[asterisk-dev] [Code Review] Missing rtp changes in chan_skinny

[Dan Austin]

I also see three other rtp using channels that
are missing the get_codec function-
	chan_h323
	chan_unistim
	chan_multicast_rtp 

Should all three also be updated?

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Russell Bryant
Sent: Wednesday, July 28, 2010 2:04 PM
To: Russell Bryant; Asterisk Developers
Subject: Re: [asterisk-dev] [Code Review] Missing rtp changes in chan_skinny


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This is an automatically generated e-mail. To reply, visit:
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Ship it!


- Russell


On 2010-07-28 15:17:59, DEA wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/816/
> -----------------------------------------------------------
> 
> (Updated 2010-07-28 15:17:59)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Add get_codec function to chan_skinny.  Prevents segfaults on each and every call made from
> a skinny endpoint
> 
> 
> This addresses bug 16046.
>     https://issues.asterisk.org/view.php?id=16046
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/chan_skinny.c 280158 
> 
> Diff: https://reviewboard.asterisk.org/r/816/diff
> 
> 
> Testing
> -------
> 
> Made a call without crashing
> 
> 
> Thanks,
> 
> DEA
> 
>


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