[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/#comment21558>

    These are supposed to be AST_TRANSPORT_xxx declarations.  SIP_TRANSPORT_xxx declarations don't exist.
    
    Please at least compile the patch.


- rmudgett


On April 25, 2014, 12:37 p.m., Patrick Laimbock wrote:
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> (Updated April 25, 2014, 12:37 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,
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> Patrick Laimbock
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>

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