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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  peers' and removes an out-of-date comment from res_srtp.c and instead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: http://svn.digium.com/svn/asterisk/trunk@292310 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
twilson 2010-10-19 19:35:24 +00:00
parent 5dfdf6d60c
commit bce9e87be1
2 changed files with 3 additions and 9 deletions
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2
channels/chan_sip.c
View File

@ -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)));

10
res/res_srtp.c
View File

@ -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"