Hello all,
I am currently using Asterisk 13 with WebRTC client sipml5 on Google Chrome. I recently discovered that if there is an active call that includes an instance of sipml5 and the browser is closed before the hangup button is pressed on sipml5, there is no detection from the Asterisk server that sipml5 was closed (the call doesn’t actually end). Is there a way to configure Asterisk to detect when sipml5 is closed incorrectly so that way the server will know to terminate the call?
Thanks,
Kj