Files
asterisk/apps/app_parkandannounce.c

248 lines
7.4 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* Author: Ben Miller <bgmiller@dccinc.com>
* With TONS of help from Mark!
*
* 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 ParkAndAnnounce application for Asterisk
*
* \author Ben Miller <bgmiller@dccinc.com>
* \arg With TONS of help from Mark!
*
* \ingroup applications
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/features.h"
#include "asterisk/say.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
/*** DOCUMENTATION
<application name="ParkAndAnnounce" language="en_US">
<synopsis>
Park and Announce.
</synopsis>
<syntax>
<parameter name="announce_template" required="true" argsep=":">
<argument name="announce" required="true">
<para>Colon-separated list of files to announce. The word
<literal>PARKED</literal> will be replaced by a say_digits of the extension in which
the call is parked.</para>
</argument>
<argument name="announce1" multiple="true" />
</parameter>
<parameter name="timeout" required="true">
<para>Time in seconds before the call returns into the return
context.</para>
</parameter>
<parameter name="dial" required="true">
<para>The app_dial style resource to call to make the
announcement. Console/dsp calls the console.</para>
</parameter>
<parameter name="return_context">
<para>The goto-style label to jump the call back into after
timeout. Default <literal>priority+1</literal>.</para>
</parameter>
</syntax>
<description>
<para>Park a call into the parkinglot and announce the call to another channel.</para>
<para>The variable <variable>PARKEDAT</variable> will contain the parking extension
into which the call was placed. Use with the Local channel to allow the dialplan to make
use of this information.</para>
</description>
<see-also>
<ref type="application">Park</ref>
<ref type="application">ParkedCall</ref>
</see-also>
</application>
***/
static char *app = "ParkAndAnnounce";
static int parkandannounce_exec(struct ast_channel *chan, const char *data)
{
int res = -1;
int lot, timeout = 0, dres;
char *dialtech, *tmp[100], buf[13];
int looptemp, i;
char *s;
struct ast_party_id caller_id;
struct ast_channel *dchan;
struct outgoing_helper oh = { 0, };
int outstate;
struct ast_format tmpfmt;
struct ast_format_cap *cap_slin = ast_format_cap_alloc_nolock();
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(template);
AST_APP_ARG(timeout);
AST_APP_ARG(dial);
AST_APP_ARG(return_context);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "ParkAndAnnounce requires arguments: (announce_template,timeout,dial,[return_context])\n");
res = -1;
goto parkcleanup;
}
if (!cap_slin) {
res = -1;
goto parkcleanup;
}
ast_format_cap_add(cap_slin, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
s = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, s);
if (args.timeout)
timeout = atoi(args.timeout) * 1000;
if (ast_strlen_zero(args.dial)) {
ast_log(LOG_WARNING, "PARK: A dial resource must be specified i.e: Console/dsp or DAHDI/g1/5551212\n");
res = -1;
goto parkcleanup;
}
dialtech = strsep(&args.dial, "/");
ast_verb(3, "Dial Tech,String: (%s,%s)\n", dialtech, args.dial);
if (!ast_strlen_zero(args.return_context)) {
ast_clear_flag(chan, AST_FLAG_IN_AUTOLOOP);
ast_parseable_goto(chan, args.return_context);
}
ast_verb(3, "Return Context: (%s,%s,%d) ID: %s\n", ast_channel_context(chan), ast_channel_exten(chan),
ast_channel_priority(chan),
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, ""));
if (!ast_exists_extension(chan, ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan),
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL))) {
ast_verb(3, "Warning: Return Context Invalid, call will return to default|s\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
}
/* Save the CallerID because the masquerade turns chan into a ZOMBIE. */
ast_party_id_init(&caller_id);
ast_channel_lock(chan);
ast_party_id_copy(&caller_id, &ast_channel_caller(chan)->id);
ast_channel_unlock(chan);
/* we are using masq_park here to protect * from touching the channel once we park it. If the channel comes out of timeout
before we are done announcing and the channel is messed with, Kablooeee. So we use Masq to prevent this. */
res = ast_masq_park_call(chan, NULL, timeout, &lot);
if (res) {
/* Parking failed. */
ast_party_id_free(&caller_id);
res = -1;
goto parkcleanup;
}
ast_verb(3, "Call parked in space: %d, timeout: %d, return-context: %s\n",
lot, timeout, args.return_context ? args.return_context : "");
/* Now place the call to the extension */
snprintf(buf, sizeof(buf), "%d", lot);
oh.parent_channel = chan;
oh.vars = ast_variable_new("_PARKEDAT", buf, "");
dchan = __ast_request_and_dial(dialtech, cap_slin, chan, args.dial, 30000,
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
&outstate,
S_COR(caller_id.number.valid, caller_id.number.str, NULL),
S_COR(caller_id.name.valid, caller_id.name.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
&oh);
ast_variables_destroy(oh.vars);
ast_party_id_free(&caller_id);
if (dchan) {
if (ast_channel_state(dchan) == AST_STATE_UP) {
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_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
ast_verb(4, "Channel %s was answered.\n", ast_channel_name(dchan));
} else {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${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_verb(4, "Channel %s was never answered.\n", ast_channel_name(dchan));
ast_log(LOG_WARNING, "PARK: Channel %s was never answered for the announce.\n", ast_channel_name(dchan));
ast_hangup(dchan);
res = -1;
goto parkcleanup;
}
} else {
ast_log(LOG_WARNING, "PARK: Unable to allocate announce channel.\n");
res = -1;
goto parkcleanup;
}
ast_stopstream(dchan);
/* now we have the call placed and are ready to play stuff to it */
ast_verb(4, "Announce Template:%s\n", args.template);
for (looptemp = 0; looptemp < ARRAY_LEN(tmp); looptemp++) {
if ((tmp[looptemp] = strsep(&args.template, ":")) != NULL)
continue;
else
break;
}
for (i = 0; i < looptemp; i++) {
ast_verb(4, "Announce:%s\n", tmp[i]);
if (!strcmp(tmp[i], "PARKED")) {
ast_say_digits(dchan, lot, "", ast_channel_language(dchan));
} else {
dres = ast_streamfile(dchan, tmp[i], ast_channel_language(dchan));
if (!dres) {
dres = ast_waitstream(dchan, "");
} 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_log(LOG_WARNING, "ast_streamfile of %s failed on %s\n", tmp[i], ast_channel_name(dchan));
}
}
}
ast_stopstream(dchan);
ast_hangup(dchan);
parkcleanup:
cap_slin = ast_format_cap_destroy(cap_slin);
return res;
}
static int unload_module(void)
{
return ast_unregister_application(app);
}
static int load_module(void)
{
/* return ast_register_application(app, park_exec); */
return ast_register_application_xml(app, parkandannounce_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Call Parking and Announce Application");