SIP Transfers
Transfer calls to SIP-compatible endpoints.
SIP transfers enable your AI agent to hand off calls to SIP endpoints such as PBX systems, SIP phones, or softphones.
Looking for other transfer types? See the Call Transfers overview for phone number (TEL) and dynamic transfer options.
Prerequisites
- Your SIP endpoint must be reachable from Synthflow
- SIP (UDP/TCP 5060) and RTP ports open and forwarded
Setup
Custom X-Headers
You can define custom X-headers in the UI to pass additional metadata with your SIP INVITE requests. This is useful for routing logic, caller identification, or integration with your PBX system.

Header values can be populated using variables or info extractors.
Variables
Use {} syntax to inject dynamic values into a header:
If you use a custom action before the transfer, the transfer waits for the action to resolve before initiating. This ensures variables returned by the action are available in your SIP headers.
Info Extractors
An info extractor lets the AI agent pull a specific piece of information from the conversation and pass it as a header value. Each extractor has two fields:
The agent evaluates the prompt against the conversation at transfer time and writes the result into the specified header. Use info extractors when the value you need doesn’t exist as a variable — for example, a caller’s intent, sentiment, or a summary of the conversation so far.
For more on how extractors work, see Information Extractors.
Example: Passing Call Context for Agent Handoff
A common pattern is to combine variables and info extractors so the human agent receives full context on transfer:
Field Requirements
Avoid adding too many X-headers. Excessive headers increase the size of SIP INVITE packets, which can lead to packet fragmentation and loss—especially over UDP. Keep your headers minimal and only include data that your PBX or routing logic actually needs.
Supported SIP URI Formats
SIP Methods: REFER vs INVITE
The SIP method used depends on your transfer mode:
SIP REFER considerations:
- Caller ID shows the original caller’s number
- Requires your SIP server to support REFER
- Transfer happens entirely on your telephony infrastructure
SIP INVITE considerations:
- Caller ID shows your Synthflow agent’s number
- Allows whisper messages and conversation summaries
- Synthflow maintains control until transfer completes
Security
Synthflow supports Access Control Lists (ACL) to secure inbound SIP calls. Configure your SIP server to accept connections only from Synthflow’s IP addresses. Username/password authentication is not currently supported for inbound call transfers.
Transfer Failure Handling
For mission-critical transfers, use warm transfer mode to enable graceful failure recovery and alternative routing.