[asterisk-dev] [Code Review] 4422: res_pjsip_refer: Handle INVITE with Replaces failure after answer.

rmudgett reviewboard at asterisk.org
Thu Feb 19 11:30:20 CST 2015

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(Updated Feb. 19, 2015, 11:30 a.m.)


This change has been marked as submitted.

Review request for Asterisk Developers.


Committed in revision 431956

Repository: Asterisk


* 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.


  /branches/13/res/res_pjsip_refer.c 431750 

Diff: https://reviewboard.asterisk.org/r/4422/diff/


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.



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