On Thu, 2012-03-08 at 16:50 +0000, Gavin Henry wrote: > >> > >> Ah, this makes sense now. So as of today the status of TLS and SRTP in > >> anything > >> other than 1.4.X is unknown? > > > > > > Umm... no :-) > > OK, sorry :-) > > > Asterisk 1.4 did not have support for SRTP or SIP/TLS. Thus, neither of > > these were tested with Polycom phones the last time we did interop testing > > with those phones. > > Ah, I forgot when it was added. > afaicr, it was in 1.6.2 > > The status of SIP/TLS and SRTP support in the Asterisk releases that have > > them are not 'unknown'; they are there and expected to be working. I was > > just pointing out that Digium has not specifically tested Polycom phones for > > interop with these features, and certainly has not specifically tested usage > > of TLS certificates issued by any particular CA. >
btw, "commercial" certs are not so special. Somewhere in the chain (root-ca), there is a self-signed cert. You can make such chain yourself, root-ca -> sub-ca -> sub-ca and finally a server+client cert. Or, you can get a free cert from cacert.org hw -- _____________________________________________________________________ -- 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