[asterisk-dev] [Code Review] 4362: chan_sip Invite: Replaces hangup bug fix

[Matt Jordan]

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After reading through the analysis on the underlying ASTERISK issue, I don't have any findings with the patch. I'm always a little concerned when we have to add a new state to keep track of on the sip_pvt, but right now I can't think of another property that would be appropriate.

It'd probably be good for someone who has spent more time in the chan_sip transfer code to look at this as well, just to make sure I'm not missing anything.

- Matt Jordan


On Jan. 20, 2015, 12:36 p.m., Jeremiah Gowdy wrote:
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> (Updated Jan. 20, 2015, 12:36 p.m.)
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> Review request for Asterisk Developers and Matt Jordan.
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> Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22436
>     https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22436
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> Repository: Asterisk
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> Description
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> chan_sip: This patch fixes a bug in chan_sip's handling of Invite: Replaces which currently never hangs up on the replaced call.  It adds an additional flag to track the fact that we're doing a replaces and then uses that flag to determine if we should send a BYE.
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> Diffs
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>   /branches/11/channels/sip/include/sip.h 430836 
>   /branches/11/channels/chan_sip.c 430836 
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> Diff: https://reviewboard.asterisk.org/r/4362/diff/
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> Testing
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> This is running in production for a beta product we have now.  Our development and QA staff have done manual testing and found no issues.
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> Thanks,
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> Jeremiah Gowdy
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