On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote:
> I'm in the process of testing a TLS/SRTP install. My experience is improving
> with each new challenge, but this one is a great test of my 2 month
> experience with Asterisk.
> [myphones]
>
> ;exten => 6001,1,Dial(SIP/6001)
> ;exten => 6001,2,Hangup()
> exten => 6001,1,Set(_SIPSRTP_CRYPTO=enable)
> exten => 6001,2,Dial(SIP/${EXTEN})
>
There is no such thing as the _SIPSRTP_CRYPTO variable. That was from a very
old version of the SRTP patch. Ignore pretty much anything on issue 5413 and
instead look at
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics. You would
use encryption=yes/no in sip.conf and Set(CHANNEL(secure_bridge_signaling)=1)
to force SRTP calls. I'm assuming that you are using Asterisk 1.8 instead of
one of the patches on issue 5413--if not, then do that. ;-)
--
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