We have the following issue, happening, which is quite annoying and costly some times:
we have the following setup:
Operator(softphone jitsi) --(SIP)–> Asterisk → SIP provider → (the world).
The issues, follows this workflow:
- The operator makes a call.
- During the call, the operator experiences intermittent connectivity problem and for some time (lets say seconds to few minutes), does not reach Asterisk and does not hear any voice.
- The operator decides to hit ‘Hang Up’ button, which as far as jitsi is concernet “terminates” the call.
- In the reality, the call is NOT terminated, because the “hang up” signal does not reach the Asterisk, because of the mentioned connectivity problem.
- We are now at point, where the asterisk thinks, that he has open channels to the Operator device and to the dialed number.
- The connectivity is restored after short disruption but from Jitsi standpoint, we are no longer in a call.
- This situation stays for hours, unless the call/channels are manually terminated.
Is there anything I can do to avoid this situation, like in-call (in-channel) keepalives, which check if the Operator’s phone is unreachable and terminates both channels?
We are using Asterisk 11.25.0