asterisk/apps/app_alarmreceiver.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2004 - 2005 Steve Rodgers
*
* Steve Rodgers <hwstar@rodgers.sdcoxmail.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 Central Station Alarm receiver for Ademco Contact ID
* \author Steve Rodgers <hwstar@rodgers.sdcoxmail.com>
*
* *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
*
* Use at your own risk. Please consult the GNU GPL license document included with Asterisk. *
*
* *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING *** WARNING ***
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <math.h>
#include <sys/wait.h>
#include <sys/time.h>
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/ulaw.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/config.h"
#include "asterisk/localtime.h"
#include "asterisk/callerid.h"
#include "asterisk/astdb.h"
#include "asterisk/utils.h"
#define ALMRCV_CONFIG "alarmreceiver.conf"
#define ADEMCO_CONTACT_ID "ADEMCO_CONTACT_ID"
struct event_node{
char data[17];
struct event_node *next;
};
typedef struct event_node event_node_t;
static const char app[] = "AlarmReceiver";
/*** DOCUMENTATION
<application name="AlarmReceiver" language="en_US">
<synopsis>
Provide support for receiving alarm reports from a burglar or fire alarm panel.
</synopsis>
<syntax />
<description>
<para>This application should be called whenever there is an alarm panel calling in to dump its events.
The application will handshake with the alarm panel, and receive events, validate them, handshake them,
and store them until the panel hangs up. Once the panel hangs up, the application will run the system
command specified by the eventcmd setting in <filename>alarmreceiver.conf</filename> and pipe the
events to the standard input of the application.
The configuration file also contains settings for DTMF timing, and for the loudness of the
acknowledgement tones.</para>
<note><para>Only 1 signalling format is supported at this time: Ademco Contact ID.</para></note>
</description>
<see-also>
<ref type="filename">alarmreceiver.conf</ref>
</see-also>
</application>
***/
/* Config Variables */
static int fdtimeout = 2000;
static int sdtimeout = 200;
static int toneloudness = 4096;
static int log_individual_events = 0;
static char event_spool_dir[128] = {'\0'};
static char event_app[128] = {'\0'};
static char db_family[128] = {'\0'};
static char time_stamp_format[128] = {"%a %b %d, %Y @ %H:%M:%S %Z"};
/* Misc variables */
static char event_file[14] = "/event-XXXXXX";
/*
* Attempt to access a database variable and increment it,
* provided that the user defined db-family in alarmreceiver.conf
* The alarmreceiver app will write statistics to a few variables
* in this family if it is defined. If the new key doesn't exist in the
* family, then create it and set its value to 1.
*/
static void database_increment( char *key )
{
int res = 0;
unsigned v;
char value[16];
if (ast_strlen_zero(db_family))
return; /* If not defined, don't do anything */
res = ast_db_get(db_family, key, value, sizeof(value) - 1);
if (res) {
ast_verb(4, "AlarmReceiver: Creating database entry %s and setting to 1\n", key);
/* Guess we have to create it */
res = ast_db_put(db_family, key, "1");
return;
}
sscanf(value, "%30u", &v);
v++;
ast_verb(4, "AlarmReceiver: New value for %s: %u\n", key, v);
snprintf(value, sizeof(value), "%u", v);
res = ast_db_put(db_family, key, value);
if (res)
ast_verb(4, "AlarmReceiver: database_increment write error\n");
return;
}
/*
* Build a MuLaw data block for a single frequency tone
*/
static void make_tone_burst(unsigned char *data, float freq, float loudness, int len, int *x)
{
int i;
float val;
for (i = 0; i < len; i++) {
val = loudness * sin((freq * 2.0 * M_PI * (*x)++)/8000.0);
data[i] = AST_LIN2MU((int)val);
}
/* wrap back around from 8000 */
if (*x >= 8000)
*x = 0;
return;
}
/*
* Send a single tone burst for a specifed duration and frequency.
* Returns 0 if successful
*/
static int send_tone_burst(struct ast_channel *chan, float freq, int duration, int tldn)
{
int res = 0;
int i = 0;
int x = 0;
struct ast_frame *f, wf;
struct {
unsigned char offset[AST_FRIENDLY_OFFSET];
unsigned char buf[640];
} tone_block;
for (;;) {
if (ast_waitfor(chan, -1) < 0) {
res = -1;
break;
}
f = ast_read(chan);
if (!f) {
res = -1;
break;
}
if (f->frametype == AST_FRAME_VOICE) {
wf.frametype = AST_FRAME_VOICE;
ast_format_set(&wf.subclass.format, AST_FORMAT_ULAW, 0);
wf.offset = AST_FRIENDLY_OFFSET;
wf.mallocd = 0;
wf.data.ptr = tone_block.buf;
wf.datalen = f->datalen;
wf.samples = wf.datalen;
make_tone_burst(tone_block.buf, freq, (float) tldn, wf.datalen, &x);
i += wf.datalen / 8;
if (i > duration) {
ast_frfree(f);
break;
}
if (ast_write(chan, &wf)) {
ast_verb(4, "AlarmReceiver: Failed to write frame on %s\n", chan->name);
ast_log(LOG_WARNING, "AlarmReceiver Failed to write frame on %s\n",chan->name);
res = -1;
ast_frfree(f);
break;
}
}
ast_frfree(f);
}
return res;
}
/*
* Receive a string of DTMF digits where the length of the digit string is known in advance. Do not give preferential
* treatment to any digit value, and allow separate time out values to be specified for the first digit and all subsequent
* digits.
*
* Returns 0 if all digits successfully received.
* Returns 1 if a digit time out occurred
* Returns -1 if the caller hung up or there was a channel error.
*
*/
static int receive_dtmf_digits(struct ast_channel *chan, char *digit_string, int length, int fdto, int sdto)
{
int res = 0;
int i = 0;
int r;
struct ast_frame *f;
struct timeval lastdigittime;
lastdigittime = ast_tvnow();
for (;;) {
/* if outa time, leave */
if (ast_tvdiff_ms(ast_tvnow(), lastdigittime) > ((i > 0) ? sdto : fdto)) {
ast_verb(4, "AlarmReceiver: DTMF Digit Timeout on %s\n", chan->name);
ast_debug(1,"AlarmReceiver: DTMF timeout on chan %s\n",chan->name);
res = 1;
break;
}
if ((r = ast_waitfor(chan, -1) < 0)) {
ast_debug(1, "Waitfor returned %d\n", r);
continue;
}
f = ast_read(chan);
if (f == NULL) {
res = -1;
break;
}
/* If they hung up, leave */
if ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP)) {
if (f->data.uint32) {
chan->hangupcause = f->data.uint32;
}
ast_frfree(f);
res = -1;
break;
}
/* if not DTMF, just do it again */
if (f->frametype != AST_FRAME_DTMF) {
ast_frfree(f);
continue;
}
digit_string[i++] = f->subclass.integer; /* save digit */
ast_frfree(f);
/* If we have all the digits we expect, leave */
if(i >= length)
break;
lastdigittime = ast_tvnow();
}
digit_string[i] = '\0'; /* Nul terminate the end of the digit string */
return res;
}
/*
* Write the metadata to the log file
*/
static int write_metadata( FILE *logfile, char *signalling_type, struct ast_channel *chan)
{
int res = 0;
struct timeval t;
struct ast_tm now;
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
char *cl;
char *cn;
char workstring[80];
char timestamp[80];
/* Extract the caller ID location */
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
ast_copy_string(workstring,
S_COR(chan->caller.id.number.valid, chan->caller.id.number.str, ""),
sizeof(workstring));
ast_shrink_phone_number(workstring);
if (ast_strlen_zero(workstring)) {
cl = "<unknown>";
} else {
cl = workstring;
}
cn = S_COR(chan->caller.id.name.valid, chan->caller.id.name.str, "<unknown>");
/* Get the current time */
t = ast_tvnow();
ast_localtime(&t, &now, NULL);
/* Format the time */
ast_strftime(timestamp, sizeof(timestamp), time_stamp_format, &now);
res = fprintf(logfile, "\n\n[metadata]\n\n");
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 (res >= 0) {
res = fprintf(logfile, "PROTOCOL=%s\n", signalling_type);
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 (res >= 0) {
res = fprintf(logfile, "CALLINGFROM=%s\n", cl);
}
if (res >= 0) {
res = fprintf(logfile, "CALLERNAME=%s\n", cn);
}
if (res >= 0) {
res = fprintf(logfile, "TIMESTAMP=%s\n\n", timestamp);
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 (res >= 0) {
res = fprintf(logfile, "[events]\n\n");
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 (res < 0) {
ast_verb(3, "AlarmReceiver: can't write metadata\n");
ast_debug(1,"AlarmReceiver: can't write metadata\n");
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
} else {
res = 0;
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
}
return res;
}
/*
* Write a single event to the log file
*/
static int write_event( FILE *logfile, event_node_t *event)
{
int res = 0;
if (fprintf(logfile, "%s\n", event->data) < 0)
res = -1;
return res;
}
/*
* If we are configured to log events, do so here.
*
*/
static int log_events(struct ast_channel *chan, char *signalling_type, event_node_t *event)
{
int res = 0;
char workstring[sizeof(event_spool_dir)+sizeof(event_file)] = "";
int fd;
FILE *logfile;
event_node_t *elp = event;
if (!ast_strlen_zero(event_spool_dir)) {
/* Make a template */
ast_copy_string(workstring, event_spool_dir, sizeof(workstring));
strncat(workstring, event_file, sizeof(workstring) - strlen(workstring) - 1);
/* Make the temporary file */
fd = mkstemp(workstring);
if (fd == -1) {
ast_verb(3, "AlarmReceiver: can't make temporary file\n");
ast_debug(1,"AlarmReceiver: can't make temporary file\n");
res = -1;
}
if (!res) {
logfile = fdopen(fd, "w");
if (logfile) {
/* Write the file */
res = write_metadata(logfile, signalling_type, chan);
if (!res)
while ((!res) && (elp != NULL)) {
res = write_event(logfile, elp);
elp = elp->next;
}
if (!res) {
if (fflush(logfile) == EOF)
res = -1;
if (!res) {
if (fclose(logfile) == EOF)
res = -1;
}
}
} else
res = -1;
}
}
return res;
}
/*
* This function implements the logic to receive the Ademco contact ID format.
*
* The function will return 0 when the caller hangs up, else a -1 if there was a problem.
*/
static int receive_ademco_contact_id(struct ast_channel *chan, const void *data, int fdto, int sdto, int tldn, event_node_t **ehead)
{
int i, j;
int res = 0;
int checksum;
char event[17];
event_node_t *enew, *elp;
int got_some_digits = 0;
int events_received = 0;
int ack_retries = 0;
static char digit_map[15] = "0123456789*#ABC";
static unsigned char digit_weights[15] = {10,1,2,3,4,5,6,7,8,9,11,12,13,14,15};
database_increment("calls-received");
/* Wait for first event */
ast_verb(4, "AlarmReceiver: Waiting for first event from panel\n");
while (res >= 0) {
if (got_some_digits == 0) {
/* Send ACK tone sequence */
ast_verb(4, "AlarmReceiver: Sending 1400Hz 100ms burst (ACK)\n");
res = send_tone_burst(chan, 1400.0, 100, tldn);
if (!res)
res = ast_safe_sleep(chan, 100);
if (!res) {
ast_verb(4, "AlarmReceiver: Sending 2300Hz 100ms burst (ACK)\n");
res = send_tone_burst(chan, 2300.0, 100, tldn);
}
}
if ( res >= 0)
res = receive_dtmf_digits(chan, event, sizeof(event) - 1, fdto, sdto);
if (res < 0) {
if (events_received == 0) {
/* Hangup with no events received should be logged in the DB */
database_increment("no-events-received");
} else {
if (ack_retries) {
ast_verb(4, "AlarmReceiver: ACK retries during this call: %d\n", ack_retries);
database_increment("ack-retries");
}
}
ast_verb(4, "AlarmReceiver: App exiting...\n");
res = -1;
break;
}
if (res != 0) {
/* Didn't get all of the digits */
ast_verb(2, "AlarmReceiver: Incomplete string: %s, trying again...\n", event);
if (!got_some_digits) {
got_some_digits = (!ast_strlen_zero(event)) ? 1 : 0;
ack_retries++;
}
continue;
}
got_some_digits = 1;
ast_verb(2, "AlarmReceiver: Received Event %s\n", event);
ast_debug(1, "AlarmReceiver: Received event: %s\n", event);
/* Calculate checksum */
for (j = 0, checksum = 0; j < 16; j++) {
for (i = 0; i < sizeof(digit_map); i++) {
if (digit_map[i] == event[j])
break;
}
if (i == 16)
break;
checksum += digit_weights[i];
}
if (i == 16) {
ast_verb(2, "AlarmReceiver: Bad DTMF character %c, trying again\n", event[j]);
continue; /* Bad character */
}
/* Checksum is mod(15) of the total */
checksum = checksum % 15;
if (checksum) {
database_increment("checksum-errors");
ast_verb(2, "AlarmReceiver: Nonzero checksum\n");
ast_debug(1, "AlarmReceiver: Nonzero checksum\n");
continue;
}
/* Check the message type for correctness */
if (strncmp(event + 4, "18", 2)) {
if (strncmp(event + 4, "98", 2)) {
database_increment("format-errors");
ast_verb(2, "AlarmReceiver: Wrong message type\n");
ast_debug(1, "AlarmReceiver: Wrong message type\n");
continue;
}
}
events_received++;
/* Queue the Event */
if (!(enew = ast_calloc(1, sizeof(*enew)))) {
res = -1;
break;
}
enew->next = NULL;
ast_copy_string(enew->data, event, sizeof(enew->data));
/*
* Insert event onto end of list
*/
if (*ehead == NULL)
*ehead = enew;
else {
for(elp = *ehead; elp->next != NULL; elp = elp->next)
;
elp->next = enew;
}
if (res > 0)
res = 0;
/* Let the user have the option of logging the single event before sending the kissoff tone */
if ((res == 0) && (log_individual_events))
res = log_events(chan, ADEMCO_CONTACT_ID, enew);
/* Wait 200 msec before sending kissoff */
if (res == 0)
res = ast_safe_sleep(chan, 200);
/* Send the kissoff tone */
if (res == 0)
res = send_tone_burst(chan, 1400.0, 900, tldn);
}
return res;
}
/*
* This is the main function called by Asterisk Core whenever the App is invoked in the extension logic.
* This function will always return 0.
*/
static int alarmreceiver_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
event_node_t *elp, *efree;
char signalling_type[64] = "";
event_node_t *event_head = NULL;
/* Set write and read formats to ULAW */
ast_verb(4, "AlarmReceiver: Setting read and write formats to ULAW\n");
if (ast_set_write_format_by_id(chan,AST_FORMAT_ULAW)) {
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set write format to Mu-law on %s\n",chan->name);
return -1;
}
if (ast_set_read_format_by_id(chan,AST_FORMAT_ULAW)) {
ast_log(LOG_WARNING, "AlarmReceiver: Unable to set read format to Mu-law on %s\n",chan->name);
return -1;
}
/* Set default values for this invocation of the application */
ast_copy_string(signalling_type, ADEMCO_CONTACT_ID, sizeof(signalling_type));
/* Answer the channel if it is not already */
ast_verb(4, "AlarmReceiver: Answering channel\n");
if (chan->_state != AST_STATE_UP) {
if ((res = ast_answer(chan)))
return -1;
}
/* Wait for the connection to settle post-answer */
ast_verb(4, "AlarmReceiver: Waiting for connection to stabilize\n");
res = ast_safe_sleep(chan, 1250);
/* Attempt to receive the events */
if (!res) {
/* Determine the protocol to receive in advance */
/* Note: Ademco contact is the only one supported at this time */
/* Others may be added later */
if(!strcmp(signalling_type, ADEMCO_CONTACT_ID))
receive_ademco_contact_id(chan, data, fdtimeout, sdtimeout, toneloudness, &event_head);
else
res = -1;
}
/* Events queued by receiver, write them all out here if so configured */
if ((!res) && (log_individual_events == 0))
res = log_events(chan, signalling_type, event_head);
/*
* Do we exec a command line at the end?
*/
if ((!res) && (!ast_strlen_zero(event_app)) && (event_head)) {
ast_debug(1,"Alarmreceiver: executing: %s\n", event_app);
ast_safe_system(event_app);
}
/*
* Free up the data allocated in our linked list
*/
for (elp = event_head; (elp != NULL);) {
efree = elp;
elp = elp->next;
ast_free(efree);
}
return 0;
}
/*
* Load the configuration from the configuration file
*/
static int load_config(void)
{
struct ast_config *cfg;
const char *p;
struct ast_flags config_flags = { 0 };
/* Read in the config file */
cfg = ast_config_load(ALMRCV_CONFIG, config_flags);
if (!cfg) {
ast_verb(4, "AlarmReceiver: No config file\n");
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\n", ALMRCV_CONFIG);
return 0;
} else {
p = ast_variable_retrieve(cfg, "general", "eventcmd");
if (p) {
ast_copy_string(event_app, p, sizeof(event_app));
event_app[sizeof(event_app) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "loudness");
if (p) {
toneloudness = atoi(p);
if(toneloudness < 100)
toneloudness = 100;
if(toneloudness > 8192)
toneloudness = 8192;
}
p = ast_variable_retrieve(cfg, "general", "fdtimeout");
if (p) {
fdtimeout = atoi(p);
if(fdtimeout < 1000)
fdtimeout = 1000;
if(fdtimeout > 10000)
fdtimeout = 10000;
}
p = ast_variable_retrieve(cfg, "general", "sdtimeout");
if (p) {
sdtimeout = atoi(p);
if(sdtimeout < 110)
sdtimeout = 110;
if(sdtimeout > 4000)
sdtimeout = 4000;
}
p = ast_variable_retrieve(cfg, "general", "logindividualevents");
if (p)
log_individual_events = ast_true(p);
p = ast_variable_retrieve(cfg, "general", "eventspooldir");
if (p) {
ast_copy_string(event_spool_dir, p, sizeof(event_spool_dir));
event_spool_dir[sizeof(event_spool_dir) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "timestampformat");
if (p) {
ast_copy_string(time_stamp_format, p, sizeof(time_stamp_format));
time_stamp_format[sizeof(time_stamp_format) - 1] = '\0';
}
p = ast_variable_retrieve(cfg, "general", "db-family");
if (p) {
ast_copy_string(db_family, p, sizeof(db_family));
db_family[sizeof(db_family) - 1] = '\0';
}
ast_config_destroy(cfg);
}
return 1;
}
/*
* These functions are required to implement an Asterisk App.
*/
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
if (load_config()) {
if (ast_register_application_xml(app, alarmreceiver_exec))
return AST_MODULE_LOAD_FAILURE;
return AST_MODULE_LOAD_SUCCESS;
} else
return AST_MODULE_LOAD_DECLINE;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Alarm Receiver for Asterisk");