Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged).
But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify.
Is there any setup or asterisk configuration I need to enable to have A close its call ?
Note: when A is already talking with B, the call is hanged up on rtp timeout. But not during the Ringing phase.
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