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chan_sip: Add TLS and SRTP status to CLI command 'sip show channel'

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ASTERISK-23564 #close
Reported by: Patrick Laimbock
Review: https://reviewboard.asterisk.org/r/3474/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jonathan Rose
2014-05-13 17:40:00 +00:00
parent adb50be36d
commit 93e4470a65
1 changed files with 22 additions and 0 deletions
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22
channels/chan_sip.c
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@@ -21163,6 +21163,24 @@ static char *complete_sip_notify(const char *line, const char *word, int pos, in
return NULL; return NULL;
} }
static const char *transport2str(enum sip_transport transport)
{
switch (transport) {
case SIP_TRANSPORT_TLS:
return "TLS";
case SIP_TRANSPORT_UDP:
return "UDP";
case SIP_TRANSPORT_TCP:
return "TCP";
case SIP_TRANSPORT_WS:
return "WS";
case SIP_TRANSPORT_WSS:
return "WSS";
}
return "Undefined";
}
/*! \brief Show details of one active dialog */ /*! \brief Show details of one active dialog */
static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a) static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{ {
@@ -21282,6 +21300,10 @@ 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", transport2str(cur->socket.type));
ast_cli(a->fd, " Media: %s\n", cur->srtp ? "SRTP" : cur->rtp ? "RTP" : "None");
ast_cli(a->fd, "\n\n"); ast_cli(a->fd, "\n\n");
found++; found++;