[asterisk-commits] oej: branch 1.6.0 r128291 - in /branches/1.6.0: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sat Jul 5 16:56:34 CDT 2008
Author: oej
Date: Sat Jul 5 16:56:33 2008
New Revision: 128291
URL: http://svn.digium.com/view/asterisk?view=rev&rev=128291
Log:
Merged revisions 128287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r128287 | oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines
Adding TCP and TLS to "sip show settings".
TLS needs to have one configuration per configured domain at some point.
........
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/channels/chan_sip.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=128291&r1=128290&r2=128291
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Sat Jul 5 16:56:33 2008
@@ -12990,22 +12990,35 @@
return CLI_SHOWUSAGE;
ast_cli(a->fd, "\n\nGlobal Settings:\n");
ast_cli(a->fd, "----------------\n");
- ast_cli(a->fd, " SIP Port: %d\n", ntohs(bindaddr.sin_port));
- ast_cli(a->fd, " Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
+ ast_cli(a->fd, " UDP SIP Port: %d\n", ntohs(bindaddr.sin_port));
+ ast_cli(a->fd, " UDP Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
+ ast_cli(a->fd, " TCP SIP Port: ");
+ if (sip_tcp_desc.sin.sin_family != AF_INET) {
+ ast_cli(a->fd, "%d\n", ntohs(sip_tcp_desc.sin.sin_port));
+ ast_cli(a->fd, " TCP Bindaddress: %s\n", ast_inet_ntoa(sip_tcp_desc.sin.sin_addr));
+ } else {
+ ast_cli(a->fd, "Disabled");
+ }
+ if (default_tls_cfg.enabled != FALSE) {
+ ast_cli(a->fd, "%d\n", ntohs(sip_tls_desc.sin.sin_port));
+ ast_cli(a->fd, " TLS Bindaddress: %s\n", ast_inet_ntoa(sip_tls_desc.sin.sin_addr));
+ } else {
+ ast_cli(a->fd, "Disabled");
+ }
ast_cli(a->fd, " Videosupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
ast_cli(a->fd, " Textsupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
ast_cli(a->fd, " AutoCreate Peer: %s\n", cli_yesno(autocreatepeer));
ast_cli(a->fd, " Match Auth Username: %s\n", cli_yesno(global_match_auth_username));
ast_cli(a->fd, " Allow unknown access: %s\n", cli_yesno(global_allowguest));
ast_cli(a->fd, " Allow subscriptions: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
+ ast_cli(a->fd, " Allow overlap dialing: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
+ ast_cli(a->fd, " Allow promsic. redir: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
ast_cli(a->fd, " Enable call counters: %s\n", cli_yesno(global_callcounter));
- ast_cli(a->fd, " Allow overlap dialing: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
- ast_cli(a->fd, " Promsic. redir: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
ast_cli(a->fd, " SIP domain support: %s\n", cli_yesno(!AST_LIST_EMPTY(&domain_list)));
+ ast_cli(a->fd, " Realm. auth: %s\n", cli_yesno(authl != NULL));
+ ast_cli(a->fd, " Our auth realm %s\n", global_realm);
ast_cli(a->fd, " Call to non-local dom.: %s\n", cli_yesno(allow_external_domains));
ast_cli(a->fd, " URI user is phone no: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
- ast_cli(a->fd, " Our auth realm %s\n", global_realm);
- ast_cli(a->fd, " Realm. auth: %s\n", cli_yesno(authl != NULL));
ast_cli(a->fd, " Always auth rejects: %s\n", cli_yesno(global_alwaysauthreject));
ast_cli(a->fd, " Call limit peers only: %s\n", cli_yesno(global_limitonpeers));
ast_cli(a->fd, " Direct RTP setup: %s\n", cli_yesno(global_directrtpsetup));
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