[asterisk-users] TLS/SRTP calls go to circuit busy.

[Terry Wilson]

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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