asterisk/apps/app_readexten.c

289 lines
8.4 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2007-2008, Trinity College Computing Center
* Written by David Chappell <David.Chappell@trincoll.edu>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Trivial application to read an extension into a variable
*
* \author David Chappell <David.Chappell@trincoll.edu>
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/module.h"
#include "asterisk/indications.h"
#include "asterisk/channel.h"
/*** DOCUMENTATION
<application name="ReadExten" language="en_US">
<synopsis>
Read an extension into a variable.
</synopsis>
<syntax>
<parameter name="variable" required="true" />
<parameter name="filename">
<para>File to play before reading digits or tone with option <literal>i</literal></para>
</parameter>
<parameter name="context">
<para>Context in which to match extensions.</para>
</parameter>
<parameter name="option">
<optionlist>
<option name="s">
<para>Return immediately if the channel is not answered.</para>
</option>
<option name="i">
<para>Play <replaceable>filename</replaceable> as an indication tone from your
<filename>indications.conf</filename> or a directly specified list of
frequencies and durations.</para>
</option>
<option name="n">
<para>Read digits even if the channel is not answered.</para>
</option>
<option name="p">
<para>The extension entered will be considered complete when a <literal>#</literal>
is entered.</para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>An integer number of seconds to wait for a digit response. If
greater than <literal>0</literal>, that value will override the default timeout.</para>
</parameter>
</syntax>
<description>
<para>Reads a <literal>#</literal> terminated string of digits from the user into the given variable.</para>
<para>Will set READEXTENSTATUS on exit with one of the following statuses:</para>
<variablelist>
<variable name="READEXTENSTATUS">
<value name="OK">
A valid extension exists in ${variable}.
</value>
<value name="TIMEOUT">
No extension was entered in the specified time. Also sets ${variable} to "t".
</value>
<value name="INVALID">
An invalid extension, ${INVALID_EXTEN}, was entered. Also sets ${variable} to "i".
</value>
<value name="SKIP">
Line was not up and the option 's' was specified.
</value>
<value name="ERROR">
Invalid arguments were passed.
</value>
</variable>
</variablelist>
</description>
</application>
***/
enum readexten_option_flags {
OPT_SKIP = (1 << 0),
OPT_INDICATION = (1 << 1),
OPT_NOANSWER = (1 << 2),
OPT_POUND_TO_END = (1 << 3),
};
AST_APP_OPTIONS(readexten_app_options, {
AST_APP_OPTION('s', OPT_SKIP),
AST_APP_OPTION('i', OPT_INDICATION),
AST_APP_OPTION('n', OPT_NOANSWER),
AST_APP_OPTION('p', OPT_POUND_TO_END),
});
static char *app = "ReadExten";
static int readexten_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
char exten[256] = "";
int maxdigits = sizeof(exten) - 1;
int timeout = 0, digit_timeout = 0, x = 0;
char *argcopy = NULL, *status = "";
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
struct ast_tone_zone_sound *ts = NULL;
struct ast_flags flags = {0};
AST_DECLARE_APP_ARGS(arglist,
AST_APP_ARG(variable);
AST_APP_ARG(filename);
AST_APP_ARG(context);
AST_APP_ARG(options);
AST_APP_ARG(timeout);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ReadExten requires at least one argument\n");
pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", "ERROR");
return 0;
}
argcopy = ast_strdupa(data);
AST_STANDARD_APP_ARGS(arglist, argcopy);
if (ast_strlen_zero(arglist.variable)) {
ast_log(LOG_WARNING, "Usage: ReadExten(variable[,filename[,context[,options[,timeout]]]])\n");
pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", "ERROR");
return 0;
}
if (ast_strlen_zero(arglist.filename)) {
arglist.filename = NULL;
}
if (ast_strlen_zero(arglist.context)) {
arglist.context = ast_strdupa(ast_channel_context(chan));
}
if (!ast_strlen_zero(arglist.options)) {
ast_app_parse_options(readexten_app_options, &flags, NULL, arglist.options);
}
if (!ast_strlen_zero(arglist.timeout)) {
timeout = atoi(arglist.timeout);
if (timeout > 0)
timeout *= 1000;
}
if (timeout <= 0)
timeout = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->rtimeoutms : 10000;
digit_timeout = ast_channel_pbx(chan) ? ast_channel_pbx(chan)->dtimeoutms : 5000;
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
if (ast_test_flag(&flags, OPT_INDICATION) && !ast_strlen_zero(arglist.filename)) {
ts = ast_get_indication_tone(ast_channel_zone(chan), arglist.filename);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
}
do {
if (ast_channel_state(chan) != AST_STATE_UP) {
if (ast_test_flag(&flags, OPT_SKIP)) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, arglist.variable, "");
status = "SKIP";
break;
} else if (!ast_test_flag(&flags, OPT_NOANSWER)) {
/* Otherwise answer unless we're supposed to read while on-hook */
res = ast_answer(chan);
}
}
if (res < 0) {
status = "HANGUP";
break;
}
ast_playtones_stop(chan);
ast_stopstream(chan);
if (ts && ts->data[0]) {
res = ast_playtones_start(chan, 0, ts->data, 0);
} else if (arglist.filename) {
if (ast_test_flag(&flags, OPT_INDICATION) && ast_fileexists(arglist.filename, NULL, ast_channel_language(chan)) <= 0) {
/*
* We were asked to play an indication that did not exist in the config.
* If no such file exists, play it as a tonelist. With any luck they won't
* have a file named "350+440.ulaw"
* (but honestly, who would do something so silly?)
*/
res = ast_playtones_start(chan, 0, arglist.filename, 0);
} else {
res = ast_streamfile(chan, arglist.filename, ast_channel_language(chan));
}
}
for (x = 0; x < maxdigits; x++) {
ast_debug(3, "extension so far: '%s', timeout: %d\n", exten, timeout);
res = ast_waitfordigit(chan, timeout);
ast_playtones_stop(chan);
ast_stopstream(chan);
timeout = digit_timeout;
if (res < 1) { /* timeout expired or hangup */
if (ast_check_hangup(chan)) {
status = "HANGUP";
} else if (x == 0) {
pbx_builtin_setvar_helper(chan, arglist.variable, "t");
status = "TIMEOUT";
}
break;
}
if (ast_test_flag(&flags, OPT_POUND_TO_END) && res == '#') {
exten[x] = 0;
break;
}
exten[x] = res;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (!ast_matchmore_extension(chan, arglist.context, exten, 1 /* priority */,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (!ast_exists_extension(chan, arglist.context, exten, 1,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
&& res == '#') {
exten[x] = '\0';
}
break;
}
}
if (!ast_strlen_zero(status))
break;
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (ast_exists_extension(chan, arglist.context, exten, 1,
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
ast_debug(3, "User entered valid extension '%s'\n", exten);
pbx_builtin_setvar_helper(chan, arglist.variable, exten);
status = "OK";
} else {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_debug(3, "User dialed invalid extension '%s' in context '%s' on %s\n", exten, arglist.context, ast_channel_name(chan));
pbx_builtin_setvar_helper(chan, arglist.variable, "i");
pbx_builtin_setvar_helper(chan, "INVALID_EXTEN", exten);
status = "INVALID";
}
} while (0);
Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17 20:41:24 +00:00
if (ts) {
ts = ast_tone_zone_sound_unref(ts);
}
pbx_builtin_setvar_helper(chan, "READEXTENSTATUS", status);
return status[0] == 'H' ? -1 : 0;
}
static int unload_module(void)
{
int res = ast_unregister_application(app);
return res;
}
static int load_module(void)
{
int res = ast_register_application_xml(app, readexten_exec);
return res;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Read and evaluate extension validity");