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Adding TCP and TLS to "sip show settings". 

TLS needs to have one configuration per configured domain at some point.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2008-07-05 21:37:57 +00:00
parent 6de5e8552b
commit d5525e3778
1 changed files with 19 additions and 6 deletions
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25
channels/chan_sip.c
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@ -13709,22 +13709,35 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
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, " 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, " 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, " 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, " Direct RTP setup: %s\n", cli_yesno(global_directrtpsetup));
ast_cli(a->fd, " User Agent: %s\n", global_useragent);