The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
215 lines
5.6 KiB
C
215 lines
5.6 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Channel timeout related dialplan functions
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*
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* \author Mark Spencer <markster@digium.com>
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* \ingroup functions
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/app.h"
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/*** DOCUMENTATION
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<function name="TIMEOUT" language="en_US">
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<synopsis>
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Gets or sets timeouts on the channel. Timeout values are in seconds.
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</synopsis>
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<syntax>
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<parameter name="timeouttype" required="true">
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<para>The timeout that will be manipulated. The possible timeout types
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are: <literal>absolute</literal>, <literal>digit</literal> or
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<literal>response</literal></para>
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</parameter>
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</syntax>
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<description>
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<para>The timeouts that can be manipulated are:</para>
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<para><literal>absolute</literal>: The absolute maximum amount of time permitted for a call.
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Setting of 0 disables the timeout.</para>
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<para><literal>digit</literal>: The maximum amount of time permitted between digits when the
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user is typing in an extension. When this timeout expires,
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after the user has started to type in an extension, the
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extension will be considered complete, and will be
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interpreted. Note that if an extension typed in is valid,
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it will not have to timeout to be tested, so typically at
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the expiry of this timeout, the extension will be considered
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invalid (and thus control would be passed to the <literal>i</literal>
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extension, or if it doesn't exist the call would be
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terminated). The default timeout is 5 seconds.</para>
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<para><literal>response</literal>: The maximum amount of time permitted after falling through a
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series of priorities for a channel in which the user may
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begin typing an extension. If the user does not type an
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extension in this amount of time, control will pass to the
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<literal>t</literal> extension if it exists, and if not the call would be
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terminated. The default timeout is 10 seconds.</para>
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</description>
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</function>
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***/
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static int timeout_read(struct ast_channel *chan, const char *cmd, char *data,
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char *buf, size_t len)
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{
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struct timeval myt;
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if (!chan)
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return -1;
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if (!data) {
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ast_log(LOG_ERROR, "Must specify type of timeout to get.\n");
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return -1;
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}
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switch (*data) {
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case 'a':
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case 'A':
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if (ast_tvzero(*ast_channel_whentohangup(chan))) {
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ast_copy_string(buf, "0", len);
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} else {
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myt = ast_tvnow();
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snprintf(buf, len, "%.3f", ast_tvdiff_ms(*ast_channel_whentohangup(chan), myt) / 1000.0);
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}
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break;
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case 'r':
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case 'R':
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if (ast_channel_pbx(chan)) {
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snprintf(buf, len, "%.3f", ast_channel_pbx(chan)->rtimeoutms / 1000.0);
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}
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break;
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case 'd':
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case 'D':
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if (ast_channel_pbx(chan)) {
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snprintf(buf, len, "%.3f", ast_channel_pbx(chan)->dtimeoutms / 1000.0);
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}
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break;
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default:
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ast_log(LOG_ERROR, "Unknown timeout type specified.\n");
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return -1;
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}
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return 0;
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}
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static int timeout_write(struct ast_channel *chan, const char *cmd, char *data,
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const char *value)
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{
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double x = 0.0;
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long sec = 0L;
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char timestr[64];
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struct ast_tm myt;
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struct timeval when = {0,};
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int res;
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if (!chan)
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return -1;
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if (!data) {
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ast_log(LOG_ERROR, "Must specify type of timeout to set.\n");
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return -1;
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}
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if (!value)
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return -1;
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res = sscanf(value, "%30ld%30lf", &sec, &x);
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if (res == 0 || sec < 0) {
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when.tv_sec = 0;
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when.tv_usec = 0;
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} else if (res == 1) {
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when.tv_sec = sec;
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} else if (res == 2) {
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when.tv_sec = sec;
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when.tv_usec = x * 1000000;
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}
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switch (*data) {
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case 'a':
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case 'A':
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ast_channel_lock(chan);
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ast_channel_setwhentohangup_tv(chan, when);
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ast_channel_unlock(chan);
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if (VERBOSITY_ATLEAST(3)) {
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if (!ast_tvzero(*ast_channel_whentohangup(chan))) {
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when = ast_tvadd(when, ast_tvnow());
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ast_strftime(timestr, sizeof(timestr), "%Y-%m-%d %H:%M:%S.%3q %Z",
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ast_localtime(&when, &myt, NULL));
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ast_verb(3, "Channel will hangup at %s.\n", timestr);
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} else {
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ast_verb(3, "Channel hangup cancelled.\n");
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}
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}
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break;
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case 'r':
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case 'R':
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if (ast_channel_pbx(chan)) {
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ast_channel_pbx(chan)->rtimeoutms = when.tv_sec * 1000 + when.tv_usec / 1000;
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ast_verb(3, "Response timeout set to %.3f\n", ast_channel_pbx(chan)->rtimeoutms / 1000.0);
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}
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break;
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case 'd':
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case 'D':
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if (ast_channel_pbx(chan)) {
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ast_channel_pbx(chan)->dtimeoutms = when.tv_sec * 1000 + when.tv_usec / 1000;
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ast_verb(3, "Digit timeout set to %.3f\n", ast_channel_pbx(chan)->dtimeoutms / 1000.0);
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}
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break;
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default:
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ast_log(LOG_ERROR, "Unknown timeout type specified.\n");
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break;
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}
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return 0;
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}
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static struct ast_custom_function timeout_function = {
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.name = "TIMEOUT",
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.read = timeout_read,
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.read_max = 22,
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.write = timeout_write,
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};
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static int unload_module(void)
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{
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return ast_custom_function_unregister(&timeout_function);
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}
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static int load_module(void)
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{
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return ast_custom_function_register(&timeout_function);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Channel timeout dialplan functions");
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