Merged revisions 292309 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines
  
  Add sip show peer info about crypto and remove dated comment
  
  This patch adds information about the encryption setting to 'sip show
  peers' and removes an out-of-date comment from res_srtp.c and instead
  directs users to the proper documentation.
  
  (closes issue #18140)
  Reported by: chodorenko
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292310 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -16467,6 +16467,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
 		ast_cli(fd, "  RTP Engine   : %s\n", peer->engine);
 		ast_cli(fd, "  Parkinglot   : %s\n", peer->parkinglot);
 		ast_cli(fd, "  Use Reason   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
+		ast_cli(fd, "  Encryption   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
 		ast_cli(fd, "\n");
 		peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
 	} else  if (peer && type == 1) { /* manager listing */
@@ -16522,6 +16523,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
 		astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
 		astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
 		astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
+		astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");
 
 		/* - is enumerated */
 		astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));

res/res_srtp.c

@@ -32,15 +32,7 @@
          <depend>srtp</depend>
 ***/
 
-/* The SIP channel will automatically use sdescriptions if received in a SDP offer,
-   and res_srtp is loaded. SRTP with sdescriptions key exchange can be activated
-  in outgoing offers by setting _SIPSRTP_CRYPTO=enable in extension.conf before executing Dial
-
-  The dial fails if the callee doesn't support SRTP and sdescriptions.
-
-  exten => 2345,1,Set(_SIPSRTP_CRYPTO=enable)
-  exten => 2345,2,Dial(SIP/1001)
-*/
+/* See doc/tex/secure-calls.tex for SRTP usage information */
 
 #include "asterisk.h"