aboutsummaryrefslogtreecommitdiffstats
path: root/1.4.23-rc4/apps/app_dial.c
diff options
context:
space:
mode:
Diffstat (limited to '1.4.23-rc4/apps/app_dial.c')
-rw-r--r--1.4.23-rc4/apps/app_dial.c2015
1 files changed, 0 insertions, 2015 deletions
diff --git a/1.4.23-rc4/apps/app_dial.c b/1.4.23-rc4/apps/app_dial.c
deleted file mode 100644
index 303b36121..000000000
--- a/1.4.23-rc4/apps/app_dial.c
+++ /dev/null
@@ -1,2015 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, 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
- */
-
-/*** MODULEINFO
- <depend>chan_local</depend>
- ***/
-
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#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 <sys/stat.h>
-#include <netinet/in.h>
-
-#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/cdr.h"
-#include "asterisk/manager.h"
-#include "asterisk/privacy.h"
-#include "asterisk/stringfields.h"
-#include "asterisk/global_datastores.h"
-
-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 application 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 | INVALIDARGS\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"
-" If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
-"application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
-"however, the variable will be unset after use.\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. You cannot use any additional\n"
-" action post answer options in conjunction with this option.\n"
-" h - Allow the called party to hang up by sending the '*' DTMF digit, or\n"
-" whatever sequence was defined in the featuremap section for\n"
-" 'disconnect' in features.conf\n"
-" H - Allow the calling party to hang up by hitting the '*' DTMF digit, or\n"
-" whatever sequence was defined in the featuremap section for\n"
-" 'disconnect' in features.conf\n"
-" i - Asterisk will ignore any forwarding requests it may receive on this\n"
-" dial attempt.\n"
-" j - Jump to priority n+101 if all of the requested channels were busy.\n"
-" k - Allow the called party to enable parking of the call by sending\n"
-" the DTMF sequence defined for call parking in the featuremap section of features.conf.\n"
-" K - Allow the calling party to enable parking of the call by sending\n"
-" the DTMF sequence defined for call parking in the featuremap section of features.conf.\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"
-" You cannot use any additional action post answer options in conjunction\n"
-" with this option. Also, pbx services are not run on the peer (called) channel,\n"
-" so you will not be able to set timeouts via the TIMEOUT() function in this macro.\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"
-" O([x]) - \"Operator Services\" mode (Zaptel channel to Zaptel channel\n"
-" only, if specified on non-Zaptel interface, it will be ignored).\n"
-" When the destination answers (presumably an operator services\n"
-" station), the originator no longer has control of their line.\n"
-" They may hang up, but the switch will not release their line\n"
-" until the destination party hangs up (the operator). Specified\n"
-" without an arg, or with 1 as an arg, the originator hanging up\n"
-" will cause the phone to ring back immediately. With a 2 specified,\n"
-" when the \"operator\" flashes the trunk, it will ring their phone\n"
-" back.\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 the blindxfer setting in the featuremap section\n"
-" of features.conf.\n"
-" T - Allow the calling party to transfer the called party by sending the\n"
-" DTMF sequence defined in the blindxfer setting in the featuremap section\n"
-" of features.conf.\n"
-" w - Allow the called party to enable recording of the call by sending\n"
-" the DTMF sequence defined in the automon setting in the featuremap section\n"
-" of features.conf.\n"
-" W - Allow the calling party to enable recording of the call by sending\n"
-" the DTMF sequence defined in the automon setting in the featuremap section\n"
-" of 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 retrying the call. After 'retries' 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),
- OPT_OPERMODE = (1 << 24),
- OPT_CALLEE_PARK = (1 << 25),
- OPT_CALLER_PARK = (1 << 26),
- OPT_IGNORE_FORWARDING = (1 << 27),
-} 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,
- OPT_ARG_OPERMODE,
- /* 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('i', OPT_IGNORE_FORWARDING),
- AST_APP_OPTION('j', OPT_PRIORITY_JUMP),
- AST_APP_OPTION('k', OPT_CALLEE_PARK),
- AST_APP_OPTION('K', OPT_CALLER_PARK),
- 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_ARG('O', OPT_OPERMODE,OPT_ARG_OPERMODE),
- 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),
-});
-
-#define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag(flags, OPT_CALLEE_HANGUP | \
- OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
- !chan->audiohooks && !peer->audiohooks)
-
-/* 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;
- struct dial_localuser *next;
-};
-
-
-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_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_NO_ROUTE_DESTINATION: \
- case AST_CAUSE_UNREGISTERED: \
- if (chan->cdr) \
- ast_cdr_failed(chan->cdr); \
- numnochan++; \
- break; \
- case AST_CAUSE_NORMAL_CLEARING: \
- 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 const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
-{
- const char *context = S_OR(chan->macrocontext, chan->context);
- const char *exten = S_OR(chan->macroexten, chan->exten);
-
- return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
-}
-
-static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
-{
- /* XXX do we need also CallerIDnum ? */
- 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, S_OR(src->cid.cid_num, "<unknown>"),
- S_OR(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)
-{
- int numbusy = busystart;
- int numcongestion = congestionstart;
- int numnochan = nochanstart;
- int prestart = busystart + congestionstart + nochanstart;
- int orig = *to;
- struct ast_channel *peer = NULL;
- /* single is set if only one destination is enabled */
- int 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) {
- struct dial_localuser *o;
- int pos = 0; /* how many channels do we handle */
- int numlines = prestart;
- struct ast_channel *winner;
- struct ast_channel *watchers[AST_MAX_WATCHERS];
-
- watchers[pos++] = in;
- for (o = outgoing; o; o = o->next) {
- /* Keep track of important channels */
- if (ast_test_flag(o, DIAL_STILLGOING) && o->chan)
- watchers[pos++] = o->chan;
- numlines++;
- }
- if (pos == 1) { /* only the input channel is available */
- 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);
- for (o = outgoing; o; o = o->next) {
- struct ast_frame *f;
- struct ast_channel *c = o->chan;
-
- if (c == NULL)
- continue;
- if (ast_test_flag(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
- if (!peer) {
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", c->name, in->name);
- peer = c;
- ast_copy_flags(peerflags, o,
- OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
- OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
- OPT_CALLEE_PARK | OPT_CALLER_PARK |
- DIAL_NOFORWARDHTML);
- ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
- ast_copy_string(c->exten, "", sizeof(c->exten));
- }
- continue;
- }
- if (c != winner)
- continue;
- if (!ast_strlen_zero(c->call_forward)) {
- char tmpchan[256];
- char *stuff;
- char *tech;
- int cause;
-
- ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
- if ((stuff = strchr(tmpchan, '/'))) {
- *stuff++ = '\0';
- tech = tmpchan;
- } else {
- const char *forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
- snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
- stuff = tmpchan;
- tech = "Local";
- }
- /* Before processing channel, go ahead and check for forwarding */
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
- /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
- if (ast_test_flag(peerflags, OPT_IGNORE_FORWARDING)) {
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
- c = o->chan = NULL;
- cause = AST_CAUSE_BUSY;
- } else {
- /* Setup parameters */
- if ((c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause))) {
- if (single)
- ast_channel_make_compatible(o->chan, in);
- ast_channel_inherit_variables(in, o->chan);
- ast_channel_datastore_inherit(in, o->chan);
- } else
- ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
- }
- if (!c) {
- ast_clear_flag(o, DIAL_STILLGOING);
- HANDLE_CAUSE(cause, in);
- } else {
- ast_rtp_make_compatible(c, in, single);
- if (c->cid.cid_num)
- free(c->cid.cid_num);
- c->cid.cid_num = NULL;
- if (c->cid.cid_name)
- free(c->cid.cid_name);
- c->cid.cid_name = NULL;
-
- if (ast_test_flag(o, OPT_FORCECLID)) {
- c->cid.cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
- ast_string_field_set(c, accountcode, winner->accountcode);
- c->cdrflags = winner->cdrflags;
- } else {
- c->cid.cid_num = ast_strdup(in->cid.cid_num);
- c->cid.cid_name = ast_strdup(in->cid.cid_name);
- ast_string_field_set(c, accountcode, in->accountcode);
- c->cdrflags = in->cdrflags;
- }
-
- if (in->cid.cid_ani) {
- if (c->cid.cid_ani)
- free(c->cid.cid_ani);
- c->cid.cid_ani = ast_strdup(in->cid.cid_ani);
- }
- if (c->cid.cid_rdnis)
- free(c->cid.cid_rdnis);
- c->cid.cid_rdnis = ast_strdup(S_OR(in->macroexten, in->exten));
- if (ast_call(c, 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(c);
- c = o->chan = NULL;
- numnochan++;
- } else {
- senddialevent(in, c);
- /* After calling, set callerid to extension */
- if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID)) {
- char cidname[AST_MAX_EXTENSION] = "";
- ast_set_callerid(c, S_OR(in->macroexten, in->exten), 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) {
- in->hangupcause = c->hangupcause;
- ast_hangup(c);
- c = o->chan = NULL;
- ast_clear_flag(o, DIAL_STILLGOING);
- HANDLE_CAUSE(in->hangupcause, in);
- continue;
- }
- 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", c->name, in->name);
- peer = c;
- if (peer->cdr) {
- peer->cdr->answer = ast_tvnow();
- peer->cdr->disposition = AST_CDR_ANSWERED;
- }
- ast_copy_flags(peerflags, o,
- OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
- OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
- OPT_CALLEE_PARK | OPT_CALLER_PARK |
- DIAL_NOFORWARDHTML);
- ast_copy_string(c->dialcontext, "", sizeof(c->dialcontext));
- ast_copy_string(c->exten, "", sizeof(c->exten));
- /* Setup RTP early bridge if appropriate */
- if (CAN_EARLY_BRIDGE(peerflags, in, peer))
- ast_rtp_early_bridge(in, peer);
- }
- /* If call has been answered, then the eventual hangup is likely to be normal hangup */
- in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
- c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
- break;
- case AST_CONTROL_BUSY:
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "%s is busy\n", c->name);
- in->hangupcause = c->hangupcause;
- ast_hangup(c);
- c = 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", c->name);
- in->hangupcause = c->hangupcause;
- ast_hangup(c);
- c = 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", c->name);
- /* Setup early media if appropriate */
- if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
- ast_rtp_early_bridge(in, c);
- 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", c->name, in->name);
- /* Setup early media if appropriate */
- if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
- ast_rtp_early_bridge(in, c);
- 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", c->name, in->name);
- ast_indicate(in, AST_CONTROL_VIDUPDATE);
- break;
- case AST_CONTROL_SRCUPDATE:
- if (option_verbose > 2)
- ast_verbose (VERBOSE_PREFIX_3 "%s requested a source update, passing it to %s\n", c->name, in->name);
- ast_indicate(in, AST_CONTROL_SRCUPDATE);
- break;
- case AST_CONTROL_PROCEEDING:
- if (option_verbose > 2)
- ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name);
- if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
- ast_rtp_early_bridge(in, c);
- 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", c->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", c->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", c->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) {
- /* XXX are we sure the logic is correct ? or we should just switch on f->frametype ? */
- if (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 (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 (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 (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);
- } /* end for */
- if (winner == in) {
- struct ast_frame *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;
- ast_cdr_noanswer(in->cdr);
- 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)) {
- const char *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;
- ast_cdr_noanswer(in->cdr);
- *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;
- ast_cdr_noanswer(in->cdr);
- 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_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
- if (ast_write(outgoing->chan, f))
- ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
- }
- if (single && (f->frametype == AST_FRAME_CONTROL) &&
- ((f->subclass == AST_CONTROL_HOLD) ||
- (f->subclass == AST_CONTROL_UNHOLD) ||
- (f->subclass == AST_CONTROL_VIDUPDATE) ||
- (f->subclass == AST_CONTROL_SRCUPDATE))) {
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
- ast_indicate_data(outgoing->chan, f->subclass, f->data, f->datalen);
- }
- ast_frfree(f);
- }
- if (!*to && (option_verbose > 2))
- ast_verbose(VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
- if (!*to || ast_check_hangup(in)) {
- ast_cdr_noanswer(in->cdr);
- }
-
- }
-
- return peer;
-}
-
-static void replace_macro_delimiter(char *s)
-{
- for (; *s; s++)
- if (*s == '^')
- *s = '|';
-}
-
-
-/* returns true if there is a valid privacy reply */
-static int valid_priv_reply(struct ast_flags *opts, int res)
-{
- if (res < '1')
- return 0;
- if (ast_test_flag(opts, OPT_PRIVACY) && res <= '5')
- return 1;
- if (ast_test_flag(opts, OPT_SCREENING) && res <= '4')
- return 1;
- return 0;
-}
-
-static void set_dial_features(struct ast_flags *opts, struct ast_dial_features *features)
-{
- struct ast_flags perm_opts = {.flags = 0};
-
- ast_copy_flags(&perm_opts, opts,
- OPT_CALLER_TRANSFER | OPT_CALLER_PARK | OPT_CALLER_MONITOR | OPT_CALLER_HANGUP |
- OPT_CALLEE_TRANSFER | OPT_CALLEE_PARK | OPT_CALLEE_MONITOR | OPT_CALLEE_HANGUP);
-
- memset(features->options, 0, sizeof(features->options));
-
- ast_app_options2str(dial_exec_options, &perm_opts, features->options, sizeof(features->options));
- if (ast_test_flag(&perm_opts, OPT_CALLEE_TRANSFER))
- ast_set_flag(&(features->features_callee), AST_FEATURE_REDIRECT);
- if (ast_test_flag(&perm_opts, OPT_CALLER_TRANSFER))
- ast_set_flag(&(features->features_caller), AST_FEATURE_REDIRECT);
- if (ast_test_flag(&perm_opts, OPT_CALLEE_HANGUP))
- ast_set_flag(&(features->features_callee), AST_FEATURE_DISCONNECT);
- if (ast_test_flag(&perm_opts, OPT_CALLER_HANGUP))
- ast_set_flag(&(features->features_caller), AST_FEATURE_DISCONNECT);
- if (ast_test_flag(&perm_opts, OPT_CALLEE_MONITOR))
- ast_set_flag(&(features->features_callee), AST_FEATURE_AUTOMON);
- if (ast_test_flag(&perm_opts, OPT_CALLER_MONITOR))
- ast_set_flag(&(features->features_caller), AST_FEATURE_AUTOMON);
- if (ast_test_flag(&perm_opts, OPT_CALLEE_PARK))
- ast_set_flag(&(features->features_callee), AST_FEATURE_PARKCALL);
- if (ast_test_flag(&perm_opts, OPT_CALLER_PARK))
- ast_set_flag(&(features->features_caller), AST_FEATURE_PARKCALL);
-}
-
-static void end_bridge_callback (void *data)
-{
- char buf[80];
- time_t end;
- struct ast_channel *chan = data;
-
- if (!chan->cdr) {
- return;
- }
-
- time(&end);
-
- ast_channel_lock(chan);
- if (chan->cdr->answer.tv_sec) {
- snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
- pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
- }
-
- if (chan->cdr->start.tv_sec) {
- snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
- pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
- }
- ast_channel_unlock(chan);
-}
-
-static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
- bconfig->end_bridge_callback_data = originator;
-}
-
-static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags *peerflags, int *continue_exec)
-{
- int res = -1;
- struct ast_module_user *u;
- char *rest, *cur;
- struct dial_localuser *outgoing = NULL;
- struct ast_channel *peer;
- int to;
- int numbusy = 0;
- int numcongestion = 0;
- int numnochan = 0;
- int cause;
- char numsubst[256];
- char cidname[AST_MAX_EXTENSION] = "";
- int privdb_val = 0;
- int calldurationlimit = -1;
- 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;
- char status[256] = "INVALIDARGS";
- int play_to_caller = 0, play_to_callee = 0;
- int sentringing = 0, moh = 0;
- const char *outbound_group = NULL;
- int result = 0;
- time_t start_time;
- char privintro[1024];
- char privcid[256];
- char *parse;
- int opermode = 0;
- 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];
- struct ast_datastore *datastore = NULL;
- struct ast_datastore *ds_caller_features = NULL;
- struct ast_datastore *ds_callee_features = NULL;
- struct ast_dial_features *caller_features;
- int fulldial = 0, num_dialed = 0;
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
- return -1;
- }
-
- /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
- pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
- pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
-
- u = ast_module_user_add(chan);
-
- parse = ast_strdupa(data);
-
- AST_STANDARD_APP_ARGS(args, parse);
-
- if (!ast_strlen_zero(args.options) &&
- ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options)) {
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
- goto done;
- }
-
- if (ast_strlen_zero(args.peers)) {
- ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
- goto done;
- }
-
- if (ast_test_flag(&opts, OPT_OPERMODE)) {
- if (ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]))
- opermode = 1;
- else opermode = atoi(opt_args[OPT_ARG_OPERMODE]);
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Setting operator services mode to %d.\n", opermode);
- }
-
- 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 (!calldurationlimit) {
- ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
- goto done;
- }
- 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])) {
- dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
- dtmfcalled = strsep(&dtmfcalling, ":");
- }
-
- if (ast_test_flag(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
- char *limit_str, *warning_str, *warnfreq_str;
- const char *var;
-
- warnfreq_str = opt_args[OPT_ARG_DURATION_LIMIT];
- limit_str = strsep(&warnfreq_str, ":");
- warning_str = strsep(&warnfreq_str, ":");
-
- timelimit = atol(limit_str);
- if (warning_str)
- play_warning = atol(warning_str);
- if (warnfreq_str)
- warning_freq = atol(warnfreq_str);
-
- if (!timelimit) {
- ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
- goto done;
- } else if (play_warning > timelimit) {
- /* If the first warning is requested _after_ the entire call would end,
- and no warning frequency is requested, then turn off the warning. If
- a warning frequency is requested, reduce the 'first warning' time by
- that frequency until it falls within the call's total time limit.
- */
-
- if (!warning_freq) {
- play_warning = 0;
- } else {
- /* XXX fix this!! */
- while (play_warning > timelimit)
- play_warning -= warning_freq;
- if (play_warning < 1)
- play_warning = warning_freq = 0;
- }
- }
-
- 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 = S_OR(var, "timeleft");
-
- var = pbx_builtin_getvar_helper(chan,"LIMIT_TIMEOUT_FILE");
- end_sound = S_OR(var, NULL); /* XXX not much of a point in doing this! */
-
- var = pbx_builtin_getvar_helper(chan,"LIMIT_CONNECT_FILE");
- start_sound = S_OR(var, NULL); /* XXX not much of a point in doing this! */
-
- /* undo effect of S(x) in case they are both used */
- calldurationlimit = -1;
- /* more efficient to do it like S(x) does since no advanced opts */
- if (!play_warning && !start_sound && !end_sound && timelimit) {
- calldurationlimit = timelimit / 1000;
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Setting call duration limit to %d seconds.\n", calldurationlimit);
- 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_4 "timelimit = %ld\n", timelimit);
- ast_verbose(VERBOSE_PREFIX_4 "play_warning = %ld\n", play_warning);
- ast_verbose(VERBOSE_PREFIX_4 "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
- ast_verbose(VERBOSE_PREFIX_4 "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
- ast_verbose(VERBOSE_PREFIX_4 "warning_freq = %ld\n", warning_freq);
- ast_verbose(VERBOSE_PREFIX_4 "start_sound = %s\n", start_sound);
- ast_verbose(VERBOSE_PREFIX_4 "warning_sound = %s\n", warning_sound);
- ast_verbose(VERBOSE_PREFIX_4 "end_sound = %s\n", end_sound);
- }
- }
-
- 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];
- char *l = chan->cid.cid_num; /* XXX watch out, we are overwriting it */
- 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 ) {
- ast_copy_string(status, "NOANSWER", sizeof(status));
- 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_copy_string(status, "DONTCALL", sizeof(status));
- if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
- 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_copy_string(status, "TORTURE", sizeof(status));
- if (ast_opt_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
- 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 actually exists */
- snprintf(privintro, sizeof(privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
- if (mkdir(privintro, 0755) && errno != EEXIST) {
- ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(errno));
- res = -1;
- goto out;
- }
-
- 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_answer(chan);
- res = 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 (res == -1) {
- /* Delete the file regardless since they hung up during recording */
- 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);
- goto out;
- }
- if( !ast_streamfile(chan, "vm-dialout", chan->language) )
- ast_waitstream(chan, "");
- }
- }
- }
-
- if (continue_exec)
- *continue_exec = 0;
-
- /* If a channel group has been specified, get it for use when we create peer channels */
- if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
- outbound_group = ast_strdupa(outbound_group);
- pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
- } else {
- 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 | OPT_IGNORE_FORWARDING);
-
- /* Create datastore for channel dial features for caller */
- if (!(ds_caller_features = ast_channel_datastore_alloc(&dial_features_info, NULL))) {
- ast_log(LOG_WARNING, "Unable to create channel datastore for dial features. Aborting!\n");
- goto out;
- }
-
- if (!(caller_features = ast_calloc(1, sizeof(*caller_features)))) {
- ast_log(LOG_WARNING, "Unable to allocate memory for feature flags. Aborting!\n");
- goto out;
- }
-
- ast_channel_lock(chan);
- caller_features->is_caller = 1;
- set_dial_features(&opts, caller_features);
- ds_caller_features->inheritance = -1;
- ds_caller_features->data = caller_features;
- ast_channel_datastore_add(chan, ds_caller_features);
- ast_channel_unlock(chan);
-
- /* loop through the list of dial destinations */
- rest = args.peers;
- while ((cur = strsep(&rest, "&")) ) {
- struct dial_localuser *tmp;
- /* Get a technology/[device:]number pair */
- char *number = cur;
- char *interface = ast_strdupa(number);
- char *tech = strsep(&number, "/");
- /* find if we already dialed this interface */
- struct ast_dialed_interface *di;
- struct ast_dial_features *callee_features;
- AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
- num_dialed++;
- if (!number) {
- ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
- goto out;
- }
- 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_CALLEE_PARK | OPT_CALLER_PARK |
- OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
- ast_set2_flag(tmp, args.url, DIAL_NOFORWARDHTML);
- }
- ast_copy_string(numsubst, number, sizeof(numsubst));
- /* Request the peer */
-
- ast_channel_lock(chan);
- datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
- ast_channel_unlock(chan);
-
- if (datastore)
- dialed_interfaces = datastore->data;
- else {
- if (!(datastore = ast_channel_datastore_alloc(&dialed_interface_info, NULL))) {
- ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
- free(tmp);
- goto out;
- }
-
- datastore->inheritance = DATASTORE_INHERIT_FOREVER;
-
- if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
- free(tmp);
- goto out;
- }
-
- datastore->data = dialed_interfaces;
- AST_LIST_HEAD_INIT(dialed_interfaces);
-
- ast_channel_lock(chan);
- ast_channel_datastore_add(chan, datastore);
- ast_channel_unlock(chan);
- }
-
- AST_LIST_LOCK(dialed_interfaces);
- AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
- if (!strcasecmp(di->interface, interface)) {
- ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
- di->interface);
- break;
- }
- }
- AST_LIST_UNLOCK(dialed_interfaces);
-
- if (di) {
- fulldial++;
- free(tmp);
- continue;
- }
-
- /* It is always ok to dial a Local interface. We only keep track of
- * which "real" interfaces have been dialed. The Local channel will
- * inherit this list so that if it ends up dialing a real interface,
- * it won't call one that has already been called. */
- if (strcasecmp(tech, "Local")) {
- if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
- AST_LIST_UNLOCK(dialed_interfaces);
- free(tmp);
- goto out;
- }
- strcpy(di->interface, interface);
-
- AST_LIST_LOCK(dialed_interfaces);
- AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
- AST_LIST_UNLOCK(dialed_interfaces);
- }
-
- 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);
- if (!rest) /* we are on the last destination */
- chan->hangupcause = cause;
- free(tmp);
- continue;
- }
-
- pbx_builtin_setvar_helper(tmp->chan, "DIALEDPEERNUMBER", numsubst);
-
- /* Setup outgoing SDP to match incoming one */
- ast_rtp_make_compatible(tmp->chan, chan, !outgoing && !rest);
-
- /* 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 = ast_strdup(chan->cid.cid_num);
-
- if (tmp->chan->cid.cid_name)
- free(tmp->chan->cid.cid_name);
- tmp->chan->cid.cid_name = ast_strdup(chan->cid.cid_name);
-
- if (tmp->chan->cid.cid_ani)
- free(tmp->chan->cid.cid_ani);
- tmp->chan->cid.cid_ani = ast_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);
- /* XXX don't we free previous values ? */
- tmp->chan->cid.cid_rdnis = ast_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);
-
- /* Inherit context and extension */
- if (!ast_strlen_zero(chan->macrocontext))
- ast_copy_string(tmp->chan->dialcontext, chan->macrocontext, sizeof(tmp->chan->dialcontext));
- else
- ast_copy_string(tmp->chan->dialcontext, chan->context, sizeof(tmp->chan->dialcontext));
- if (!ast_strlen_zero(chan->macroexten))
- ast_copy_string(tmp->chan->exten, chan->macroexten, sizeof(tmp->chan->exten));
- else
- ast_copy_string(tmp->chan->exten, chan->exten, sizeof(tmp->chan->exten));
-
- /* Save callee features */
- if (!(ds_callee_features = ast_channel_datastore_alloc(&dial_features_info, NULL))) {
- ast_log(LOG_WARNING, "Unable to create channel datastore for dial features. Aborting!\n");
- ast_free(tmp);
- goto out;
- }
-
- if (!(callee_features = ast_calloc(1, sizeof(*callee_features)))) {
- ast_log(LOG_WARNING, "Unable to allocate memory for feature flags. Aborting!\n");
- ast_free(tmp);
- goto out;
- }
-
- ast_channel_lock(tmp->chan);
- callee_features->is_caller = 0;
- set_dial_features(&opts, callee_features);
- ds_callee_features->inheritance = -1;
- ds_callee_features->data = callee_features;
- ast_channel_datastore_add(tmp->chan, ds_callee_features);
- ast_channel_unlock(tmp->chan);
-
- /* 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);
- if (option_verbose > 2)
- ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
- if (tmp->chan->hangupcause) {
- chan->hangupcause = tmp->chan->hangupcause;
- }
- ast_hangup(tmp->chan);
- tmp->chan = NULL;
- free(tmp);
- 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, S_OR(chan->macroexten, chan->exten), 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;
- }
-
- if (ast_strlen_zero(args.timeout)) {
- to = -1;
- } else {
- to = atoi(args.timeout);
- if (to > 0)
- to *= 1000;
- else
- ast_log(LOG_WARNING, "Invalid timeout specified: '%s'\n", args.timeout);
- }
-
- if (!outgoing) {
- strcpy(status, "CHANUNAVAIL");
- if(fulldial == num_dialed) {
- res = -1;
- goto out;
- }
- } else {
- /* Our status will at least be NOANSWER */
- strcpy(status, "NOANSWER");
- if (ast_test_flag(outgoing, OPT_MUSICBACK)) {
- moh = 1;
- if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
- char *original_moh = ast_strdupa(chan->musicclass);
- ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
- ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
- ast_string_field_set(chan, musicclass, original_moh);
- } else {
- ast_moh_start(chan, NULL, NULL);
- }
- ast_indicate(chan, AST_CONTROL_PROGRESS);
- } else if (ast_test_flag(outgoing, OPT_RINGBACK)) {
- ast_indicate(chan, AST_CONTROL_RINGING);
- sentringing++;
- }
- }
-
- 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);
-
- /* The ast_channel_datastore_remove() function could fail here if the
- * datastore was moved to another channel during a masquerade. If this is
- * the case, don't free the datastore here because later, when the channel
- * to which the datastore was moved hangs up, it will attempt to free this
- * datastore again, causing a crash
- */
- if (!ast_channel_datastore_remove(chan, datastore))
- ast_channel_datastore_free(datastore);
- if (!peer) {
- if (result) {
- res = result;
- } else if (to) { /* Musta gotten hung up */
- res = -1;
- } else { /* Nobody answered, next please? */
- res = 0;
- }
- /* almost done, although the 'else' block is 400 lines */
- } else {
- const char *number;
-
- strcpy(status, "ANSWER");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
- /* 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 = 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)) && privdb_val == AST_PRIVACY_UNKNOWN) {
- int res2;
- int loopcount = 0;
-
- /* 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])) {
- char *original_moh = ast_strdupa(chan->musicclass);
- ast_indicate(chan, -1);
- ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
- ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
- ast_string_field_set(chan, musicclass, original_moh);
- } 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 */
- for (loopcount = 0; loopcount < 3; loopcount++) {
- if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
- break;
- if (!res2) /* on timeout, play the message again */
- res2 = ast_play_and_wait(peer,"priv-callpending");
- if (!valid_priv_reply(&opts, res2))
- res2 = 0;
- /* priv-callpending script:
- "I have a caller waiting, who introduces themselves as:"
- */
- if (!res2)
- res2 = ast_play_and_wait(peer,privintro);
- if (!valid_priv_reply(&opts, res2))
- res2 = 0;
- /* now get input from the called party, as to their choice */
- if( !res2 ) {
- /* XXX can we have both, or they are mutually exclusive ? */
- 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 caller 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 caller to the torture menus.
- Dial 4 to send this caller to a simple "go away" menu.
- */
- if (valid_priv_reply(&opts, res2))
- break;
- /* invalid option */
- res2 = ast_play_and_wait(peer, "vm-sorry");
- }
-
- 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;
- }
- ast_autoservice_stop(chan);
-
- 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);
- }
- ast_copy_string(status, "NOANSWER", sizeof(status));
- 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;
- 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;
- 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);
- 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: /* bad input or -1 if failure to start autoservice */
- /* 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");
- ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
- res=0;
- goto out;
- }
-
- /* XXX once again, this path is only taken in the case '1', so it could be
- * moved there, although i am not really sure that this is correct - maybe
- * the check applies to other cases as well.
- */
- /* 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])) {
- res = 0;
- } else {
- int digit = 0;
- /* 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;
-
- }
-
- if (chan && peer && ast_test_flag(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
- replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
- ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
- /* peer goes to the same context and extension as chan, so just copy info from chan*/
- ast_copy_string(peer->context, chan->context, sizeof(peer->context));
- ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
- peer->priority = chan->priority + 2;
- ast_pbx_start(peer);
- hanguptree(outgoing, NULL);
- if (continue_exec)
- *continue_exec = 1;
- res = 0;
- goto done;
- }
-
- if (ast_test_flag(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
- struct ast_app *theapp;
- const char *macro_result;
-
- res = ast_autoservice_start(chan);
- if (res) {
- ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
- res = -1;
- }
-
- theapp = pbx_findapp("Macro");
-
- if (theapp && !res) { /* XXX why check res here ? */
- replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
- res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
- 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 && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
- char *macro_transfer_dest;
-
- 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*/
- replace_macro_delimiter(macro_transfer_dest);
- if (!ast_parseable_goto(chan, macro_transfer_dest))
- ast_set_flag(peerflags, OPT_GO_ON);
-
- }
- }
- }
- }
-
- if (!res) {
- if (calldurationlimit > 0) {
- peer->whentohangup = time(NULL) + calldurationlimit;
- } else if (calldurationlimit != -1 && timelimit > 0) {
- /* Not enough granularity to make it less, but we can't use the special value 0 */
- peer->whentohangup = time(NULL) + 1;
- }
- 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) {
- struct ast_bridge_config config;
-
- 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);
- if (ast_test_flag(peerflags, OPT_CALLEE_PARK))
- ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
- if (ast_test_flag(peerflags, OPT_CALLER_PARK))
- ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
- if (ast_test_flag(peerflags, OPT_GO_ON))
- ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
-
- 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;
- config.end_bridge_callback = end_bridge_callback;
- config.end_bridge_callback_data = chan;
- config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
- 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);
- res = -1;
- goto done;
- }
- if (opermode &&
- (((!strncasecmp(chan->name,"Zap",3)) && (!strncasecmp(peer->name,"Zap",3))) ||
- ((!strncasecmp(chan->name,"Dahdi",5)) && (!strncasecmp(peer->name,"Dahdi",5)))))
- {
- struct oprmode oprmode;
-
- oprmode.peer = peer;
- oprmode.mode = opermode;
-
- ast_channel_setoption(chan,
- AST_OPTION_OPRMODE,&oprmode,sizeof(struct oprmode),0);
- }
- res = ast_bridge_call(chan,peer,&config);
- } else {
- res = -1;
- }
-
- 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);
- }
- ast_rtp_early_bridge(chan, NULL);
- 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) {
- if (calldurationlimit)
- chan->whentohangup = 0;
- res = 0;
- }
-done:
- ast_module_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, NULL);
-}
-
-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 = -1;
- struct ast_module_user *u;
- struct ast_flags peerflags;
-
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
- return -1;
- }
-
- u = ast_module_user_add(chan);
-
- announce = ast_strdupa(data);
-
- memset(&peerflags, 0, sizeof(peerflags));
-
- if ((dialdata = strchr(announce, '|'))) {
- *dialdata++ = '\0';
- if (sscanf(dialdata, "%d", &sleep) == 1) {
- sleep *= 1000;
- } else {
- ast_log(LOG_ERROR, "%s requires the numerical argument <sleep>\n",rapp);
- goto done;
- }
- if ((dialdata = strchr(dialdata, '|'))) {
- *dialdata++ = '\0';
- if (sscanf(dialdata, "%d", &loops) != 1) {
- ast_log(LOG_ERROR, "%s requires the numerical argument <loops>\n",rapp);
- goto done;
- }
- }
- }
-
- if ((dialdata = strchr(dialdata, '|'))) {
- *dialdata++ = '\0';
- } else {
- ast_log(LOG_ERROR, "%s requires more arguments\n",rapp);
- goto done;
- }
-
- if (sleep < 1000)
- sleep = 10000;
-
- if (!loops)
- loops = -1; /* run forever */
-
- context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
-
- res = 0;
- while (loops) {
- int continue_exec;
-
- chan->data = "Retrying";
- if (ast_test_flag(chan, AST_FLAG_MOH))
- ast_moh_stop(chan);
-
- res = dial_exec_full(chan, dialdata, &peerflags, &continue_exec);
- if (continue_exec)
- break;
-
- if (res == 0) {
- if (ast_test_flag(&peerflags, OPT_DTMF_EXIT)) {
- if (!ast_strlen_zero(announce)) {
- if (ast_fileexists(announce, NULL, chan->language) > 0) {
- if(!(res = ast_streamfile(chan, announce, chan->language)))
- ast_waitstream(chan, AST_DIGIT_ANY);
- } else
- ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", announce);
- }
- if (!res && sleep) {
- if (!ast_test_flag(chan, AST_FLAG_MOH))
- ast_moh_start(chan, NULL, NULL);
- res = ast_waitfordigit(chan, sleep);
- }
- } else {
- if (!ast_strlen_zero(announce)) {
- if (ast_fileexists(announce, NULL, chan->language) > 0) {
- if (!(res = ast_streamfile(chan, announce, chan->language)))
- res = ast_waitstream(chan, "");
- } else
- ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", announce);
- }
- if (sleep) {
- if (!ast_test_flag(chan, AST_FLAG_MOH))
- ast_moh_start(chan, NULL, 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 (loops == 0)
- res = 0;
- else if (res == 1)
- res = 0;
-
- if (ast_test_flag(chan, AST_FLAG_MOH))
- ast_moh_stop(chan);
- done:
- ast_module_user_remove(u);
- return res;
-}
-
-static int unload_module(void)
-{
- int res;
-
- res = ast_unregister_application(app);
- res |= ast_unregister_application(rapp);
-
- ast_module_user_hangup_all();
-
- return res;
-}
-
-static 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;
-}
-
-AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");