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asterisk/apps/app_privacy.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* 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 Block all calls without Caller*ID, require phone # to be entered
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/utils.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/image.h"
#include "asterisk/callerid.h"
#include "asterisk/app.h"
#include "asterisk/config.h"
/*** DOCUMENTATION
<application name="PrivacyManager" language="en_US">
<synopsis>
Require phone number to be entered, if no CallerID sent
</synopsis>
<syntax>
<parameter name="maxretries">
<para>Total tries caller is allowed to input a callerid. Defaults to <literal>3</literal>.</para>
</parameter>
<parameter name="minlength">
<para>Minimum allowable digits in the input callerid number. Defaults to <literal>10</literal>.</para>
</parameter>
<parameter name="options">
<para>Position reserved for options.</para>
</parameter>
<parameter name="context">
<para>Context to check the given callerid against patterns.</para>
</parameter>
</syntax>
<description>
<para>If no Caller*ID is sent, PrivacyManager answers the channel and asks
the caller to enter their phone number. The caller is given
<replaceable>maxretries</replaceable> attempts to do so. The application does
<emphasis>nothing</emphasis> if Caller*ID was received on the channel.</para>
<para>The application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="PRIVACYMGRSTATUS">
<para>The status of the privacy manager's attempt to collect a phone number from the user.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
</description>
<see-also>
<ref type="application">Zapateller</ref>
</see-also>
</application>
***/
static char *app = "PrivacyManager";
static int privacy_exec(struct ast_channel *chan, const char *data)
{
int res=0;
int retries;
int maxretries = 3;
int minlength = 10;
int x = 0;
char phone[30];
char *parse = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(maxretries);
AST_APP_ARG(minlength);
AST_APP_ARG(options);
AST_APP_ARG(checkcontext);
);
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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
if (chan->caller.id.number.valid
&& !ast_strlen_zero(chan->caller.id.number.str)) {
ast_verb(3, "CallerID number present: Skipping\n");
} else {
/*Answer the channel if it is not already*/
if (chan->_state != AST_STATE_UP) {
if ((res = ast_answer(chan))) {
return -1;
}
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.maxretries)) {
if (sscanf(args.maxretries, "%30d", &x) == 1 && x > 0) {
maxretries = x;
} else {
ast_log(LOG_WARNING, "Invalid max retries argument: '%s'\n", args.maxretries);
}
}
if (!ast_strlen_zero(args.minlength)) {
if (sscanf(args.minlength, "%30d", &x) == 1 && x > 0) {
minlength = x;
} else {
ast_log(LOG_WARNING, "Invalid min length argument: '%s'\n", args.minlength);
}
}
/* Play unidentified call */
res = ast_safe_sleep(chan, 1000);
if (!res) {
res = ast_streamfile(chan, "privacy-unident", chan->language);
}
if (!res) {
res = ast_waitstream(chan, "");
}
/* Ask for 10 digit number, give 3 attempts */
for (retries = 0; retries < maxretries; retries++) {
if (!res) {
res = ast_streamfile(chan, "privacy-prompt", chan->language);
}
if (!res) {
res = ast_waitstream(chan, "");
}
if (!res) {
res = ast_readstring(chan, phone, sizeof(phone) - 1, /* digit timeout ms */ 3200, /* first digit timeout */ 5000, "#");
}
if (res < 0) {
break;
}
/* Make sure we get at least digits */
if (strlen(phone) >= minlength ) {
/* if we have a checkcontext argument, do pattern matching */
if (!ast_strlen_zero(args.checkcontext)) {
if (!ast_exists_extension(NULL, args.checkcontext, phone, 1, NULL)) {
res = ast_streamfile(chan, "privacy-incorrect", chan->language);
if (!res) {
res = ast_waitstream(chan, "");
}
} else {
break;
}
} else {
break;
}
} else {
res = ast_streamfile(chan, "privacy-incorrect", chan->language);
if (!res) {
res = ast_waitstream(chan, "");
}
}
}
/* Got a number, play sounds and send them on their way */
if ((retries < maxretries) && res >= 0) {
res = ast_streamfile(chan, "privacy-thankyou", chan->language);
if (!res) {
res = ast_waitstream(chan, "");
}
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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
/*
* This is a caller entered number that is going to be used locally.
* Therefore, the given number presentation is allowed and should
* be passed out to other channels. This is the point of the
* privacy application.
*/
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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
chan->caller.id.name.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
chan->caller.id.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
chan->caller.id.number.plan = 0;/* Unknown */
ast_set_callerid(chan, phone, "Privacy Manager", 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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14 15:48:36 +00:00
ast_verb(3, "Changed Caller*ID number to '%s'\n", phone);
pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
} else {
pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "FAILED");
}
}
return 0;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
return ast_register_application_xml(app, privacy_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Require phone number to be entered, if no CallerID sent");