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peers' and removes an out-of-date comment from res_srtp.c and instead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/branches/1.8@292309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Terry Wilson 15 years ago
parent b334bdef1e
commit 668d532d6b

@ -16394,6 +16394,7 @@ 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, " RTP Engine : %s\n", peer->engine);
ast_cli(fd, " Parkinglot : %s\n", peer->parkinglot); 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, " 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"); ast_cli(fd, "\n");
peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr"); peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */ } else if (peer && type == 1) { /* manager listing */
@ -16449,6 +16450,7 @@ 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-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-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-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 */ /* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF))); astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));

@ -32,15 +32,7 @@
<depend>srtp</depend> <depend>srtp</depend>
***/ ***/
/* The SIP channel will automatically use sdescriptions if received in a SDP offer, /* See doc/tex/secure-calls.tex for SRTP usage information */
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)
*/
#include "asterisk.h" #include "asterisk.h"

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