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asterisk/apps/app_dial.c
Russell Bryant fc9d3ba21b Merged revisions 13550 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r13550 | russell | 2006-03-19 04:59:55 -0500 (Sun, 19 Mar 2006) | 4 lines

revert the change made in revision 12927 in favor of keeping the original
behavior of the option.  The documentation has now been updated to reflect
the actual behavior.  (issue #6523)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-19 10:11:29 +00:00

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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 dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
*
* \author Mark Spencer <markster@digium.com>
*
* \ingroup applications
*/
#include <stdlib.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <sys/time.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/say.h"
#include "asterisk/config.h"
#include "asterisk/features.h"
#include "asterisk/musiconhold.h"
#include "asterisk/callerid.h"
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/rtp.h"
#include "asterisk/manager.h"
#include "asterisk/privacy.h"
#include "asterisk/stringfields.h"
static char *tdesc = "Dialing Application";
static char *app = "Dial";
static char *synopsis = "Place a call and connect to the current channel";
static char *descrip =
" Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):\n"
"This applicaiton will place calls to one or more specified channels. As soon\n"
"as one of the requested channels answers, the originating channel will be\n"
"answered, if it has not already been answered. These two channels will then\n"
"be active in a bridged call. All other channels that were requested will then\n"
"be hung up.\n"
" Unless there is a timeout specified, the Dial application will wait\n"
"indefinitely until one of the called channels answers, the user hangs up, or\n"
"if all of the called channels are busy or unavailable. Dialplan executing will\n"
"continue if no requested channels can be called, or if the timeout expires.\n\n"
" This application sets the following channel variables upon completion:\n"
" DIALEDTIME - This is the time from dialing a channel until when it\n"
" is disconnected.\n"
" ANSWEREDTIME - This is the amount of time for actual call.\n"
" DIALSTATUS - This is the status of the call:\n"
" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
" DONTCALL | TORTURE\n"
" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
"wants to send the caller to the 'torture' script.\n"
" This application will report normal termination if the originating channel\n"
"hangs up, or if the call is bridged and either of the parties in the bridge\n"
"ends the call.\n"
" The optional URL will be sent to the called party if the channel supports it.\n"
" If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
"application will be put into that group (as in Set(GROUP()=...).\n\n"
" Options:\n"
" A(x) - Play an announcement to the called party, using 'x' as the file.\n"
" C - Reset the CDR for this call.\n"
" d - Allow the calling user to dial a 1 digit extension while waiting for\n"
" a call to be answered. Exit to that extension if it exists in the\n"
" current context, or the context defined in the EXITCONTEXT variable,\n"
" if it exists.\n"
" D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
" party has answered, but before the call gets bridged. The 'called'\n"
" DTMF string is sent to the called party, and the 'calling' DTMF\n"
" string is sent to the calling party. Both parameters can be used\n"
" alone.\n"
" f - Force the callerid of the *calling* channel to be set as the\n"
" extension associated with the channel using a dialplan 'hint'.\n"
" For example, some PSTNs do not allow CallerID to be set to anything\n"
" other than the number assigned to the caller.\n"
" g - Proceed with dialplan execution at the current extension if the\n"
" destination channel hangs up.\n"
" G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
" the specified priority and the called party to the specified priority+1.\n"
" Optionally, an extension, or extension and context may be specified. \n"
" Otherwise, the current extension is used.\n"
" h - Allow the called party to hang up by sending the '*' DTMF digit.\n"
" H - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
" j - Jump to priority n+101 if all of the requested channels were busy.\n"
" L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
" left. Repeat the warning every 'z' ms. The following special\n"
" variables can be used with this option:\n"
" * LIMIT_PLAYAUDIO_CALLER yes|no (default yes)\n"
" Play sounds to the caller.\n"
" * LIMIT_PLAYAUDIO_CALLEE yes|no\n"
" Play sounds to the callee.\n"
" * LIMIT_TIMEOUT_FILE File to play when time is up.\n"
" * LIMIT_CONNECT_FILE File to play when call begins.\n"
" * LIMIT_WARNING_FILE File to play as warning if 'y' is defined.\n"
" The default is to say the time remaining.\n"
" m([class]) - Provide hold music to the calling party until a requested\n"
" channel answers. A specific MusicOnHold class can be\n"
" specified.\n"
" M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
" to the calling channel. Arguments can be specified to the Macro\n"
" using '^' as a delimeter. The Macro can set the variable\n"
" MACRO_RESULT to specify the following actions after the Macro is\n"
" finished executing.\n"
" * ABORT Hangup both legs of the call.\n"
" * CONGESTION Behave as if line congestion was encountered.\n"
" * BUSY Behave as if a busy signal was encountered. This will also\n"
" have the application jump to priority n+101 if the\n"
" 'j' option is set.\n"
" * CONTINUE Hangup the called party and allow the calling party\n"
" to continue dialplan execution at the next priority.\n"
" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
" specified priority. Optionally, an extension, or\n"
" extension and priority can be specified.\n"
" n - This option is a modifier for the screen/privacy mode. It specifies\n"
" that no introductions are to be saved in the priv-callerintros\n"
" directory.\n"
" N - This option is a modifier for the screen/privacy mode. It specifies\n"
" that if callerID is present, do not screen the call.\n"
" o - Specify that the CallerID that was present on the *calling* channel\n"
" be set as the CallerID on the *called* channel. This was the\n"
" behavior of Asterisk 1.0 and earlier.\n"
" p - This option enables screening mode. This is basically Privacy mode\n"
" without memory.\n"
" P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
" it is provided. The current extension is used if a database\n"
" family/key is not specified.\n"
" r - Indicate ringing to the calling party. Pass no audio to the calling\n"
" party until the called channel has answered.\n"
" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
" answered the call.\n"
" t - Allow the called party to transfer the calling party by sending the\n"
" DTMF sequence defined in features.conf.\n"
" T - Allow the calling party to transfer the called party by sending the\n"
" DTMF sequence defined in features.conf.\n"
" w - Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch recording in features.conf.\n"
" W - Allow the calling party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch recording in features.conf.\n";
/* RetryDial App by Anthony Minessale II <anthmct@yahoo.com> Jan/2005 */
static char *rapp = "RetryDial";
static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
static char *rdescrip =
" RetryDial(announce|sleep|retries|dialargs): This application will attempt to\n"
"place a call using the normal Dial application. If no channel can be reached,\n"
"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
"seconds before retying the call. After 'retires' number of attempts, the\n"
"calling channel will continue at the next priority in the dialplan. If the\n"
"'retries' setting is set to 0, this application will retry endlessly.\n"
" While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
"extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
"one, The call will jump to that extension immediately.\n"
" The 'dialargs' are specified in the same format that arguments are provided\n"
"to the Dial application.\n";
enum {
OPT_ANNOUNCE = (1 << 0),
OPT_RESETCDR = (1 << 1),
OPT_DTMF_EXIT = (1 << 2),
OPT_SENDDTMF = (1 << 3),
OPT_FORCECLID = (1 << 4),
OPT_GO_ON = (1 << 5),
OPT_CALLEE_HANGUP = (1 << 6),
OPT_CALLER_HANGUP = (1 << 7),
OPT_PRIORITY_JUMP = (1 << 8),
OPT_DURATION_LIMIT = (1 << 9),
OPT_MUSICBACK = (1 << 10),
OPT_CALLEE_MACRO = (1 << 11),
OPT_SCREEN_NOINTRO = (1 << 12),
OPT_SCREEN_NOCLID = (1 << 13),
OPT_ORIGINAL_CLID = (1 << 14),
OPT_SCREENING = (1 << 15),
OPT_PRIVACY = (1 << 16),
OPT_RINGBACK = (1 << 17),
OPT_DURATION_STOP = (1 << 18),
OPT_CALLEE_TRANSFER = (1 << 19),
OPT_CALLER_TRANSFER = (1 << 20),
OPT_CALLEE_MONITOR = (1 << 21),
OPT_CALLER_MONITOR = (1 << 22),
OPT_GOTO = (1 << 23),
} dial_exec_option_flags;
#define DIAL_STILLGOING (1 << 30)
#define DIAL_NOFORWARDHTML (1 << 31)
enum {
OPT_ARG_ANNOUNCE = 0,
OPT_ARG_SENDDTMF,
OPT_ARG_GOTO,
OPT_ARG_DURATION_LIMIT,
OPT_ARG_MUSICBACK,
OPT_ARG_CALLEE_MACRO,
OPT_ARG_PRIVACY,
OPT_ARG_DURATION_STOP,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE,
} dial_exec_option_args;
AST_APP_OPTIONS(dial_exec_options, {
AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
AST_APP_OPTION('C', OPT_RESETCDR),
AST_APP_OPTION('d', OPT_DTMF_EXIT),
AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
AST_APP_OPTION('f', OPT_FORCECLID),
AST_APP_OPTION('g', OPT_GO_ON),
AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
AST_APP_OPTION('H', OPT_CALLER_HANGUP),
AST_APP_OPTION('j', OPT_PRIORITY_JUMP),
AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
AST_APP_OPTION('p', OPT_SCREENING),
AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
AST_APP_OPTION('r', OPT_RINGBACK),
AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
AST_APP_OPTION('W', OPT_CALLER_MONITOR),
});
/* We define a custom "local user" structure because we
use it not only for keeping track of what is in use but
also for keeping track of who we're dialing. */
struct dial_localuser {
struct ast_channel *chan;
unsigned int flags;
int forwards;
struct dial_localuser *next;
};
LOCAL_USER_DECL;
static void hanguptree(struct dial_localuser *outgoing, struct ast_channel *exception)
{
/* Hang up a tree of stuff */
struct dial_localuser *oo;
while (outgoing) {
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception))
ast_hangup(outgoing->chan);
oo = outgoing;
outgoing=outgoing->next;
free(oo);
}
}
#define AST_MAX_FORWARDS 8
#define AST_MAX_WATCHERS 256
#define HANDLE_CAUSE(cause, chan) do { \
switch(cause) { \
case AST_CAUSE_BUSY: \
if (chan->cdr) \
ast_cdr_busy(chan->cdr); \
numbusy++; \
break; \
case AST_CAUSE_CONGESTION: \
if (chan->cdr) \
ast_cdr_failed(chan->cdr); \
numcongestion++; \
break; \
case AST_CAUSE_UNREGISTERED: \
if (chan->cdr) \
ast_cdr_failed(chan->cdr); \
numnochan++; \
break; \
default: \
numnochan++; \
break; \
} \
} while (0)
static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
{
char rexten[2] = { exten, '\0' };
if (context) {
if (!ast_goto_if_exists(chan, context, rexten, pri))
return 1;
} else {
if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
return 1;
else if (!ast_strlen_zero(chan->macrocontext)) {
if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
return 1;
}
}
return 0;
}
static char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
{
char *context;
char *exten;
if (!ast_strlen_zero(chan->macrocontext))
context = chan->macrocontext;
else
context = chan->context;
if (!ast_strlen_zero(chan->macroexten))
exten = chan->macroexten;
else
exten = chan->exten;
if (ast_get_hint(NULL, 0, name, namelen, chan, context, exten))
return name;
else
return "";
}
static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
{
manager_event(EVENT_FLAG_CALL, "Dial",
"Source: %s\r\n"
"Destination: %s\r\n"
"CallerID: %s\r\n"
"CallerIDName: %s\r\n"
"SrcUniqueID: %s\r\n"
"DestUniqueID: %s\r\n",
src->name, dst->name, src->cid.cid_num ? src->cid.cid_num : "<unknown>",
src->cid.cid_name ? src->cid.cid_name : "<unknown>", src->uniqueid,
dst->uniqueid);
}
static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_localuser *outgoing, int *to, struct ast_flags *peerflags, int *sentringing, char *status, size_t statussize, int busystart, int nochanstart, int congestionstart, int priority_jump, int *result)
{
struct dial_localuser *o;
int found;
int numlines;
int numbusy = busystart;
int numcongestion = congestionstart;
int numnochan = nochanstart;
int prestart = busystart + congestionstart + nochanstart;
int cause;
int orig = *to;
struct ast_frame *f;
struct ast_channel *peer = NULL;
struct ast_channel *watchers[AST_MAX_WATCHERS];
int pos;
int single;
struct ast_channel *winner;
const char *context = NULL;
char cidname[AST_MAX_EXTENSION];
single = (outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK));
if (single) {
/* Turn off hold music, etc */
ast_deactivate_generator(in);
/* If we are calling a single channel, make them compatible for in-band tone purpose */
ast_channel_make_compatible(outgoing->chan, in);
}
while (*to && !peer) {
o = outgoing;
found = -1;
pos = 1;
numlines = prestart;
watchers[0] = in;
while (o) {
/* Keep track of important channels */
if (ast_test_flag(o, DIAL_STILLGOING) && o->chan) {
watchers[pos++] = o->chan;
found = 1;
}
o = o->next;
numlines++;
}
if (found < 0) {
if (numlines == (numbusy + numcongestion + numnochan)) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_2 "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, numbusy, numcongestion, numnochan);
if (numbusy)
strcpy(status, "BUSY");
else if (numcongestion)
strcpy(status, "CONGESTION");
else if (numnochan)
strcpy(status, "CHANUNAVAIL");
if (ast_opt_priority_jumping || priority_jump)
ast_goto_if_exists(in, in->context, in->exten, in->priority + 101);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, numbusy, numcongestion, numnochan);
}
*to = 0;
return NULL;
}
winner = ast_waitfor_n(watchers, pos, to);
o = outgoing;
while (o) {
if (ast_test_flag(o, DIAL_STILLGOING) && o->chan && (o->chan->_state == AST_STATE_UP)) {
if (!peer) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
peer = o->chan;
ast_copy_flags(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
DIAL_NOFORWARDHTML);
}
} else if (o->chan && (o->chan == winner)) {
if (!ast_strlen_zero(o->chan->call_forward)) {
char tmpchan[256];
char *stuff;
char *tech;
const char *forward_context;
ast_copy_string(tmpchan, o->chan->call_forward, sizeof(tmpchan));
if ((stuff = strchr(tmpchan, '/'))) {
*stuff = '\0';
stuff++;
tech = tmpchan;
} else {
forward_context = pbx_builtin_getvar_helper(o->chan, "FORWARD_CONTEXT");
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", o->chan->call_forward, forward_context ? forward_context : o->chan->context);
stuff = tmpchan;
tech = "Local";
}
/* Before processing channel, go ahead and check for forwarding */
o->forwards++;
if (o->forwards < AST_MAX_FORWARDS) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, o->chan->name);
/* Setup parameters */
o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
if (!o->chan)
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", o->chan->name);
cause = AST_CAUSE_CONGESTION;
o->chan = NULL;
}
if (!o->chan) {
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(cause, in);
} else {
ast_rtp_make_compatible(o->chan, in);
if (o->chan->cid.cid_num)
free(o->chan->cid.cid_num);
o->chan->cid.cid_num = NULL;
if (o->chan->cid.cid_name)
free(o->chan->cid.cid_name);
o->chan->cid.cid_name = NULL;
if (ast_test_flag(o, OPT_FORCECLID)) {
char *newcid = NULL;
if (!ast_strlen_zero(in->macroexten))
newcid = in->macroexten;
else
newcid = in->exten;
o->chan->cid.cid_num = strdup(newcid);
ast_string_field_set(o->chan, accountcode, winner->accountcode);
o->chan->cdrflags = winner->cdrflags;
if (!o->chan->cid.cid_num)
ast_log(LOG_WARNING, "Out of memory\n");
} else {
if (in->cid.cid_num) {
o->chan->cid.cid_num = strdup(in->cid.cid_num);
if (!o->chan->cid.cid_num)
ast_log(LOG_WARNING, "Out of memory\n");
}
if (in->cid.cid_name) {
o->chan->cid.cid_name = strdup(in->cid.cid_name);
if (!o->chan->cid.cid_name)
ast_log(LOG_WARNING, "Out of memory\n");
}
ast_string_field_set(o->chan, accountcode, in->accountcode);
o->chan->cdrflags = in->cdrflags;
}
if (in->cid.cid_ani) {
if (o->chan->cid.cid_ani)
free(o->chan->cid.cid_ani);
o->chan->cid.cid_ani = strdup(in->cid.cid_ani);
if (!o->chan->cid.cid_ani)
ast_log(LOG_WARNING, "Out of memory\n");
}
if (o->chan->cid.cid_rdnis)
free(o->chan->cid.cid_rdnis);
if (!ast_strlen_zero(in->macroexten))
o->chan->cid.cid_rdnis = strdup(in->macroexten);
else
o->chan->cid.cid_rdnis = strdup(in->exten);
if (ast_call(o->chan, tmpchan, 0)) {
ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
ast_clear_flag(o, DIAL_STILLGOING);
ast_hangup(o->chan);
o->chan = NULL;
numnochan++;
} else {
senddialevent(in, o->chan);
/* After calling, set callerid to extension */
if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID))
ast_set_callerid(o->chan, ast_strlen_zero(in->macroexten) ? in->exten : in->macroexten, get_cid_name(cidname, sizeof(cidname), in), NULL);
}
}
/* Hangup the original channel now, in case we needed it */
ast_hangup(winner);
continue;
}
f = ast_read(winner);
if (f) {
if (f->frametype == AST_FRAME_CONTROL) {
switch(f->subclass) {
case AST_CONTROL_ANSWER:
/* This is our guy if someone answered. */
if (!peer) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
peer = o->chan;
ast_copy_flags(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
DIAL_NOFORWARDHTML);
}
/* If call has been answered, then the eventual hangup is likely to be normal hangup */
in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
o->chan->hangupcause = AST_CAUSE_NORMAL_CLEARING;
break;
case AST_CONTROL_BUSY:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(AST_CAUSE_BUSY, in);
break;
case AST_CONTROL_CONGESTION:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(AST_CAUSE_CONGESTION, in);
break;
case AST_CONTROL_RINGING:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
(*sentringing)++;
}
break;
case AST_CONTROL_PROGRESS:
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", o->chan->name,in->name);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
break;
case AST_CONTROL_VIDUPDATE:
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
ast_indicate(in, AST_CONTROL_VIDUPDATE);
break;
case AST_CONTROL_PROCEEDING:
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
case AST_CONTROL_HOLD:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Call on %s placed on hold\n", o->chan->name);
ast_indicate(in, AST_CONTROL_HOLD);
break;
case AST_CONTROL_UNHOLD:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Call on %s left from hold\n", o->chan->name);
ast_indicate(in, AST_CONTROL_UNHOLD);
break;
case AST_CONTROL_OFFHOOK:
case AST_CONTROL_FLASH:
/* Ignore going off hook and flash */
break;
case -1:
if (!ast_test_flag(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
ast_indicate(in, -1);
(*sentringing) = 0;
}
break;
default:
if (option_debug)
ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
}
} else if (single && (f->frametype == AST_FRAME_VOICE) &&
!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
if (ast_write(in, f))
ast_log(LOG_WARNING, "Unable to forward voice frame\n");
} else if (single && (f->frametype == AST_FRAME_IMAGE) &&
!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
if (ast_write(in, f))
ast_log(LOG_WARNING, "Unable to forward image\n");
} else if (single && (f->frametype == AST_FRAME_TEXT) &&
!(ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK))) {
if (ast_write(in, f))
ast_log(LOG_WARNING, "Unable to send text\n");
} else if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML))
if(ast_channel_sendhtml(in, f->subclass, f->data, f->datalen) == -1)
ast_log(LOG_WARNING, "Unable to send URL\n");
ast_frfree(f);
} else {
in->hangupcause = o->chan->hangupcause;
ast_hangup(o->chan);
o->chan = NULL;
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(in->hangupcause, in);
}
}
o = o->next;
}
if (winner == in) {
f = ast_read(in);
#if 0
if (f && (f->frametype != AST_FRAME_VOICE))
printf("Frame type: %d, %d\n", f->frametype, f->subclass);
else if (!f || (f->frametype != AST_FRAME_VOICE))
printf("Hangup received on %s\n", in->name);
#endif
if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
/* Got hung up */
*to = -1;
strcpy(status, "CANCEL");
if (f)
ast_frfree(f);
return NULL;
}
if (f && (f->frametype == AST_FRAME_DTMF)) {
if (ast_test_flag(peerflags, OPT_DTMF_EXIT)) {
context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
if (onedigit_goto(in, context, (char) f->subclass, 1)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
*to=0;
*result = f->subclass;
strcpy(status, "CANCEL");
ast_frfree(f);
return NULL;
}
}
if (ast_test_flag(peerflags, OPT_CALLER_HANGUP) &&
(f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
*to=0;
strcpy(status, "CANCEL");
ast_frfree(f);
return NULL;
}
}
/* Forward HTML stuff */
if (single && f && (f->frametype == AST_FRAME_HTML) && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML))
if(ast_channel_sendhtml(outgoing->chan, f->subclass, f->data, f->datalen) == -1)
ast_log(LOG_WARNING, "Unable to send URL\n");
if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF))) {
if (ast_write(outgoing->chan, f))
ast_log(LOG_WARNING, "Unable to forward voice\n");
}
if (single && (f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_VIDUPDATE)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", in->name,outgoing->chan->name);
ast_indicate(outgoing->chan, AST_CONTROL_VIDUPDATE);
}
ast_frfree(f);
}
if (!*to && (option_verbose > 2))
ast_verbose(VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
}
return peer;
}
static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags *peerflags)
{
int res=-1;
struct localuser *u;
char *tech, *number, *rest, *cur;
char privcid[256];
char privintro[1024];
struct dial_localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
int to;
int numbusy = 0;
int numcongestion = 0;
int numnochan = 0;
int cause;
char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION];
char cidname[AST_MAX_EXTENSION];
char toast[80];
char *newnum;
char *l;
int privdb_val=0;
unsigned int calldurationlimit=0;
struct ast_bridge_config config;
long timelimit = 0;
long play_warning = 0;
long warning_freq=0;
const char *warning_sound=NULL;
const char *end_sound=NULL;
const char *start_sound=NULL;
char *dtmfcalled=NULL, *dtmfcalling=NULL;
const char *var;
char status[256];
int play_to_caller=0,play_to_callee=0;
int sentringing=0, moh=0;
const char *outbound_group = NULL;
const char *macro_result = NULL;
char *macro_transfer_dest = NULL;
int digit = 0, result = 0;
time_t start_time, answer_time, end_time;
struct ast_app *app = NULL;
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(peers);
AST_APP_ARG(timeout);
AST_APP_ARG(options);
AST_APP_ARG(url);
);
struct ast_flags opts = { 0, };
char *opt_args[OPT_ARG_ARRAY_SIZE];
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
return -1;
}
LOCAL_USER_ADD(u);
if (!(parse = ast_strdupa(data))) {
LOCAL_USER_REMOVE(u);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.options)) {
if (ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options)) {
LOCAL_USER_REMOVE(u);
return -1;
}
}
if (ast_strlen_zero(args.peers)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
LOCAL_USER_REMOVE(u);
return -1;
}
if (ast_test_flag(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
calldurationlimit = atoi(opt_args[OPT_ARG_DURATION_STOP]);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n", calldurationlimit);
}
if (ast_test_flag(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
parse = opt_args[OPT_ARG_SENDDTMF];
dtmfcalled = strsep(&parse, ":");
dtmfcalling = parse;
}
if (ast_test_flag(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
char *limit_str, *warning_str, *warnfreq_str;
parse = opt_args[OPT_ARG_DURATION_LIMIT];
limit_str = strsep(&parse, ":");
warning_str = strsep(&parse, ":");
warnfreq_str = parse;
timelimit = atol(limit_str);
if (warning_str)
play_warning = atol(warning_str);
if (warnfreq_str)
warning_freq = atol(warnfreq_str);
if (!timelimit) {
timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0;
warning_sound = NULL;
}
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
play_to_caller = var ? ast_true(var) : 1;
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLEE");
play_to_callee = var ? ast_true(var) : 0;
if (!play_to_caller && !play_to_callee)
play_to_caller=1;
var = pbx_builtin_getvar_helper(chan,"LIMIT_WARNING_FILE");
warning_sound = var ? var : "timeleft";
var = pbx_builtin_getvar_helper(chan,"LIMIT_TIMEOUT_FILE");
end_sound = var ? var : NULL;
var = pbx_builtin_getvar_helper(chan,"LIMIT_CONNECT_FILE");
start_sound = var ? var : NULL;
/* undo effect of S(x) in case they are both used */
calldurationlimit = 0;
/* more efficient do it like S(x) does since no advanced opts*/
if (!play_warning && !start_sound && !end_sound && timelimit) {
calldurationlimit = timelimit/1000;
timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0;
} else if (option_verbose > 2) {
ast_verbose(VERBOSE_PREFIX_3 "Limit Data for this call:\n");
ast_verbose(VERBOSE_PREFIX_3 "- timelimit = %ld\n", timelimit);
ast_verbose(VERBOSE_PREFIX_3 "- play_warning = %ld\n", play_warning);
ast_verbose(VERBOSE_PREFIX_3 "- play_to_caller= %s\n", play_to_caller ? "yes" : "no");
ast_verbose(VERBOSE_PREFIX_3 "- play_to_callee= %s\n", play_to_callee ? "yes" : "no");
ast_verbose(VERBOSE_PREFIX_3 "- warning_freq = %ld\n", warning_freq);
ast_verbose(VERBOSE_PREFIX_3 "- start_sound = %s\n", start_sound ? start_sound : "UNDEF");
ast_verbose(VERBOSE_PREFIX_3 "- warning_sound = %s\n", warning_sound ? warning_sound : "UNDEF");
ast_verbose(VERBOSE_PREFIX_3 "- end_sound = %s\n", end_sound ? end_sound : "UNDEF");
}
}
if (ast_test_flag(&opts, OPT_RESETCDR) && chan->cdr)
ast_cdr_reset(chan->cdr, NULL);
if (ast_test_flag(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
if (ast_test_flag(&opts, OPT_PRIVACY) || ast_test_flag(&opts, OPT_SCREENING)) {
char callerid[60];
l = chan->cid.cid_num;
if (!ast_strlen_zero(l)) {
ast_shrink_phone_number(l);
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Privacy DB is '%s', clid is '%s'\n",
opt_args[OPT_ARG_PRIVACY], l);
privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
}
else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Privacy Screening, clid is '%s'\n", l);
privdb_val = AST_PRIVACY_UNKNOWN;
}
} else {
char *tnam, *tn2;
tnam = ast_strdupa(chan->name);
/* clean the channel name so slashes don't try to end up in disk file name */
for(tn2 = tnam; *tn2; tn2++) {
if( *tn2=='/')
*tn2 = '='; /* any other chars to be afraid of? */
}
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Privacy-- callerid is empty\n");
snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
l = callerid;
privdb_val = AST_PRIVACY_UNKNOWN;
}
ast_copy_string(privcid,l,sizeof(privcid));
if( strncmp(privcid,"NOCALLERID",10) != 0 && ast_test_flag(&opts, OPT_SCREEN_NOCLID) ) { /* if callerid is set, and ast_test_flag(&opts, OPT_SCREEN_NOCLID) is set also */
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "CallerID set (%s); N option set; Screening should be off\n", privcid);
privdb_val = AST_PRIVACY_ALLOW;
}
else if(ast_test_flag(&opts, OPT_SCREEN_NOCLID) && strncmp(privcid,"NOCALLERID",10) == 0 ) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "CallerID blank; N option set; Screening should happen; dbval is %d\n", privdb_val);
}
if(privdb_val == AST_PRIVACY_DENY ) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
res=0;
goto out;
}
else if(privdb_val == AST_PRIVACY_KILL ) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
res = 0;
goto out; /* Is this right? */
}
else if(privdb_val == AST_PRIVACY_TORTURE ) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
res = 0;
goto out; /* is this right??? */
}
else if(privdb_val == AST_PRIVACY_UNKNOWN ) {
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there-- this should be done before the
call is actually dialed */
/* make sure the priv-callerintros dir exists? */
snprintf(privintro,sizeof(privintro), "priv-callerintros/%s", privcid);
if( ast_fileexists(privintro,NULL,NULL ) > 0 && strncmp(privcid,"NOCALLERID",10) != 0) {
/* the DELUX version of this code would allow this caller the
option to hear and retape their previously recorded intro.
*/
}
else {
int duration; /* for feedback from play_and_wait */
/* the file doesn't exist yet. Let the caller submit his
vocal intro for posterity */
/* priv-recordintro script:
"At the tone, please say your name:"
*/
ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the privintro file */
}
}
}
/* If a channel group has been specified, get it for use when we create peer channels */
outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP);
cur = args.peers;
do {
/* Remember where to start next time */
rest = strchr(cur, '&');
if (rest) {
*rest = 0;
rest++;
}
/* Get a technology/[device:]number pair */
tech = cur;
number = strchr(tech, '/');
if (!number) {
ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
goto out;
}
*number = '\0';
number++;
if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
goto out;
}
if (opts.flags) {
ast_copy_flags(tmp, &opts,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
ast_set2_flag(tmp, args.url, DIAL_NOFORWARDHTML);
}
ast_copy_string(numsubst, number, sizeof(numsubst));
/* If we're dialing by extension, look at the extension to know what to dial */
if ((newnum = strstr(numsubst, "BYEXTENSION"))) {
/* strlen("BYEXTENSION") == 11 */
ast_copy_string(restofit, newnum + 11, sizeof(restofit));
snprintf(newnum, sizeof(numsubst) - (newnum - numsubst), "%s%s", chan->exten,restofit);
if (option_debug)
ast_log(LOG_DEBUG, "Dialing by extension %s\n", numsubst);
}
/* Request the peer */
tmp->chan = ast_request(tech, chan->nativeformats, numsubst, &cause);
if (!tmp->chan) {
/* If we can't, just go on to the next call */
ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n", tech, cause, ast_cause2str(cause));
HANDLE_CAUSE(cause, chan);
cur = rest;
if (!cur)
chan->hangupcause = cause;
continue;
}
pbx_builtin_setvar_helper(tmp->chan, "DIALEDPEERNUMBER", numsubst);
if (!ast_strlen_zero(tmp->chan->call_forward)) {
char tmpchan[256];
char *stuff;
char *tech;
ast_copy_string(tmpchan, tmp->chan->call_forward, sizeof(tmpchan));
if ((stuff = strchr(tmpchan, '/'))) {
*stuff = '\0';
stuff++;
tech = tmpchan;
} else {
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", tmp->chan->call_forward, tmp->chan->context);
stuff = tmpchan;
tech = "Local";
}
tmp->forwards++;
if (tmp->forwards < AST_MAX_FORWARDS) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", chan->name, tech, stuff, tmp->chan->name);
ast_hangup(tmp->chan);
/* Setup parameters */
tmp->chan = ast_request(tech, chan->nativeformats, stuff, &cause);
if (!tmp->chan)
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", tmp->chan->name);
ast_hangup(tmp->chan);
tmp->chan = NULL;
cause = AST_CAUSE_CONGESTION;
}
if (!tmp->chan) {
HANDLE_CAUSE(cause, chan);
cur = rest;
continue;
}
}
/* Setup outgoing SDP to match incoming one */
ast_rtp_make_compatible(tmp->chan, chan);
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tmp->chan);
tmp->chan->appl = "AppDial";
tmp->chan->data = "(Outgoing Line)";
tmp->chan->whentohangup = 0;
if (tmp->chan->cid.cid_num)
free(tmp->chan->cid.cid_num);
tmp->chan->cid.cid_num = NULL;
if (tmp->chan->cid.cid_name)
free(tmp->chan->cid.cid_name);
tmp->chan->cid.cid_name = NULL;
if (tmp->chan->cid.cid_ani)
free(tmp->chan->cid.cid_ani);
tmp->chan->cid.cid_ani = NULL;
if (chan->cid.cid_num)
tmp->chan->cid.cid_num = strdup(chan->cid.cid_num);
if (chan->cid.cid_name)
tmp->chan->cid.cid_name = strdup(chan->cid.cid_name);
if (chan->cid.cid_ani)
tmp->chan->cid.cid_ani = strdup(chan->cid.cid_ani);
/* Copy language from incoming to outgoing */
ast_string_field_set(tmp->chan, language, chan->language);
ast_string_field_set(tmp->chan, accountcode, chan->accountcode);
tmp->chan->cdrflags = chan->cdrflags;
if (ast_strlen_zero(tmp->chan->musicclass))
ast_string_field_set(tmp->chan, musicclass, chan->musicclass);
if (chan->cid.cid_rdnis)
tmp->chan->cid.cid_rdnis = strdup(chan->cid.cid_rdnis);
/* Pass callingpres setting */
tmp->chan->cid.cid_pres = chan->cid.cid_pres;
/* Pass type of number */
tmp->chan->cid.cid_ton = chan->cid.cid_ton;
/* Pass type of tns */
tmp->chan->cid.cid_tns = chan->cid.cid_tns;
/* Presense of ADSI CPE on outgoing channel follows ours */
tmp->chan->adsicpe = chan->adsicpe;
/* Pass the transfer capability */
tmp->chan->transfercapability = chan->transfercapability;
/* If we have an outbound group, set this peer channel to it */
if (outbound_group)
ast_app_group_set_channel(tmp->chan, outbound_group);
/* Place the call, but don't wait on the answer */
res = ast_call(tmp->chan, numsubst, 0);
/* Save the info in cdr's that we called them */
if (chan->cdr)
ast_cdr_setdestchan(chan->cdr, tmp->chan->name);
/* check the results of ast_call */
if (res) {
/* Again, keep going even if there's an error */
if (option_debug)
ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
else if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
ast_hangup(tmp->chan);
tmp->chan = NULL;
cur = rest;
continue;
} else {
senddialevent(chan, tmp->chan);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Called %s\n", numsubst);
if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID))
ast_set_callerid(tmp->chan, ast_strlen_zero(chan->macroexten) ? chan->exten : chan->macroexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
}
/* Put them in the list of outgoing thingies... We're ready now.
XXX If we're forcibly removed, these outgoing calls won't get
hung up XXX */
ast_set_flag(tmp, DIAL_STILLGOING);
tmp->next = outgoing;
outgoing = tmp;
/* If this line is up, don't try anybody else */
if (outgoing->chan->_state == AST_STATE_UP)
break;
cur = rest;
} while (cur);
if (!ast_strlen_zero(args.timeout)) {
to = atoi(args.timeout);
if (to > 0)
to *= 1000;
else
ast_log(LOG_WARNING, "Invalid timeout specified: '%s'\n", args.timeout);
} else
to = -1;
if (outgoing) {
/* Our status will at least be NOANSWER */
strcpy(status, "NOANSWER");
if (ast_test_flag(outgoing, OPT_MUSICBACK)) {
moh=1;
ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK]);
} else if (ast_test_flag(outgoing, OPT_RINGBACK)) {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
}
} else
strcpy(status, "CHANUNAVAIL");
time(&start_time);
peer = wait_for_answer(chan, outgoing, &to, peerflags, &sentringing, status, sizeof(status), numbusy, numnochan, numcongestion, ast_test_flag(&opts, OPT_PRIORITY_JUMP), &result);
if (!peer) {
if (result) {
res = result;
} else if (to)
/* Musta gotten hung up */
res = -1;
else
/* Nobody answered, next please? */
res = 0;
goto out;
}
if (peer) {
time(&answer_time);
#ifdef OSP_SUPPORT
/* Once call is answered, ditch the OSP Handle */
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", "");
#endif
strcpy(status, "ANSWER");
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
hanguptree(outgoing, peer);
outgoing = NULL;
/* If appropriate, log that we have a destination channel */
if (chan->cdr)
ast_cdr_setdestchan(chan->cdr, peer->name);
if (peer->name)
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
number = (char *)pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
if (!number)
number = numsubst;
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
if (option_debug)
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", args.url);
ast_channel_sendurl( peer, args.url );
}
if (ast_test_flag(&opts, OPT_PRIVACY) || ast_test_flag(&opts, OPT_SCREENING)) {
int res2;
int loopcount = 0;
if( privdb_val == AST_PRIVACY_UNKNOWN ) {
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there-- this should be done before the
call is actually dialed */
/* all ring indications and moh for the caller has been halted as soon as the
target extension was picked up. We are going to have to kill some
time and make the caller believe the peer hasn't picked up yet */
if (ast_test_flag(&opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
ast_indicate(chan, -1);
ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK]);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
}
/* Start autoservice on the other chan ?? */
res2 = ast_autoservice_start(chan);
/* Now Stream the File */
if (!res2) {
do {
if (!res2)
res2 = ast_play_and_wait(peer,"priv-callpending");
if( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
res2 = 0;
/* priv-callpending script:
"I have a caller waiting, who introduces themselves as:"
*/
if (!res2)
res2 = ast_play_and_wait(peer,privintro);
if( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
res2 = 0;
/* now get input from the called party, as to their choice */
if( !res2 ) {
if( ast_test_flag(&opts, OPT_PRIVACY) )
res2 = ast_play_and_wait(peer,"priv-callee-options");
if( ast_test_flag(&opts, OPT_SCREENING) )
res2 = ast_play_and_wait(peer,"screen-callee-options");
}
/*! \page DialPrivacy Dial Privacy scripts
\par priv-callee-options script:
"Dial 1 if you wish this caller to reach you directly in the future,
and immediately connect to their incoming call
Dial 2 if you wish to send this caller to voicemail now and
forevermore.
Dial 3 to send this callerr to the torture menus, now and forevermore.
Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
Dial 5 to allow this caller to come straight thru to you in the future,
but right now, just this once, send them to voicemail."
\par screen-callee-options script:
"Dial 1 if you wish to immediately connect to the incoming call
Dial 2 if you wish to send this caller to voicemail.
Dial 3 to send this callerr to the torture menus.
Dial 4 to send this caller to a simple "go away" menu.
*/
if(!res2 || res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2 > '5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) {
/* invalid option */
res2 = ast_play_and_wait(peer, "vm-sorry");
}
loopcount++; /* give the callee a couple chances to make a choice */
} while( (!res2 || res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2 > '5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4')) && loopcount < 2 );
}
switch(res2) {
case '1':
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_ALLOW);
}
break;
case '2':
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to DENY\n",
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_DENY);
}
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
res=0;
goto out;
case '3':
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to TORTURE\n",
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_TORTURE);
}
ast_copy_string(status, "TORTURE", sizeof(status));
res = 0;
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
ast_hangup(peer); /* hang up on the caller -- he didn't want to talk anyway! */
goto out; /* Is this right? */
case '4':
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to KILL\n",
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_KILL);
}
ast_copy_string(status, "DONTCALL", sizeof(status));
res = 0;
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
ast_hangup(peer); /* hang up on the caller -- he didn't want to talk anyway! */
goto out; /* Is this right? */
case '5':
if( ast_test_flag(&opts, OPT_PRIVACY) ) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "--Set privacy database entry %s/%s to ALLOW\n",
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_ALLOW);
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
ast_hangup(peer); /* hang up on the caller -- he didn't want to talk anyway! */
res=0;
goto out;
} /* if not privacy, then 5 is the same as "default" case */
default:
/* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
/* well, there seems basically two choices. Just patch the caller thru immediately,
or,... put 'em thru to voicemail. */
/* since the callee may have hung up, let's do the voicemail thing, no database decision */
ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
res=0;
goto out;
}
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
ast_indicate(chan, -1);
sentringing=0;
}
res2 = ast_autoservice_stop(chan);
/* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
just clog things up, and it's not useful information, not being tied to a CID */
if( strncmp(privcid,"NOCALLERID",10) == 0 || ast_test_flag(&opts, OPT_SCREEN_NOINTRO) ) {
ast_filedelete(privintro, NULL);
if( ast_fileexists(privintro, NULL, NULL ) > 0 )
ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", privintro);
else if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", privintro);
}
}
}
if (ast_test_flag(&opts, OPT_ANNOUNCE) && !ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
/* Start autoservice on the other chan */
res = ast_autoservice_start(chan);
/* Now Stream the File */
if (!res)
res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
if (!res) {
digit = ast_waitstream(peer, AST_DIGIT_ANY);
}
/* Ok, done. stop autoservice */
res = ast_autoservice_stop(chan);
if (digit > 0 && !res)
res = ast_senddigit(chan, digit);
else
res = digit;
} else
res = 0;
if (chan && peer && ast_test_flag(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
char *ch;
for (ch = opt_args[OPT_ARG_GOTO]; *ch; ch++) {
if (*ch == '^')
*ch = '|';
}
ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
ast_parseable_goto(peer, opt_args[OPT_ARG_GOTO]);
peer->priority++;
ast_pbx_start(peer);
hanguptree(outgoing, NULL);
LOCAL_USER_REMOVE(u);
return 0;
}
if (ast_test_flag(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
char *ch;
res = ast_autoservice_start(chan);
if (res) {
ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
res = -1;
}
app = pbx_findapp("Macro");
if (app && !res) {
for (ch = opt_args[OPT_ARG_CALLEE_MACRO]; *ch; ch++) {
if (*ch == '^')
*ch = '|';
}
res = pbx_exec(peer, app, opt_args[OPT_ARG_CALLEE_MACRO], 1);
ast_log(LOG_DEBUG, "Macro exited with status %d\n", res);
res = 0;
} else {
ast_log(LOG_ERROR, "Could not find application Macro\n");
res = -1;
}
if (ast_autoservice_stop(chan) < 0) {
ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
res = -1;
}
if (!res) {
if ((macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
if (!strcasecmp(macro_result, "BUSY")) {
ast_copy_string(status, macro_result, sizeof(status));
if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
if (!ast_goto_if_exists(chan, NULL, NULL, chan->priority + 101)) {
ast_set_flag(peerflags, OPT_GO_ON);
}
} else
ast_set_flag(peerflags, OPT_GO_ON);
res = -1;
}
else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
ast_copy_string(status, macro_result, sizeof(status));
ast_set_flag(peerflags, OPT_GO_ON);
res = -1;
}
else if (!strcasecmp(macro_result, "CONTINUE")) {
/* hangup peer and keep chan alive assuming the macro has changed
the context / exten / priority or perhaps
the next priority in the current exten is desired.
*/
ast_set_flag(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(macro_result, "ABORT")) {
/* Hangup both ends unless the caller has the g flag */
res = -1;
} else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
res = -1;
/* perform a transfer to a new extension */
if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
/* no brainer mode... substitute ^ with | and feed it to builtin goto */
for (res = 0; res < strlen(macro_transfer_dest); res++)
if (macro_transfer_dest[res] == '^')
macro_transfer_dest[res] = '|';
if (!ast_parseable_goto(chan, macro_transfer_dest))
ast_set_flag(peerflags, OPT_GO_ON);
}
}
}
}
}
if (!res) {
if (calldurationlimit > 0) {
time_t now;
time(&now);
chan->whentohangup = now + calldurationlimit;
}
if (!ast_strlen_zero(dtmfcalled)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the called party.\n", dtmfcalled);
res = ast_dtmf_stream(peer,chan,dtmfcalled,250);
}
if (!ast_strlen_zero(dtmfcalling)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
res = ast_dtmf_stream(chan,peer,dtmfcalling,250);
}
}
if (!res) {
memset(&config,0,sizeof(struct ast_bridge_config));
if (play_to_caller)
ast_set_flag(&(config.features_caller), AST_FEATURE_PLAY_WARNING);
if (play_to_callee)
ast_set_flag(&(config.features_callee), AST_FEATURE_PLAY_WARNING);
if (ast_test_flag(peerflags, OPT_CALLEE_TRANSFER))
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
if (ast_test_flag(peerflags, OPT_CALLER_TRANSFER))
ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
if (ast_test_flag(peerflags, OPT_CALLEE_HANGUP))
ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
if (ast_test_flag(peerflags, OPT_CALLER_HANGUP))
ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
if (ast_test_flag(peerflags, OPT_CALLEE_MONITOR))
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
if (ast_test_flag(peerflags, OPT_CALLER_MONITOR))
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
config.timelimit = timelimit;
config.play_warning = play_warning;
config.warning_freq = warning_freq;
config.warning_sound = warning_sound;
config.end_sound = end_sound;
config.start_sound = start_sound;
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
/* Be sure no generators are left on it */
ast_deactivate_generator(chan);
/* Make sure channels are compatible */
res = ast_channel_make_compatible(chan, peer);
if (res < 0) {
ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
ast_hangup(peer);
LOCAL_USER_REMOVE(u);
return -1;
}
res = ast_bridge_call(chan,peer,&config);
time(&end_time);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
} else {
time(&end_time);
res = -1;
}
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
if (res != AST_PBX_NO_HANGUP_PEER) {
if (!chan->_softhangup)
chan->hangupcause = peer->hangupcause;
ast_hangup(peer);
}
}
out:
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
hanguptree(outgoing, NULL);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
if (option_debug)
ast_log(LOG_DEBUG, "Exiting with DIALSTATUS=%s.\n", status);
if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE))
res = 0;
LOCAL_USER_REMOVE(u);
return res;
}
static int dial_exec(struct ast_channel *chan, void *data)
{
struct ast_flags peerflags;
memset(&peerflags, 0, sizeof(peerflags));
return dial_exec_full(chan, data, &peerflags);
}
static int retrydial_exec(struct ast_channel *chan, void *data)
{
char *announce = NULL, *dialdata = NULL;
const char *context = NULL;
int sleep = 0, loops = 0, res = 0;
struct localuser *u;
struct ast_flags peerflags;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
return -1;
}
LOCAL_USER_ADD(u);
if (!(announce = ast_strdupa(data))) {
LOCAL_USER_REMOVE(u);
return -1;
}
memset(&peerflags, 0, sizeof(peerflags));
if ((dialdata = strchr(announce, '|'))) {
*dialdata = '\0';
dialdata++;
if ((sleep = atoi(dialdata))) {
sleep *= 1000;
} else {
ast_log(LOG_ERROR, "%s requires the numerical argument <sleep>\n",rapp);
LOCAL_USER_REMOVE(u);
return -1;
}
if ((dialdata = strchr(dialdata, '|'))) {
*dialdata = '\0';
dialdata++;
if (!(loops = atoi(dialdata))) {
ast_log(LOG_ERROR, "%s requires the numerical argument <loops>\n",rapp);
LOCAL_USER_REMOVE(u);
return -1;
}
}
}
if ((dialdata = strchr(dialdata, '|'))) {
*dialdata = '\0';
dialdata++;
} else {
ast_log(LOG_ERROR, "%s requires more arguments\n",rapp);
LOCAL_USER_REMOVE(u);
return -1;
}
if (sleep < 1000)
sleep = 10000;
if (!loops)
loops = -1;
context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
while (loops) {
chan->data = "Retrying";
if (ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_stop(chan);
if ((res = dial_exec_full(chan, dialdata, &peerflags)) == 0) {
if (ast_test_flag(&peerflags, OPT_DTMF_EXIT)) {
if (!(res = ast_streamfile(chan, announce, chan->language)))
res = ast_waitstream(chan, AST_DIGIT_ANY);
if (!res && sleep) {
if (!ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_start(chan, NULL);
res = ast_waitfordigit(chan, sleep);
}
} else {
if (!(res = ast_streamfile(chan, announce, chan->language)))
res = ast_waitstream(chan, "");
if (sleep) {
if (!ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_start(chan, NULL);
if (!res)
res = ast_waitfordigit(chan, sleep);
}
}
}
if (res < 0)
break;
else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
if (onedigit_goto(chan, context, (char) res, 1)) {
res = 0;
break;
}
}
loops--;
}
if (ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_stop(chan);
LOCAL_USER_REMOVE(u);
return loops ? res : 0;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
res |= ast_unregister_application(rapp);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app, dial_exec, synopsis, descrip);
res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}
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