Does the FreePBX send out SIP REFER for call transfers out of the service provider trunk?
My softphone initiates a Transfer, a REFER is sent to the FreePBX to which it responds with 202, the Freepbx then initiates an INVITE towards the service provider instead of REFER, is there a way to change this behavior?
FreePBX does not support this directly, though you could write some custom dialplan that invokes the Asterisk Transfer command; see
However, this is usually not a good idea. You’ll have no record of how long the call lasted or even whether the transferred-to party answered. You won’t be able to end the call, record it, listen for DTMF, etc.
There is usually no economic advantage for doing this; the provider will still charge you for both legs of the call. If your motive is to save bandwidth or reduce latency, you can accomplish that by enabling Direct Media for the trunk(s) involved, which will cause the RTP to flow directly between the provider media server addresses. Of course, this precludes recording the call, though (if SIP INFO is supported), you may still be able to respond to DTMF.
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