From 9653b5d500c12bb5a60b4ad244706c15d20523f2 Mon Sep 17 00:00:00 2001
From: Terry Wilson <twilson@digium.com>
Date: Tue, 19 Oct 2010 19:35:24 +0000
Subject: [PATCH] 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peers' and removes an out-of-date comment from res_srtp.c and instead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 | 2 ++
res/res_srtp.c | 10 +---------
2 files changed, 3 insertions(+), 9 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d4a5a23638..7f400fcf73 100644
--- a/channels/chan_sip.c
+++ b/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)));
diff --git a/res/res_srtp.c b/res/res_srtp.c
index 808444ca07..3a330dba1f 100644
--- a/res/res_srtp.c
+++ b/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"
--
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