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

[Terry Wilson]

On 03/03/2011 02:22 PM, Mitch Johnson wrote:
> Thanks so much for pointing this out.  I was curious why the commands in the documentation differed to the commands I was using.
>
> That problem is fixed, but now I have a new issue.  I can call with no issues, however, as soon as I answer one of the calls I see the error: ast_srtp_unprotect:  SRTP unprotect: authentication failure.  Below is a snippet of the debug as the call is answered.
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.

Terry