Hello all. I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same behavior) and is having an issue when it comes to reINVITE on BYEs. Apparently one of the SIP providers that I am using does not always process reINVITEs correctly, and would return a 500 Internal Server Error message on some (but not all) of these transactions. To get around this issue, I have been using directrtpsetup = yes in my sip.conf, and it worked quite well. However, even with this option set, Asterisk would reINVITE itself back into the audio path as soon as the caller hangs up. The behavior I am seeing is that if the SIP provider sends back a 500 Internal Server Error on the reINVITE, Asterisk will not hang up the call until the called party hangs up. The transaction goes something like this:

1. Caller calls a number using a target SIP server.
2. — Early Media