Is it possible to replace the Asterisk server with a new Asterisk server at the same IP address, and resume/take over all SIP calls in progress?
In other words, if a call between two user agents is bridged through an Asterisk server at IP 184.108.40.206, and I replace the server with a new one at IP 220.127.116.11 is there a way to have asterisk resume the call in progress? (Passing in the IP, session, ID, etc of the user agents or other info involved in active calls)
We looked at call survival upon server failover in the past but this was not possible. Looking again to see if anything has changed.