We have configured an Asterisk server with the dart-sip-ua batch manager (GitHub - flutter-webrtc/dart-sip-ua: A dart-lang version of the SIP UA stack.). However, we have encountered an issue where the connection breaks and the call is not restored. Is there a way to send invite in the call to the user through AMI (Asterisk Manager Interface)? Alternatively, has anyone had experience using this package and can provide guidance on implementing this functionality? We would greatly appreciate any ideas and suggestions.
hmm
sip do not have any way of reestablishing a failed call
the only way is that you in the dialplan try to make some logic to guess where you should redial or hangup
alternative you should look at increrasing session timers and rtp timeoud if they are to low
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