Hi Team ,
I am Doing Create & Dial on a channel when the channel is answered what i am doing is play asterisk default recording and on playbackFinished Event i hangup the call
but when i tried to load test this on asterisk randomly i am getting this error
bridge.c:1963 ast_bridge_depart: Channel PJSIP/625024f1-000014a5 cannot be departed.
I am using the same Trunk to dial the call no change in that
I don’t underStand why because its very random and the point the error comes i checked the current channels on asterisk its 2500 channels
and i am using ari-client library for this
Can anyone Help why this error is coming