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

[Mitch Johnson]

> Once again, thanks for your reply.  I had done some research already but forget to include it in my previous email.  I did find a bug that is remarkably similar to the issues that I'm having.  The bug number is 18674.

Thanks,

Mitch Johnson

> Message: 8
> Date: Fri, 04 Mar 2011 00:34:45 -0600
> From: Terry Wilson <twilson at digium.com>
> Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID: <4D708805.3060409 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
> 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
>