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

[rmudgett]

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/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21505>

    Guidelines say you cannot declare a variable in the middle of a code block.



/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21506>

    It would be better if this were put into its own function.  Something like transport2str() that returns the transport string.



/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21508>

    Add curlies.



/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21507>

    red blob


- rmudgett


On April 24, 2014, 12:33 p.m., Patrick Laimbock wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3474/
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> 
> (Updated April 24, 2014, 12:33 p.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-23564
>     https://issues.asterisk.org/jira/browse/ASTERISK-23564
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> Repository: Asterisk
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> Description
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> 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.
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> 
> Diffs
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>   /branches/11/channels/chan_sip.c 412921 
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> Diff: https://reviewboard.asterisk.org/r/3474/diff/
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> Testing
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> 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.
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> Thanks,
> 
> Patrick Laimbock
> 
>

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