[asterisk-dev] [Code Review] 3116: chan_sip: eliminate channel state down prior to hangup

[wdoekes]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3116/#review10572
-----------------------------------------------------------

Ship it!


I don't have anything to say about the ast_setstate, but Matt called it, so I guess it's good.

- wdoekes


On Jan. 10, 2014, 4:49 p.m., Scott Griepentrog wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3116/
> -----------------------------------------------------------
> 
> (Updated Jan. 10, 2014, 4:49 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-23010
>     https://issues.asterisk.org/jira/browse/ASTERISK-23010
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Setting the channel state to down is not done anywhere in chan_sip except handle_invite_replaces() after masquerading, and can interfere with the subsequent hangup of the channel.  Removing it appears to make sense, and causes no problems, although I was unable to observe a new BYE message in the packet trace of test sip_one_legged_transfer.
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_sip.c 405265 
> 
> Diff: https://reviewboard.asterisk.org/r/3116/diff/
> 
> 
> Testing
> -------
> 
> Ran all SIP tests in testsuite before and after, no new failures.
> 
> 
> Thanks,
> 
> Scott Griepentrog
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140113/8b8ff90a/attachment-0001.html>