[asterisk-dev] [Code Review] 3066: bridge_native_rtp: Deadlock during 4-way conference creation

Kevin Harwell reviewboard at asterisk.org
Thu Dec 12 12:52:54 CST 2013



> On Dec. 12, 2013, 9:58 a.m., opticron wrote:
> > branches/12/bridges/bridge_native_rtp.c, lines 114-122
> > <https://reviewboard.asterisk.org/r/3066/diff/1/?file=49482#file49482line114>
> >
> >     This doxygen should get a retval section as well (probably my fault).

The return value for this function is always zero, so I'll changed it to void.


- Kevin


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3066/#review10392
-----------------------------------------------------------


On Dec. 11, 2013, 5:20 p.m., Kevin Harwell wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3066/
> -----------------------------------------------------------
> 
> (Updated Dec. 11, 2013, 5:20 p.m.)
> 
> 
> Review request for Asterisk Developers and kmoore.
> 
> 
> Bugs: ASTERISK-22749
>     https://issues.asterisk.org/jira/browse/ASTERISK-22749
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This contains the patch on the issue (submitted by kmoore), as well as the fix for the adding in a bridge lock while calling bridge_start from the framehook callback.  Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start.  The addition to the given patch adds in the necessary bridge locking in order to avoid a deadlock.
> 
> 
> Diffs
> -----
> 
>   branches/12/main/channel.c 403687 
>   branches/12/include/asterisk/channel.h 403687 
>   branches/12/channels/chan_sip.c 403687 
>   branches/12/channels/chan_pjsip.c 403687 
>   branches/12/bridges/bridge_native_rtp.c 403687 
> 
> Diff: https://reviewboard.asterisk.org/r/3066/diff/
> 
> 
> Testing
> -------
> 
> Added channels via DTMF attended transfer to get to a 4-way bridge and then removed them and noted all channels and the bridge had been torn down.
> 
> 
> Thanks,
> 
> Kevin Harwell
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131212/71e1eaf3/attachment-0001.html>


More information about the asterisk-dev mailing list