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SIP REFER is a method defined in the Session Initiation Protocol (SIP) that allows one endpoint in a call to instruct another endpoint to initiate a new SIP request to a third party. In practice, it is most commonly used to perform call transfers. Instead of the transferring device directly establishing the new call itself, it sends a REFER request that tells the receiving endpoint where to send a new INVITE. The destination is specified in the Refer-To header, which contains the SIP URI of the party to be contacted. Once the REFER request is accepted— with a 202 Accepted response—the receiving endpoint initiates the new call to the referred destination.

In a typical call transfer scenario, two parties are already connected. When one endpoint wants to transfer the call, it sends a REFER request to its SIP server or call control platform. The server acknowledges the request and then initiates a new INVITE to the target identified in the Refer-To header. This creates a new call leg between the platform and the referred endpoint. Because the REFER method triggers another SIP transaction rather than carrying media itself, it operates strictly at the signaling layer of the SIP protocol.

 

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SIP REFER Support in Sinch Elastic SIP Trunking

 

Sinch Elastic SIP Trunking supports SIP REFER requests initiated by customers for both origination and termination traffic.

When a REFER request is sent to the Sinch Elastic SIP Trunking platform, the handling of the transfer depends on the URI contained in the Refer-To header.

  • Same Domain with New Phone Number:
    If the Refer-To URI contains a new phone number but retains the same domain, the call will be forwarded to the PSTN. Calls forwarded to the PSTN are treated as new outbound calls on the trunk and are therefore subject to the same routing rules, policies, and restrictions that apply to any standard outbound call. The new call leg to the PSTN  will be billed separately as if it were a regular call from your platform.
  • Different Domain:
    If the Refer-To URI contains a different domain, the call will instead be forwarded to the specified SIP destination. In this case, the destination does not need to be included in your Access Control List (ACL), and any routing rules or restrictions configured for the trunk will not be applied to this transfer. Calls transferred in this way do not incur any new fees. 

Referred calls can be found in your portal on the Call Detail Records page.  The new calls legs will have a second call record showing the referred call.

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Last update:
‎03-23-2026 11:23 AM
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