Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv11160/channels
Modified Files:
chan_sip.c
Log Message:
add 'sip show settings' CLI command (issue #4806)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.810
retrieving revision 1.811
diff -u -d -r1.810 -r1.811
--- chan_sip.c 23 Aug 2005 00:50:38 -0000 1.810
+++ chan_sip.c 23 Aug 2005 01:44:28 -0000 1.811
@@ -7091,6 +7091,24 @@
return RESULT_SUCCESS;
}
+/*--- print_codec_to_cli: Print codec list from preference to CLI/manager */
+static void print_codec_to_cli(int fd, struct ast_codec_pref *pref)
+{
+ int x, codec;
+
+ for(x = 0; x < 32 ; x++) {
+ codec = ast_codec_pref_index(pref, x);
+ if (!codec)
+ break;
+ ast_cli(fd, "%s", ast_getformatname(codec));
+ if (x < 31 && ast_codec_pref_index(pref, x + 1))
+ ast_cli(fd, ",");
+ }
+ if (!x)
+ ast_cli(fd, "none");
+}
+
+
static char mandescr_show_peer[] =
"Description: Show one SIP peer with details on current status.\n"
" The XML format is under development, feedback welcome! /oej\n"
@@ -7223,18 +7241,8 @@
ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1,
peer->capability);
ast_cli(fd, "%s\n", codec_buf);
ast_cli(fd, " Codec Order : (");
- pref = &peer->prefs;
- for(x = 0; x < 32 ; x++) {
- codec = ast_codec_pref_index(pref,x);
- if (!codec)
- break;
- ast_cli(fd, "%s", ast_getformatname(codec));
- if (x < 31 && ast_codec_pref_index(pref, x+1))
- ast_cli(fd, "|");
- }
+ print_codec_to_cli(fd, &peer->prefs);
- if (!x)
- ast_cli(fd, "none");
ast_cli(fd, ")\n");
ast_cli(fd, " Status : ");
@@ -7418,6 +7426,89 @@
#undef FORMAT2
}
+/*--- sip_show_settings: List global settings for the SIP channel ---*/
+static int sip_show_settings(int fd, int argc, char *argv[])
+{
+ char tmp[BUFSIZ];
+ int realtimepeers = 0;
+ int realtimeusers = 0;
+
+ realtimepeers = ast_check_realtime("sippeers");
+ realtimeusers = ast_check_realtime("sipusers");
+
+ if (argc != 3)
+ return RESULT_SHOWUSAGE;
+ ast_cli(fd, "\n\nGlobal Settings:\n");
+ ast_cli(fd, "----------------\n");
+ ast_cli(fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
+ ast_cli(fd, " Bindaddress: %s\n", ast_inet_ntoa(tmp,
sizeof(tmp), bindaddr.sin_addr));
+ ast_cli(fd, " Videosupport: %s\n", videosupport ? "Yes" :
"No");
+ ast_cli(fd, " AutoCreatePeer: %s\n", autocreatepeer ? "Yes" :
"No");
+ ast_cli(fd, " Allow unknown access: %s\n", global_allowguest ? "Yes"
: "No");
+ ast_cli(fd, " Promsic. redir: %s\n",
ast_test_flag(&global_flags, SIP_PROMISCREDIR) ? "Yes" : "No");
+ ast_cli(fd, " URI user is phone no: %s\n",
ast_test_flag(&global_flags, SIP_USEREQPHONE) ? "Yes" : "No");
+ ast_cli(fd, " Our auth realm %s\n", global_realm);
+ ast_cli(fd, " Realm. auth: %s\n", authl ? "Yes": "No");
+ ast_cli(fd, " User Agent: %s\n", default_useragent);
+ ast_cli(fd, " MWI checking interval: %d secs\n", global_mwitime);
+ ast_cli(fd, " Reg. context: %s\n",
ast_strlen_zero(regcontext) ? "(not set)" : regcontext);
+ ast_cli(fd, " Caller ID: %s\n", default_callerid);
+ ast_cli(fd, " From: Domain: %s\n", default_fromdomain);
+ ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" :
"Off");
+ ast_cli(fd, " Call Events: %s\n", callevents ? "On" :
"Off");
+ ast_cli(fd, " IP ToS: 0x%x\n", tos);
+#ifdef OSP_SUPPORT
+ ast_cli(fd, " OSP Support: Yes\n");
+#else
+ ast_cli(fd, " OSP Support: No\n");
+#endif
+ if (!realtimepeers && !realtimeusers)
+ ast_cli(fd, " SIP realtime: Disabled\n" );
+ else
+ ast_cli(fd, " SIP realtime: Enabled\n" );
+
+ ast_cli(fd, "\nGlobal Signalling Settings:\n");
+ ast_cli(fd, "---------------------------\n");
+ ast_cli(fd, " Codecs: ");
+ print_codec_to_cli(fd, &prefs);
+ ast_cli(fd, "\n");
+ ast_cli(fd, " Relax DTMF: %s\n", relaxdtmf ? "Yes" : "No");
+ ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" :
"No");
+ ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout,
global_rtptimeout ? "" : "(Disabled)" );
+ ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout,
global_rtpholdtimeout ? "" : "(Disabled)");
+ ast_cli(fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
+ ast_cli(fd, " DNS SRV lookup: %s\n", srvlookup ? "Yes" : "No");
+ ast_cli(fd, " Pedantic SIP support: %s\n", pedanticsipchecking ?
"Yes" : "No");
+ ast_cli(fd, " Reg. max duration: %d secs\n", max_expiry);
+ ast_cli(fd, " Reg. default duration: %d secs\n", default_expiry);
+ ast_cli(fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout);
+ ast_cli(fd, " Outbound reg. attempts: %d\n", global_regattempts_max);
+ ast_cli(fd, "\nDefault Settings:\n");
+ ast_cli(fd, "-----------------\n");
+ ast_cli(fd, " Context: %s\n", default_context);
+ ast_cli(fd, " Nat: %s\n",
nat2str(ast_test_flag(&global_flags, SIP_NAT)));
+ ast_cli(fd, " DTMF: %s\n",
dtmfmode2str(ast_test_flag(&global_flags, SIP_DTMF)));
+ ast_cli(fd, " Qualify: %d\n", default_qualify);
+ ast_cli(fd, " Use ClientCode: %s\n",
ast_test_flag(&global_flags, SIP_USECLIENTCODE) ? "Yes" : "No");
+ ast_cli(fd, " Progress inband: %s\n",
(ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ?
"Never" : (ast_test_flag(&global_flags, SIP_PROG_INBAND) == SIP_PROG_INBAND_NO)
? "No" : "Yes" );
+ ast_cli(fd, " Language: %s\n",
ast_strlen_zero(default_language) ? "(Defaults to English)" : default_language);
+ ast_cli(fd, " Musicclass: %s\n", global_musicclass);
+
+
+ if (realtimepeers || realtimeusers) {
+ ast_cli(fd, "\nRealtime SIP Settings:\n");
+ ast_cli(fd, "----------------------\n");
+ ast_cli(fd, " Realtime Peers: %s\n", realtimepeers ?
"Yes" : "No");
+ ast_cli(fd, " Realtime Users: %s\n", realtimeusers ?
"Yes" : "No");
+ ast_cli(fd, " Cache Friends: %s\n",
ast_test_flag(&global_flags_page2, SIP_PAGE2_RTCACHEFRIENDS) ? "Yes" : "No");
+ ast_cli(fd, " No update: %s\n",
ast_test_flag(&global_flags_page2, SIP_PAGE2_RTNOUPDATE) ? "Yes" : "No");
+ ast_cli(fd, " Ignore Reg. Expire: %s\n",
ast_test_flag(&global_flags_page2, SIP_PAGE2_RTIGNOREREGEXPIRE) ? "Yes" : "No");
+ ast_cli(fd, " Auto Clear: %d\n",
global_rtautoclear);
+ }
+ ast_cli(fd, "\n----\n");
+ return RESULT_SUCCESS;
+}
+
/* Forward declaration */
static int __sip_show_channels(int fd, int argc, char *argv[], int
subscriptions);
@@ -8271,6 +8362,11 @@
"Usage: sip show objects\n"
" Shows status of known SIP objects\n";
+static char show_settings_usage[] =
+"Usage: sip show settings\n"
+" Provides detailed list of the configuration of the SIP channel.\n";
+
+
/*--- func_header_read: Read SIP header (dialplan function) */
static char *func_header_read(struct ast_channel *chan, char *cmd, char *data,
char *buf, size_t len)
@@ -11747,6 +11843,7 @@
{ { "sip", "show", "channels", NULL }, sip_show_channels, "Show active
SIP channels", show_channels_usage},
{ { "sip", "show", "channel", NULL }, sip_show_channel, "Show detailed
SIP channel info", show_channel_usage, complete_sipch },
{ { "sip", "show", "history", NULL }, sip_show_history, "Show SIP
dialog history", show_history_usage, complete_sipch },
+ { { "sip", "show", "settings", NULL }, sip_show_settings, "Show SIP
global settings", show_settings_usage },
{ { "sip", "debug", NULL }, sip_do_debug, "Enable SIP debugging",
debug_usage },
{ { "sip", "debug", "ip", NULL }, sip_do_debug, "Enable SIP debugging
on IP", debug_usage },
{ { "sip", "debug", "peer", NULL }, sip_do_debug, "Enable SIP debugging
on Peername", debug_usage, complete_sip_debug_peer },
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