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(Updated Feb. 19, 2015, 11:30 a.m.) Status ------ This change has been marked as submitted. Review request for Asterisk Developers. Changes ------- Committed in revision 431956 Repository: Asterisk Description ------- * Fixed hangup handling of the session->channel after answer if the ast_channel_move() or ast_bridge_impart() fails. We are still the thread controlling the session->channel so we need to call ast_hangup() to kill the channel. * Fixed debug messages in refer_incoming_invite_request() referencing incorrect channnels on success. Code comments now say why the session->channel cannot be used. Diffs ----- /branches/13/res/res_pjsip_refer.c 431750 Diff: https://reviewboard.asterisk.org/r/4422/diff/ Testing ------- Using testsuite test tests/channels/pjsip/transfers/attended_transfer/nominal/callee_remote 1) Ran with patch. The debug log on ast2 was as expected. 2) Ran with patch and sabotaged code to "fail" ast_channel_move()/ast_bridge_impart(). The debug log on ast2 was as expected. 3) Ran with patch and sabotaged code to "fail" the initial test if the INVITE was a re-INVITE. The debug log on ast2 was as expected. Funny thing is the testsuite test passed for the three scenarios but a reactor timeout happened on 2 and 3. Thanks, rmudgett
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