app_dial.c

	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");
		/* XXX should we set status to DENY ? */
		/* XXX what about the privacy flags ? */
		break;
	}

	if (res2 == '1') { /* the only case where we actually connect */
		/* 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(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
			ast_filedelete(pa->privintro, NULL);
			if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
				ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
			else
				ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
		}
		return 0; /* the good exit path */
	} else {
		return -1;
	}
}

/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
static int setup_privacy_args(struct privacy_args *pa,
	struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
{
	char callerid[60];
	int res;
	char *l;

	if (ast_channel_caller(chan)->id.number.valid
		&& !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
		l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
		ast_shrink_phone_number(l);
		if (ast_test_flag64(opts, OPT_PRIVACY) ) {
			ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
			pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
		} else {
			ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
			pa->privdb_val = AST_PRIVACY_UNKNOWN;
		}
	} else {
		char *tnam, *tn2;

		tnam = ast_strdupa(ast_channel_name(chan));
		/* clean the channel name so slashes don't try to end up in disk file name */
		for (tn2 = tnam; *tn2; tn2++) {
			if (*tn2 == '/')  /* any other chars to be afraid of? */
				*tn2 = '=';
		}
		ast_verb(3, "Privacy-- callerid is empty\n");

		snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
		l = callerid;
		pa->privdb_val = AST_PRIVACY_UNKNOWN;
	}

	ast_copy_string(pa->privcid, l, sizeof(pa->privcid));

	if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
		/* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
		ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
		pa->privdb_val = AST_PRIVACY_ALLOW;
	} else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
		ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
	}

	if (pa->privdb_val == AST_PRIVACY_DENY) {
		ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
		ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
		return 0;
	} else if (pa->privdb_val == AST_PRIVACY_KILL) {
		ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
		return 0; /* Is this right? */
	} else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
		ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
		return 0; /* is this right??? */
	} else if (pa->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(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
		if ((res = ast_mkdir(pa->privintro, 0755))) {
			ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
			return -1;
		}

		snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
		if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->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:"

			*/
			int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
			ast_answer(chan);
			res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, NULL, silencethreshold, 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 pa.privintro file */
			if (res == -1) {
				/* Delete the file regardless since they hung up during recording */
				ast_filedelete(pa->privintro, NULL);
				if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
					ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
				else
					ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
				return -1;
			}
			if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
				ast_waitstream(chan, "");
		}
	}
	return 1; /* success */
}

static void end_bridge_callback(void *data)
{
	struct ast_channel *chan = data;

	ast_channel_lock(chan);
	ast_channel_stage_snapshot(chan);
	set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
	set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
	ast_channel_stage_snapshot_done(chan);
	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_handle_playtones(struct ast_channel *chan, const char *data)
{
	struct ast_tone_zone_sound *ts = NULL;
	int res;
	const char *str = data;

	if (ast_strlen_zero(str)) {
		ast_debug(1,"Nothing to play\n");
		return -1;
	}

	ts = ast_get_indication_tone(ast_channel_zone(chan), str);

	if (ts && ts->data[0]) {
		res = ast_playtones_start(chan, 0, ts->data, 0);
	} else {
		res = -1;
	}

	if (ts) {
		ts = ast_tone_zone_sound_unref(ts);
	}

	if (res) {
		ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
	}

	return res;
}

/*!
 * \internal
 * \brief Setup the after bridge goto location on the peer.
 * \since 12.0.0
 *
 * \param chan Calling channel for bridge.
 * \param peer Peer channel for bridge.
 * \param opts Dialing option flags.
 * \param opt_args Dialing option argument strings.
 *
 * \return Nothing
 */
static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
{
	const char *context;
	const char *extension;
	int priority;

	if (ast_test_flag64(opts, OPT_PEER_H)) {
		ast_channel_lock(chan);
		context = ast_strdupa(ast_channel_context(chan));
		ast_channel_unlock(chan);
		ast_bridge_set_after_h(peer, context);
	} else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
		ast_channel_lock(chan);
		context = ast_strdupa(ast_channel_context(chan));
		extension = ast_strdupa(ast_channel_exten(chan));
		priority = ast_channel_priority(chan);
		ast_channel_unlock(chan);
		ast_bridge_set_after_go_on(peer, context, extension, priority,
			opt_args[OPT_ARG_CALLEE_GO_ON]);
	}
}

static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
{
	int res = -1; /* default: error */
	char *rest, *cur; /* scan the list of destinations */
	struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
	struct chanlist *outgoing;
	struct chanlist *tmp;
	struct ast_channel *peer;
	int to; /* timeout */
	struct cause_args num = { chan, 0, 0, 0 };
	int cause;

	struct ast_bridge_config config = { { 0, } };
	struct timeval calldurationlimit = { 0, };
	char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
	struct privacy_args pa = {
		.sentringing = 0,
		.privdb_val = 0,
		.status = "INVALIDARGS",
	};
	int sentringing = 0, moh = 0;
	const char *outbound_group = NULL;
	int result = 0;
	char *parse;
	int opermode = 0;
	int delprivintro = 0;
	AST_DECLARE_APP_ARGS(args,
		AST_APP_ARG(peers);
		AST_APP_ARG(timeout);
		AST_APP_ARG(options);
		AST_APP_ARG(url);
	);
	struct ast_flags64 opts = { 0, };
	char *opt_args[OPT_ARG_ARRAY_SIZE];
	int fulldial = 0, num_dialed = 0;
	int ignore_cc = 0;
	char device_name[AST_CHANNEL_NAME];
	char forced_clid_name[AST_MAX_EXTENSION];
	char stored_clid_name[AST_MAX_EXTENSION];
	int force_forwards_only;	/*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
	/*!
	 * \brief Forced CallerID party information to send.
	 * \note This will not have any malloced strings so do not free it.
	 */
	struct ast_party_id forced_clid;
	/*!
	 * \brief Stored CallerID information if needed.
	 *
	 * \note If OPT_ORIGINAL_CLID set then this is the o option
	 * CallerID.  Otherwise it is the dialplan extension and hint
	 * name.
	 *
	 * \note This will not have any malloced strings so do not free it.
	 */
	struct ast_party_id stored_clid;
	/*!
	 * \brief CallerID party information to store.
	 * \note This will not have any malloced strings so do not free it.
	 */
	struct ast_party_caller caller;
	int max_forwards;

	/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
	ast_channel_lock(chan);
	ast_channel_stage_snapshot(chan);
	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, "ANSWEREDTIME_MS", "");
	pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
	pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
	pbx_builtin_setvar_helper(chan, "RINGTIME", "");
	pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
	pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
	pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
	ast_channel_stage_snapshot_done(chan);
	max_forwards = ast_max_forwards_get(chan);
	ast_channel_unlock(chan);

	if (max_forwards <= 0) {
		ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
				ast_channel_name(chan));
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
		return -1;
	}

	if (ast_strlen_zero(data)) {
		ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
		return -1;
	}

	if (ast_check_hangup_locked(chan)) {
		/*
		 * Caller hung up before we could dial.  If dial is executed
		 * within an AGI then the AGI has likely eaten all queued
		 * frames before executing the dial in DeadAGI mode.  With
		 * the caller hung up and no pending frames from the caller's
		 * read queue, dial would not know that the call has hung up
		 * until a called channel answers.  It is rather annoying to
		 * whoever just answered the non-existent call.
		 *
		 * Dial should not continue execution in DeadAGI mode, hangup
		 * handlers, or the h exten.
		 */
		ast_verb(3, "Caller hung up before dial.\n");
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
		return -1;
	}

	parse = ast_strdupa(data);

	AST_STANDARD_APP_ARGS(args, parse);

	if (!ast_strlen_zero(args.options) &&
		ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
		goto done;
	}

	if (ast_strlen_zero(args.peers)) {
		ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
		goto done;
	}

	if (ast_cc_call_init(chan, &ignore_cc)) {
		goto done;
	}

	if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
		delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);

		if (delprivintro < 0 || delprivintro > 1) {
			ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
			delprivintro = 0;
		}
	}

	if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
		opt_args[OPT_ARG_RINGBACK] = NULL;
	}

	if (ast_test_flag64(&opts, OPT_OPERMODE)) {
		opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
		ast_verb(3, "Setting operator services mode to %d.\n", opermode);
	}

	if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
		calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
		if (!calldurationlimit.tv_sec) {
			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", pa.status);
			goto done;
		}
		ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
	}

	if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
		dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
		dtmfcalled = strsep(&dtmf_progress, ":");
		dtmfcalling = strsep(&dtmf_progress, ":");
	}

	if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
		if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
			goto done;
	}

	/* Setup the forced CallerID information to send if used. */
	ast_party_id_init(&forced_clid);
	force_forwards_only = 0;
	if (ast_test_flag64(&opts, OPT_FORCECLID)) {
		if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
			ast_channel_lock(chan);
			forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
			ast_channel_unlock(chan);
			forced_clid_name[0] = '\0';
			forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
				sizeof(forced_clid_name), chan);
			force_forwards_only = 1;
		} else {
			/* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
			ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
				&forced_clid.number.str);
		}
		if (!ast_strlen_zero(forced_clid.name.str)) {
			forced_clid.name.valid = 1;
		}
		if (!ast_strlen_zero(forced_clid.number.str)) {
			forced_clid.number.valid = 1;
		}
	}
	if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
		&& !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
		forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
	}
	forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
	if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
		&& !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
		int pres;

		pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
		if (0 <= pres) {
			forced_clid.number.presentation = pres;
		}
	}

	/* Setup the stored CallerID information if needed. */
	ast_party_id_init(&stored_clid);
	if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
		if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
			ast_channel_lock(chan);
			ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
			if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
				stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
			}
			if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
				stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
			}
			if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
				stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
			}
			if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
				stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
			}
			ast_channel_unlock(chan);
		} else {
			/* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
			ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
				&stored_clid.number.str);
			if (!ast_strlen_zero(stored_clid.name.str)) {
				stored_clid.name.valid = 1;
			}
			if (!ast_strlen_zero(stored_clid.number.str)) {
				stored_clid.number.valid = 1;
			}
		}
	} else {
		/*
		 * In case the new channel has no preset CallerID number by the
		 * channel driver, setup the dialplan extension and hint name.
		 */
		stored_clid_name[0] = '\0';
		stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
			sizeof(stored_clid_name), chan);
		if (ast_strlen_zero(stored_clid.name.str)) {
			stored_clid.name.str = NULL;
		} else {
			stored_clid.name.valid = 1;
		}
		ast_channel_lock(chan);
		stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
		stored_clid.number.valid = 1;
		ast_channel_unlock(chan);
	}

	if (ast_test_flag64(&opts, OPT_RESETCDR)) {
		ast_cdr_reset(ast_channel_name(chan), 0);
	}
	if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
		opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));

	if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
		res = setup_privacy_args(&pa, &opts, opt_args, chan);
		if (res <= 0)
			goto out;
		res = -1; /* reset default */
	}

	if (continue_exec)
		*continue_exec = 0;

	/* If a channel group has been specified, get it for use when we create peer channels */

	ast_channel_lock(chan);
	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 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
		outbound_group = ast_strdupa(outbound_group);
	}
	ast_channel_unlock(chan);

	/* Set per dial instance flags.  These flags are also passed back to RetryDial. */
	ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
		| OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
		| OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);

	/* PREDIAL: Run gosub on the caller's channel */
	if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
		&& !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
		ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
		ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
	}

	/* loop through the list of dial destinations */
	rest = args.peers;
	while ((cur = strsep(&rest, "&")) ) {
		struct ast_channel *tc; /* channel for this destination */
		/* Get a technology/resource pair */
		char *number = cur;
		char *tech = strsep(&number, "/");
		size_t tech_len;
		size_t number_len;
		struct ast_stream_topology *topology;

		num_dialed++;
		if (ast_strlen_zero(number)) {
			ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
			goto out;
		}

		tech_len = strlen(tech) + 1;
		number_len = strlen(number) + 1;
		tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
		if (!tmp) {
			goto out;
		}

		/* Save tech, number, and interface. */
		cur = tmp->stuff;
		strcpy(cur, tech);
		tmp->tech = cur;
		cur += tech_len;
		strcpy(cur, tech);
		cur[tech_len - 1] = '/';
		tmp->interface = cur;
		cur += tech_len;
		strcpy(cur, number);
		tmp->number = cur;

		if (opts.flags) {
			/* Set per outgoing call leg options. */
			ast_copy_flags64(tmp, &opts,
				OPT_CANCEL_ELSEWHERE |
				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_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
				OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
				OPT_RING_WITH_EARLY_MEDIA);
			ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
		}

		/* Request the peer */

		ast_channel_lock(chan);
		/*
		 * Seed the chanlist's connected line information with previously
		 * acquired connected line info from the incoming channel.  The
		 * previously acquired connected line info could have been set
		 * through the CONNECTED_LINE dialplan function.
		 */
		ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));

		topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));

		ast_channel_unlock(chan);

		tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);

		ast_stream_topology_free(topology);

		if (!tc) {
			/* 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",
				tmp->tech, cause, ast_cause2str(cause));
			handle_cause(cause, &num);
			if (!rest) {
				/* we are on the last destination */
				ast_channel_hangupcause_set(chan, cause);
			}
			if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
				if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
					ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
				}
			}
			chanlist_free(tmp);
			continue;
		}

		ast_channel_get_device_name(tc, device_name, sizeof(device_name));
		if (!ignore_cc) {
			ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
		}

		ast_channel_lock_both(tc, chan);
		ast_channel_stage_snapshot(tc);

		pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);

		/* Setup outgoing SDP to match incoming one */
		if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
			/* We are on the only destination. */
			ast_rtp_instance_early_bridge_make_compatible(tc, chan);
		}

		/* Inherit specially named variables from parent channel */
		ast_channel_inherit_variables(chan, tc);
		ast_channel_datastore_inherit(chan, tc);
		ast_max_forwards_decrement(tc);

		ast_channel_appl_set(tc, "AppDial");
		ast_channel_data_set(tc, "(Outgoing Line)");

		memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));

		/* Determine CallerID to store in outgoing channel. */
		ast_party_caller_set_init(&caller, ast_channel_caller(tc));
		if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
			caller.id = stored_clid;
			ast_channel_set_caller_event(tc, &caller, NULL);
			ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
		} else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
			ast_channel_caller(tc)->id.number.str, NULL))) {
			/*
			 * The new channel has no preset CallerID number by the channel
			 * driver.  Use the dialplan extension and hint name.
			 */
			caller.id = stored_clid;
			if (!caller.id.name.valid
				&& !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
					ast_channel_connected(chan)->id.name.str, NULL))) {
				/*
				 * No hint name available.  We have a connected name supplied by
				 * the dialplan we can use instead.
				 */
				caller.id.name.valid = 1;
				caller.id.name = ast_channel_connected(chan)->id.name;
			}
			ast_channel_set_caller_event(tc, &caller, NULL);
			ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
		} else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
			NULL))) {
			/* The new channel has no preset CallerID name by the channel driver. */
			if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
				ast_channel_connected(chan)->id.name.str, NULL))) {
				/*
				 * We have a connected name supplied by the dialplan we can
				 * use instead.
				 */
				caller.id.name.valid = 1;
				caller.id.name = ast_channel_connected(chan)->id.name;
				ast_channel_set_caller_event(tc, &caller, NULL);
			}
		}

		/* Determine CallerID for outgoing channel to send. */
		if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
			struct ast_party_connected_line connected;

			ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
			connected.id = forced_clid;
			ast_channel_set_connected_line(tc, &connected, NULL);
		} else {
			ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
		}

		ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));

		ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;

		ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
		if (ast_strlen_zero(ast_channel_musicclass(tc))) {
			ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
		}

		/* Pass ADSI CPE and transfer capability */
		ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
		ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));

		/* If we have an outbound group, set this peer channel to it */
		if (outbound_group)
			ast_app_group_set_channel(tc, outbound_group);
		/* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
		if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
			ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);

		/* Check if we're forced by configuration */
		if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
			 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);


		/* Inherit context and extension */
		ast_channel_dialcontext_set(tc, ast_strlen_zero(ast_channel_macrocontext(chan)) ? ast_channel_context(chan) : ast_channel_macrocontext(chan));
		if (!ast_strlen_zero(ast_channel_macroexten(chan)))
			ast_channel_exten_set(tc, ast_channel_macroexten(chan));
		else
			ast_channel_exten_set(tc, ast_channel_exten(chan));

		ast_channel_stage_snapshot_done(tc);

		/* Save the original channel name to detect call pickup masquerading in. */
		tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));

		ast_channel_unlock(tc);
		ast_channel_unlock(chan);

		/* Put channel in the list of outgoing thingies. */
		tmp->chan = tc;
		AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
	}

	/*
	 * PREDIAL: Run gosub on all of the callee channels
	 *
	 * We run the callee predial before ast_call() in case the user
	 * wishes to do something on the newly created channels before
	 * the channel does anything important.
	 *
	 * Inside the target gosub we will be able to do something with
	 * the newly created channel name ie: now the calling channel
	 * can know what channel will be used to call the destination
	 * ex: now we will know that SIP/abc-123 is calling SIP/def-124
	 */
	if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
		&& !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
		&& !AST_LIST_EMPTY(&out_chans)) {
		const char *predial_callee;

		ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
		predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
		if (predial_callee) {
			ast_autoservice_start(chan);
			AST_LIST_TRAVERSE(&out_chans, tmp, node) {
				ast_pre_call(tmp->chan, predial_callee);
			}
			ast_autoservice_stop(chan);
			ast_free((char *) predial_callee);
		}
	}

	/* Start all outgoing calls */
	AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
		res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
		ast_channel_lock(chan);

		/* check the results of ast_call */
		if (res) {
			/* Again, keep going even if there's an error */
			ast_debug(1, "ast call on peer returned %d\n", res);
			ast_verb(3, "Couldn't call %s\n", tmp->interface);
			if (ast_channel_hangupcause(tmp->chan)) {
				ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
			}
			ast_channel_unlock(chan);
			ast_cc_call_failed(chan, tmp->chan, tmp->interface);
			ast_hangup(tmp->chan);
			tmp->chan = NULL;
			AST_LIST_REMOVE_CURRENT(node);
			chanlist_free(tmp);
			continue;
		}

		ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
		ast_channel_unlock(chan);

		ast_verb(3, "Called %s\n", tmp->interface);
		ast_set_flag64(tmp, DIAL_STILLGOING);

		/* If this line is up, don't try anybody else */
		if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
			break;
		}
	}
	AST_LIST_TRAVERSE_SAFE_END;

	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'. Setting timeout to infinite\n", args.timeout);
			to = -1;
		}
	}

	outgoing = AST_LIST_FIRST(&out_chans);
	if (!outgoing) {
		strcpy(pa.status, "CHANUNAVAIL");
		if (fulldial == num_dialed) {
			res = -1;
			goto out;
		}
	} else {
		/* Our status will at least be NOANSWER */
		strcpy(pa.status, "NOANSWER");
		if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
			moh = 1;
			if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
				char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
				ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
				ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
				ast_channel_musicclass_set(chan, original_moh);
			} else {
				ast_moh_start(chan, NULL, NULL);
			}
			ast_indicate(chan, AST_CONTROL_PROGRESS);
		} else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
			if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
				if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
					ast_indicate(chan, AST_CONTROL_RINGING);
					sentringing++;
				} else {
					ast_indicate(chan, AST_CONTROL_PROGRESS);
				}
			} else {
				ast_indicate(chan, AST_CONTROL_RINGING);
				sentringing++;
			}
		}
	}

	peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
		dtmf_progress, ignore_cc, &forced_clid, &stored_clid);

	if (!peer) {
		if (result) {
			res = result;
		} else if (to) { /* Musta gotten hung up */
			res = -1;
		} else { /* Nobody answered, next please? */
			res = 0;
		}
	} else {
		const char *number;
		const char *name;
		int dial_end_raised = 0;
		int cause = -1;

		if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
			ast_answer(chan);
		}

		/* 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.  */

		if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
			&& !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
			cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
			if (cause <= 0) {
				if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
					cause = 0;
				} else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
					|| cause < 0) {
					ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
						opt_args[OPT_ARG_HANGUPCAUSE]);
					cause = -1;
				}
			}
		}
		hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);

		/* If appropriate, log that we have a destination channel and set the answer time */

		ast_channel_lock(peer);
		name = ast_strdupa(ast_channel_name(peer));

		number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
		if (ast_strlen_zero(number)) {
			number = NULL;
		} else {
			number = ast_strdupa(number);
		}
		ast_channel_unlock(peer);

		ast_channel_lock(chan);
		ast_channel_stage_snapshot(chan);

		strcpy(pa.status, "ANSWER");
		pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);

		pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
		pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);

		ast_channel_stage_snapshot_done(chan);
		ast_channel_unlock(chan);

		if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
			ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
			ast_channel_sendurl( peer, args.url );
		}
		if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
			if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
				ast_channel_publish_dial(chan, peer, NULL, pa.status);
				/* hang up on the callee -- he didn't want to talk anyway! */
				ast_autoservice_chan_hangup_peer(chan, peer);
				res = 0;
				goto out;
			}
		}
		if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
			res = 0;
		} else {
			int digit = 0;
			struct ast_channel *chans[2];
			struct ast_channel *active_chan;

			chans[0] = chan;
			chans[1] = peer;

			/* we need to stream the announcement to the called party when the OPT_ARG_ANNOUNCE (-A) is setted */

			/* stream the file */
			res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
			if (res) {
				res = 0;
				ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
			}

			ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
			while (ast_channel_stream(peer)) {
				int ms;

				ms = ast_sched_wait(ast_channel_sched(peer));

				if (ms < 0 && !ast_channel_timingfunc(peer)) {
					ast_stopstream(peer);
					break;
				}
				if (ms < 0)
					ms = 1000;

				active_chan = ast_waitfor_n(chans, 2, &ms);
				if (active_chan) {
					struct ast_channel *other_chan;
					struct ast_frame *fr = ast_read(active_chan);

					if (!fr) {
						ast_autoservice_chan_hangup_peer(chan, peer);
						res = -1;
						goto done;
					}
					switch (fr->frametype) {
					case AST_FRAME_DTMF_END:
						digit = fr->subclass.integer;
						if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
							ast_stopstream(peer);
							res = ast_senddigit(chan, digit, 0);
						}
						break;
					case AST_FRAME_CONTROL:
						switch (fr->subclass.integer) {
						case AST_CONTROL_HANGUP:
							ast_frfree(fr);
							ast_autoservice_chan_hangup_peer(chan, peer);
							res = -1;
							goto done;
						case AST_CONTROL_CONNECTED_LINE:
							/* Pass COLP update to the other channel. */
							if (active_chan == chan) {
								other_chan = peer;
							} else {
								other_chan = chan;
							}
							if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)
								&& ast_channel_connected_line_macro(active_chan,
									other_chan, fr, other_chan == chan, 1)) {
								ast_indicate_data(other_chan, fr->subclass.integer,
									fr->data.ptr, fr->datalen);
							}
							break;
						default:
							break;
						}
						break;
					default:
						/* Ignore all others */
						break;
					}
					ast_frfree(fr);
				}
				ast_sched_runq(ast_channel_sched(peer));
			}
			ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
		}

		if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
			/* chan and peer are going into the PBX; as such neither are considered
			 * outgoing channels any longer */
			ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);

			ast_replace_subargument_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*/