[asterisk-dev] [Code Review] Prevent segfault when asterisk restarts. Happens if call arrives before fully booted.

[rmudgett]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1407/#review4240
-----------------------------------------------------------

Ship it!


I'll go along with this specific fix.

- rmudgett


On Sept. 5, 2011, 5:15 a.m., Alec Davis wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1407/
> -----------------------------------------------------------
> 
> (Updated Sept. 5, 2011, 5:15 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> If a call arrives before asterisk is fully booted generally it will segfault.
> 
> 
> Diffs
> -----
> 
>   trunk/main/pbx.c 333893 
> 
> Diff: https://reviewboard.asterisk.org/r/1407/diff
> 
> 
> Testing
> -------
> 
> restarted asterisk, and before all modules have finished loading made a call into it.
> Warning message appears, and call is dropped.
> No orphaned channels.
> 
>   == Using SIP RTP CoS mark 5
>   == Registered translator 'slin 96000khz -> 32000khz' from format slin96 to slin32, table cost, 850000, computational cost 999999
> [2011-09-05 21:57:11.617417] WARNING[30782]: pbx.c:5363 ast_pbx_start: PBX requires Asterisk to be fully booted
> [2011-09-05 21:57:11.617973] WARNING[30782]: chan_sip.c:22917 handle_request_invite: Failed to start PBX :(
>   == Registered translator 'slin 96000khz -> 44100khz' from format slin96 to slin44, table cost, 850000, computational cost 999999
> 
> 
> Thanks,
> 
> Alec
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110906/985d5604/attachment.htm>