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

[Mark Michelson]

> On Jan. 23, 2015, 3:30 p.m., Matt Jordan wrote:
> > 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.
> 
> Joshua Colp wrote:
>     I'm in the same boat. I don't have any findings. I think Mark should take a look and then this'll be fine.

Yeah, going through the problem, there may be other ways of fixing the issue, but this seems like the way that is least invasive and least likely to cause issues.


- Mark


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On Jan. 20, 2015, 6:36 p.m., Jeremiah Gowdy wrote:
> 
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> https://reviewboard.asterisk.org/r/4362/
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> 
> (Updated Jan. 20, 2015, 6:36 p.m.)
> 
> 
> Review request for Asterisk Developers and Matt Jordan.
> 
> 
> Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22436
>     https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22436
> 
> 
> Repository: Asterisk
> 
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   /branches/11/channels/sip/include/sip.h 430836 
>   /branches/11/channels/chan_sip.c 430836 
> 
> Diff: https://reviewboard.asterisk.org/r/4362/diff/
> 
> 
> Testing
> -------
> 
> 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.
> 
> 
> Thanks,
> 
> Jeremiah Gowdy
> 
>

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