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.
When I dial 6003 from 6001, it takes 35 seconds until I get the error
message that 6003 is circuit-busy.
Any help would greatly be appreciated. Below is the error message and the
extensions and sip.conf files.
*CLI> == Using SIP RTP CoS mark 5
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. 😉
The best thing to do at this point would be to file a bug report with
the info at which point it will eventually probably be assigned to me
(unless some awesome person comes up with a fix first!) to look at. If I
have a bit of free time, I’ll try to take a peek at it. If you can post
the sip debug output of the entire offer/answer exchange to the bug
report, it will help greatly.