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| 19:42.23 | dwayne | newtonr, I have not yet opened an issue in the bug tracker. At first I wasn't sure if it would have been a duplicate |
| 19:42.34 | dwayne | I have since gathered a little more data on it |
| 19:45.42 | newtonr | dwayne, Yeah I didn't immediately find any straight up duplicate that was open. I say test in SVN of 1.8, then file an issue with appropriate pcaps, full logs with verbose,debug (logger.conf) and sip debug and we'll take a look |
| 19:46.15 | dwayne | OK. Hopefully, with a little more time I may have a patch too |
| 19:46.29 | newtonr | That's always welcome! :) |
| 19:48.59 | dwayne | newtonr, right now it looks like the call sending side (SSL client) has an existing SIP TLS connection to the receiving side that may have been closed on the receiving side, but not on the call sending side. I need to determine if the SIP TLS INVITE is received on the called box |
| 19:49.58 | dwayne | the call origination side is waiting for a response that doesn't come |
| 19:57.48 | newtonr | fun fun |
| 20:46.21 | dwayne | newtonr, do you have any idea of whether or not the side receiving the SIP TLS call (acting as the SSL Server) should be tearing down the SSL connection immediately after the call? It seems like its strange that the connection is staying up |
| 20:46.59 | newtonr | No idea. You might want to ask in asterisk-dev |
| 20:47.08 | dwayne | maybe it makes sense if there are concurrent calls to the destination from the same source |
| 20:47.17 | dwayne | ok |
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| 21:12.08 | dwayne | newtonr, a little more progress - the called side is NOT receiving the INVITE after this happens |
| 21:12.19 | dwayne | s/progress/information |
| 21:17.15 | dwayne | sorry, strike that .... someone was playing with the test environment |
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