> 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 <twil...@digium.com> > Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4d708805.3060...@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 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users