> 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.
Heh. These were changed from SIP_TRANSPORT_xxx to AST_TRANSPORT_xxx in v12. - rmudgett ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3474/#review11751 ----------------------------------------------------------- On April 25, 2014, 12:37 p.m., Patrick Laimbock wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3474/ > ----------------------------------------------------------- > > (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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