I’m running Asterisk 1.0.6 using its SIP functionality only. I have a test setup with three SIP ATAs (A,B,C) connected to * and testing call waiting.
- B calls A 2) C calls A 3) A does a flash-hook to put B on hold. UA(A) sends a reINVITE(HOLD) to * that should be forwarded to B, but * does not do this.
I tried setting the ‘canreinvite’ flag to ‘yes’ in sip.conf but that just seems to have the effect of * generating the reInvites to establish a direct RTP connection between the ATAs.