4a58261694
Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
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_REGISTER_FILE()
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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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