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authorJonathan Rose <jrose@digium.com>2014-05-13 18:09:13 +0000
committerJonathan Rose <jrose@digium.com>2014-05-13 18:09:13 +0000
commite81b873fa2f23a277194722a00ec445127a68e57 (patch)
tree9317d38a48d05765de3f2b127f2af1ec0e9a202f /channels
parent0eda637fc477e32df1346fa4e5fbf0fdce907c8e (diff)
chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'
ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ ........ Merged revisions 413876 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413877 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c4
1 files changed, 4 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 16a8a55e0..a9a13790c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -21527,6 +21527,10 @@ static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_a
}
}
+ /* add transport and media types */
+ ast_cli(a->fd, " Transport: %s\n", ast_transport2str(cur->socket.type));
+ ast_cli(a->fd, " Media: %s\n", cur->srtp ? "SRTP" : cur->rtp ? "RTP" : "None");
+
ast_cli(a->fd, "\n\n");
found++;