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

[Joshua Colp]

> 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.

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.


- Joshua


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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
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> 
>   /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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