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. ;-)
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users