[asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI

[Matt Jordan]

> On April 25, 2014, 1:03 p.m., rmudgett wrote:
> > /branches/11/channels/chan_sip.c, lines 21287-21295
> > <https://reviewboard.asterisk.org/r/3474/diff/3/?file=57909#file57909line21287>
> >
> >     These are supposed to be AST_TRANSPORT_xxx declarations.  SIP_TRANSPORT_xxx declarations don't exist.
> >     
> >     Please at least compile the patch.
> 
> rmudgett wrote:
>     Heh.  These were changed from SIP_TRANSPORT_xxx to AST_TRANSPORT_xxx in v12.

We can take care of that in the merge-ness. If this is the only problem left, I'd say it's ready to go.


- Matt


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On April 25, 2014, 12:37 p.m., Patrick Laimbock wrote:
> 
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> 
> (Updated April 25, 2014, 12:37 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-23564
>     https://issues.asterisk.org/jira/browse/ASTERISK-23564
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP.
> 
> 
> Diffs
> -----
> 
>   /branches/11/channels/chan_sip.c 412921 
> 
> Diff: https://reviewboard.asterisk.org/r/3474/diff/
> 
> 
> Testing
> -------
> 
> Testing was done on Asterisk-11.8.1 with TLS & RPT, TLS & SRTP, non-TLS & RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario.
> 
> 
> Thanks,
> 
> Patrick Laimbock
> 
>

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