We have a phone behind a router doing SIP ALG connected to an asterisk server, and under some cirumstances SIP ALG dynamically remaps the DESTINATION port as well as the source port, sending the SIP BYE message to port 506X (e.g. 5063) instead of 5060. Asterisk of course therefore does not receive the BYE message, and the call never ends. Asterisk keeps sending alaw packets to the phone, but the phone isn’t sending anything back.
Is there a way to configure asterisk so that if a certain length of time passes (e.g. 1 minute) and no RTP packets have been received, to have the call automatically end?
I have attached a screenshot showing the phone “intended” on sending the BYE message to 5060 (note the request-line) but that it actually arrived destined for 5063.
Thanks for any advice you can give,