[asterisk-dev] [Code Review] 3617: Fix for AMI and SIP TCP being unresponsive to sending ouput.

[Matt Jordan]

> On June 12, 2014, 11:42 p.m., Matt Jordan wrote:
> > Ship It!

The necessity of stripping a value we control in our own API down to a 0/1 not withstanding, there's no reason why we shouldn't fix this rather critical and very time sensitive issue :-)


- Matt


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On June 12, 2014, 10:48 p.m., rmudgett wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3617/
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> 
> (Updated June 12, 2014, 10:48 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-23673
>     https://issues.asterisk.org/jira/browse/ASTERISK-23673
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Oops.  I broke it.
> 
> Unable to login to AMI and get output so it looks like you didn't get connected.
> 
> SIP TCP connections are unable to send responses.
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/main/tcptls.c 416055 
>   /branches/1.8/main/manager.c 416055 
>   /branches/1.8/main/http.c 416055 
>   /branches/1.8/include/asterisk/tcptls.h 416055 
>   /branches/1.8/channels/chan_sip.c 416055 
> 
> Diff: https://reviewboard.asterisk.org/r/3617/diff/
> 
> 
> Testing
> -------
> 
> With the patch, AMI is able to get connected and async events are able to go out.
> With the patch, HTTP is able to timeout connections that don't complete.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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