[asterisk-dev] [svn-commits] mnicholson: branch 1.8 r326683 - /branches/1.8/channels/chan_sip.c

[Olle E. Johansson]

7 jul 2011 kl. 17.28 skrev SVN commits to the Digium repositories:

> -		snprintf(uri, sizeof(uri), "sip:%s@%s", p->username, ast_sockaddr_stringify_host_remote(&p->sa));
> +		snprintf(uri, sizeof(uri), "%s:%s@%s", p->socket.type == SIP_TRANSPORT_TLS ? "sips" : "sip", p->username, ast_sockaddr_stringify_host_remote(&p->sa));

YOu might want to recheck this. If I set an outbound proxy to use TLS, the transport is TLS, but the URI is not. Not ALL TLS transport messages have SIPS: 

There's also the case where NAPTR tells you to send a message over TLS but the URI stays SIP: 

Upgrading a URI to SIPS has a lot of implications for the continued life of a request.

--deleted long discussion between me, myself and I about SIP and TLS ---

/O