunk

agents/telephony/byo-sip.mdx

@@ -6,11 +6,11 @@
 
 If your numbers already live at a carrier, you can connect them to your agents without porting anything. Point the carrier's SIP trunk at Fish Audio and import the number: it stays with your carrier, who keeps billing you for the telephone-network legs, and on Fish Audio the calls bill as ordinary agent sessions. Imported numbers carry no monthly rental and no telephony charges of any kind: no phone surcharge, no transfer fees; the call bills like a web session.
 
 This works with any carrier or PBX that speaks SIP trunking: Twilio Elastic SIP Trunking, Asterisk or FreePBX, and most SIP providers. It is also the only way to use non-US/CA numbers, which the purchasable inventory does not cover.
 
 ## How it works
 
 A SIP trunk has two directions, and carriers name them from the carrier's point of view. **Origination** is traffic your carrier originates towards Fish Audio: an inbound call. **Termination** is traffic your carrier terminates for Fish Audio: an outbound call the platform hands to your carrier or PBX to complete. Twilio's trunk configuration uses the same two words, and so do the fields below.
 
 - **Inbound (origination)**: your carrier routes calls for the number over its trunk to Fish Audio's SIP endpoint. The platform matches the dialed number and hands the call to the agent bound to it. From there it is a normal [inbound call](/agents/telephony/inbound-calls).
 - **Outbound (termination)**, optional: give the import a termination host and the platform can also place calls from the number. Outbound calls and warm-transfer consult legs dial out through your trunk, with the imported number as the caller ID. You can also attach custom SIP headers that every call through the termination carries, for routing hints or tenant identification on your PBX.
@@ -84,30 +84,30 @@
 The SIP endpoint is shared, so an import must prove that calls really come from your trunk:
 
 - **Digest credentials**: the platform challenges your trunk and verifies the username and password. Use this whenever your carrier or PBX answers digest challenges (Asterisk, FreePBX, most SIP providers).
 - **Allowed source addresses**: calls are only accepted from the listed IPs or CIDR ranges. Use this for carriers that do not authenticate their origination traffic; Twilio Elastic SIP Trunking is one, so for Twilio this is the required factor.
 
 Set both when your carrier supports it.
 
 ## Carrier walkthroughs
 
 ### Twilio Elastic SIP Trunking
 
 <Steps>
   <Step title="Create a trunk">
     In the Twilio console, under **Elastic SIP Trunking**, create a trunk (or
     reuse an existing one).
   </Step>
   <Step title="Point origination at Fish Audio">
     Add an origination URI: `sip:1pv316az391.sip.livekit.cloud;transport=tcp`.
   </Step>
   <Step title="Attach your number">
     On the trunk's **Numbers** tab, add the phone number. Twilio routes its
     calls through the trunk from then on.
   </Step>
   <Step title="Import on Fish Audio">
     Twilio's origination does not answer digest challenges, so authenticate by
     source address: allow Twilio's published signaling IP ranges for the regions
     you use (see [Twilio's IP address
     list](https://www.twilio.com/docs/sip-trunking/ip-addresses)). In the
     console, the **Twilio Elastic SIP Trunking preset** button fills the ranges
     and sets the transport for you.
@@ -117,19 +117,21 @@
     (`yourprefix.pstn.twilio.com`) and attach a **Credential List**. Pass the
     host as `termination_uri` and the credentials as `termination_auth_username`
     and `termination_auth_password`. Credentials are required here: Fish Audio's
     outbound traffic does not come from fixed IPs, so Twilio IP access control
     lists cannot authorize it.
   </Step>
   <Step title="Allow transfers (optional)">
     For [cold transfers](/agents/telephony/transfers), enable **Call Transfer
-    (SIP REFER)** in the trunk's settings so Twilio honors the handoff.
+    (SIP REFER)** in the trunk's settings and also allow transfers to the PSTN.
+    If transfers are limited to SIP destinations, Twilio rejects every transfer
+    to a phone number, so transfers to phone numbers won't work.
   </Step>
 </Steps>
 
 ### Asterisk, FreePBX, and other SIP platforms
 
 - Route the number's inbound calls to `sip:1pv316az391.sip.livekit.cloud;transport=tcp`.
 - Configure digest credentials on the trunk and pass the same pair as `inbound_auth_username` and `inbound_auth_password`; add your PBX's public IPs to `inbound_allowed_addresses` for defense in depth.
 - For outbound, expose a termination host reachable from the internet and pass it as `termination_uri`, with digest credentials if your PBX requires registration or authentication.
 
 ## Outbound calls and transfers
@@ -224,7 +226,7 @@
   }'
 ```
 
 The body is the import payload without `phone_number`, `label`, and `agent_id`, and it replaces the whole configuration. Three things to know:
 
 - An empty password next to a set username keeps the stored password, so you can edit other fields without re-entering secrets. Setting a username for the first time requires a password.
 - An empty `termination_uri` removes the termination and makes the number inbound-only again.

agents/telephony/transfers.mdx

@@ -17,7 +17,7 @@
 ## Prerequisites
 
 - A phone number bound to your agent, answering inbound calls. See [Inbound calls](/agents/telephony/inbound-calls).
-- The number must support transfers. Numbers purchased from the platform inventory (`provider: "twilio"`) do; see [Phone numbers](/agents/telephony/phone-numbers). [Imported SIP numbers](/agents/telephony/byo-sip) support cold transfers when the carrier honors SIP REFER, and warm transfers when a termination is configured.
+- The number must support transfers. Numbers purchased from the platform inventory (`provider: "twilio"`) do; see [Phone numbers](/agents/telephony/phone-numbers). [Imported SIP numbers](/agents/telephony/byo-sip) support cold transfers when the carrier honors SIP REFER to the destination (on Twilio, enable Call Transfer and allow PSTN transfers on the trunk), and warm transfers when a termination is configured.
 
 ## Configure destinations
 
@@ -185,8 +185,7 @@
 ## Limitations
 
 - **Phone calls only**: web and SDK sessions have no phone leg to hand off, so the `transfer_call` tool never ships for them.
-- **Transfer-capable numbers**: the number the caller dialed must be a `twilio`-provider number.
 - **Supported countries**: destination numbers are limited to an allowlist of country codes.
 
 ## Billing