<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div>On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">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.<br></blockquote><br><blockquote type="cite">[myphones]<br><br>;exten => 6001,1,Dial(SIP/6001)<br>;exten => 6001,2,Hangup()<br>exten => 6001,1,Set(_SIPSRTP_CRYPTO=enable)<br>
exten => 6001,2,Dial(SIP/${EXTEN})<br><font class="Apple-style-span" color="#000000"><font class="Apple-style-span" color="#144FAE"><br></font></font></blockquote><div><br></div><div>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 <a href="https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial">https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial</a> and <a href="https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics">https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics</a>. 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. ;-)</div></div><br></body></html>