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

[Mark Michelson]

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Ship it!


Ship It!

- Mark Michelson


On Feb. 14, 2015, 2:05 a.m., rmudgett wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4422/
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> (Updated Feb. 14, 2015, 2:05 a.m.)
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> 
> Review request for Asterisk Developers.
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> Repository: Asterisk
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> Description
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> 
> * 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.
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> 
> Diffs
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>   /branches/13/res/res_pjsip_refer.c 431750 
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> Diff: https://reviewboard.asterisk.org/r/4422/diff/
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> Testing
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> 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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> 
> Thanks,
> 
> rmudgett
> 
>

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