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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-19 19:27:32 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-10-19 19:27:32 +0000
commit18fb76b299de35308e364e941c00fcc336db4546 (patch)
treed8a37896268a3685f12e602d5e821624224b84d3 /res
parent220c9324f908b4a3d5b3401f6f6f54f925880ac0 (diff)
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This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation.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/branches/1.8@292309 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res')
-rw-r--r--res/res_srtp.c10
1 files changed, 1 insertions, 9 deletions
diff --git a/res/res_srtp.c b/res/res_srtp.c
index 808444ca0..3a330dba1 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"