app_dial.c File Reference

dial() & retrydial() - Trivial application to dial a channel and send an URL on answer More...

#include "asterisk.h"
#include <sys/time.h>
#include <sys/signal.h>
#include <sys/stat.h>
#include <netinet/in.h>
#include "asterisk/paths.h"
#include "asterisk/lock.h"
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/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"
#include "asterisk/dsp.h"

Include dependency graph for app_dial.c:

Go to the source code of this file.

Data Structures
struct	cause_args
struct	chanlist
	List of channel drivers. More...
struct	privacy_args
Defines
#define	AST_MAX_WATCHERS   256
#define	CAN_EARLY_BRIDGE(flags, chan, peer)
#define	DIAL_NOFORWARDHTML   ((uint64_t)1 << 32)
#define	DIAL_STILLGOING   (1 << 31)
#define	OPT_CALLEE_GO_ON   ((uint64_t)1 << 35)
#define	OPT_CANCEL_ELSEWHERE   ((uint64_t)1 << 33)
#define	OPT_PEER_H   ((uint64_t)1 << 34)
#define	S_REPLACE(s, new_val)
Enumerations
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_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), OPT_CALLEE_GOSUB = (1 << 28),   OPT_CALLEE_MIXMONITOR = (1 << 29), OPT_CALLER_MIXMONITOR = (1 << 30) }
enum	{   OPT_ARG_ANNOUNCE = 0, OPT_ARG_SENDDTMF, OPT_ARG_GOTO, OPT_ARG_DURATION_LIMIT,   OPT_ARG_MUSICBACK, OPT_ARG_CALLEE_MACRO, OPT_ARG_CALLEE_GOSUB, OPT_ARG_CALLEE_GO_ON,   OPT_ARG_PRIVACY, OPT_ARG_DURATION_STOP, OPT_ARG_OPERMODE, OPT_ARG_ARRAY_SIZE }
Functions
static void	__reg_module (void)
static void	__unreg_module (void)
static int	detect_disconnect (struct ast_channel *chan, char code, struct ast_str *featurecode)
static int	dial_exec (struct ast_channel *chan, void *data)
static int	dial_exec_full (struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
static void	do_forward (struct chanlist *o, struct cause_args *num, struct ast_flags64 *peerflags, int single)
static int	do_timelimit (struct ast_channel *chan, struct ast_bridge_config *config, char *parse, struct timeval *calldurationlimit)
static void	end_bridge_callback (void *data)
static void	end_bridge_callback_data_fixup (struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator)
static const char *	get_cid_name (char *name, int namelen, struct ast_channel *chan)
static void	handle_cause (int cause, struct cause_args *num)
static void	hanguptree (struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
static int	load_module (void)
static int	onedigit_goto (struct ast_channel *chan, const char *context, char exten, int pri)
static void	replace_macro_delimiter (char *s)
static int	retrydial_exec (struct ast_channel *chan, void *data)
static void	senddialendevent (const struct ast_channel *src, const char *dialstatus)
static void	senddialevent (struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
static int	setup_privacy_args (struct privacy_args *pa, struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
	returns 1 if successful, 0 or <0 if the caller should 'goto out'
static int	unload_module (void)
static int	valid_priv_reply (struct ast_flags64 *opts, int res)
static struct ast_channel *	wait_for_answer (struct ast_channel *in, struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags, struct privacy_args *pa, const struct cause_args *num_in, int *result)
Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Dialing Application" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "a9c98e5d177805051735cb5b0b16b0a0" , .load = load_module, .unload = unload_module, }
static char *	app = "Dial"
static struct ast_module_info *	ast_module_info = &__mod_info
static char *	descrip
static struct ast_app_option	dial_exec_options [128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = ((uint64_t)1 << 33) }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'e' ] = { .flag = ((uint64_t)1 << 34) }, [ 'f' ] = { .flag = OPT_FORCECLID }, [ 'F' ] = { .flag = ((uint64_t)1 << 35) , .arg_index = OPT_ARG_CALLEE_GO_ON + 1 }, [ 'g' ] = { .flag = OPT_GO_ON }, [ 'G' ] = { .flag = OPT_GOTO , .arg_index = OPT_ARG_GOTO + 1 }, [ 'h' ] = { .flag = OPT_CALLEE_HANGUP }, [ 'H' ] = { .flag = OPT_CALLER_HANGUP }, [ 'i' ] = { .flag = OPT_IGNORE_FORWARDING }, [ 'k' ] = { .flag = OPT_CALLEE_PARK }, [ 'K' ] = { .flag = OPT_CALLER_PARK }, [ 'L' ] = { .flag = OPT_DURATION_LIMIT , .arg_index = OPT_ARG_DURATION_LIMIT + 1 }, [ 'm' ] = { .flag = OPT_MUSICBACK , .arg_index = OPT_ARG_MUSICBACK + 1 }, [ 'M' ] = { .flag = OPT_CALLEE_MACRO , .arg_index = OPT_ARG_CALLEE_MACRO + 1 }, [ 'n' ] = { .flag = OPT_SCREEN_NOINTRO }, [ 'N' ] = { .flag = OPT_SCREEN_NOCLID }, [ 'o' ] = { .flag = OPT_ORIGINAL_CLID }, [ 'O' ] = { .flag = OPT_OPERMODE , .arg_index = OPT_ARG_OPERMODE + 1 }, [ 'p' ] = { .flag = OPT_SCREENING }, [ 'P' ] = { .flag = OPT_PRIVACY , .arg_index = OPT_ARG_PRIVACY + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'U' ] = { .flag = OPT_CALLEE_GOSUB , .arg_index = OPT_ARG_CALLEE_GOSUB + 1 }, [ 'w' ] = { .flag = OPT_CALLEE_MONITOR }, [ 'W' ] = { .flag = OPT_CALLER_MONITOR }, [ 'x' ] = { .flag = OPT_CALLEE_MIXMONITOR }, [ 'X' ] = { .flag = OPT_CALLER_MIXMONITOR }, }
static char *	rapp = "RetryDial"
static char *	rdescrip
static char *	rsynopsis = "Place a call, retrying on failure allowing optional exit extension."
static char *	synopsis = "Place a call and connect to the current channel"

---

Detailed Description

dial() & retrydial() - Trivial application to dial a channel and send an URL on answer

Author:

Mark Spencer <markster@digium.com>

Definition in file app_dial.c.

---

Define Documentation

#define AST_MAX_WATCHERS   256

Definition at line 374 of file app_dial.c.

#define CAN_EARLY_BRIDGE	(	flags,
		chan,
		peer		)

Value:

(!ast_test_flag64(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)

Definition at line 341 of file app_dial.c.

Referenced by wait_for_answer().

#define DIAL_NOFORWARDHTML   ((uint64_t)1 << 32)

Definition at line 284 of file app_dial.c.

Referenced by dial_exec_full(), and wait_for_answer().

#define DIAL_STILLGOING   (1 << 31)

Definition at line 283 of file app_dial.c.

Referenced by dial_exec_full(), do_forward(), and wait_for_answer().

#define OPT_CALLEE_GO_ON   ((uint64_t)1 << 35)

Definition at line 287 of file app_dial.c.

Referenced by dial_exec_full().

#define OPT_CANCEL_ELSEWHERE   ((uint64_t)1 << 33)

Definition at line 285 of file app_dial.c.

Referenced by dial_exec_full().

#define OPT_PEER_H   ((uint64_t)1 << 34)

Definition at line 286 of file app_dial.c.

Referenced by dial_exec_full().

#define S_REPLACE	(	s,
		new_val		)

Value:

do {           \
      if (s)         \
         ast_free(s);   \
      s = (new_val);    \
   } while (0)

Definition at line 425 of file app_dial.c.

---

Enumeration Type Documentation

anonymous enum

Enumerator:

OPT_ANNOUNCE
OPT_RESETCDR
OPT_DTMF_EXIT
OPT_SENDDTMF
OPT_FORCECLID
OPT_GO_ON
OPT_CALLEE_HANGUP
OPT_CALLER_HANGUP
OPT_DURATION_LIMIT
OPT_MUSICBACK
OPT_CALLEE_MACRO
OPT_SCREEN_NOINTRO
OPT_SCREEN_NOCLID
OPT_ORIGINAL_CLID
OPT_SCREENING
OPT_PRIVACY
OPT_RINGBACK
OPT_DURATION_STOP
OPT_CALLEE_TRANSFER
OPT_CALLER_TRANSFER
OPT_CALLEE_MONITOR
OPT_CALLER_MONITOR
OPT_GOTO
OPT_OPERMODE
OPT_CALLEE_PARK
OPT_CALLER_PARK
OPT_IGNORE_FORWARDING
OPT_CALLEE_GOSUB
OPT_CALLEE_MIXMONITOR
OPT_CALLER_MIXMONITOR

Definition at line 250 of file app_dial.c.

     {
   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_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),
   OPT_CALLEE_GOSUB =      (1 << 28),
   OPT_CALLEE_MIXMONITOR = (1 << 29),
   OPT_CALLER_MIXMONITOR = (1 << 30),
};

anonymous enum

Enumerator:

OPT_ARG_ANNOUNCE
OPT_ARG_SENDDTMF
OPT_ARG_GOTO
OPT_ARG_DURATION_LIMIT
OPT_ARG_MUSICBACK
OPT_ARG_CALLEE_MACRO
OPT_ARG_CALLEE_GOSUB
OPT_ARG_CALLEE_GO_ON
OPT_ARG_PRIVACY
OPT_ARG_DURATION_STOP
OPT_ARG_OPERMODE
OPT_ARG_ARRAY_SIZE

Definition at line 289 of file app_dial.c.

     {
   OPT_ARG_ANNOUNCE = 0,
   OPT_ARG_SENDDTMF,
   OPT_ARG_GOTO,
   OPT_ARG_DURATION_LIMIT,
   OPT_ARG_MUSICBACK,
   OPT_ARG_CALLEE_MACRO,
   OPT_ARG_CALLEE_GOSUB,
   OPT_ARG_CALLEE_GO_ON,
   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,
};

---

Function Documentation

static void __reg_module

(

void

)

[static]

Definition at line 2204 of file app_dial.c.

static void __unreg_module

(

void

)

[static]

Definition at line 2204 of file app_dial.c.

static int detect_disconnect	(	struct ast_channel *	chan,
		char	code,
		struct ast_str *	featurecode
	)			[static]

Definition at line 899 of file app_dial.c.

References ast_feature_detect(), AST_FEATURE_DISCONNECT, AST_FEATURE_RETURN_STOREDIGITS, ast_str_append(), ast_str_buffer, ast_str_reset(), and ast_call_feature::feature_mask.

Referenced by wait_for_answer().

{
   struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
   struct ast_call_feature feature = { 0, };
   int res;

   ast_str_append(&featurecode, 1, "%c", code);

   res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);

   if (res != AST_FEATURE_RETURN_STOREDIGITS) {
      ast_str_reset(featurecode);
   }
   if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
      return 1;
   }

   return 0;
}

static int dial_exec	(	struct ast_channel *	chan,
		void *	data
	)			[static]

Definition at line 2054 of file app_dial.c.

References dial_exec_full().

Referenced by load_module().

{
   struct ast_flags64 peerflags;

   memset(&peerflags, 0, sizeof(peerflags));

   return dial_exec_full(chan, data, &peerflags, NULL);
}

static int dial_exec_full	(	struct ast_channel *	chan,
		void *	data,
		struct ast_flags64 *	peerflags,
		int *	continue_exec
	)			[static]

Definition at line 1310 of file app_dial.c.

References __ast_answer(), ast_channel::_state, accountcode, ast_channel::adsicpe, ast_channel::appl, asprintf, AST_APP_ARG, ast_app_group_set_channel(), ast_app_parse_options64(), ast_autoservice_start(), ast_autoservice_stop(), ast_bridge_call(), ast_call(), ast_calloc, ast_cause2str(), AST_CAUSE_INVALID_NUMBER_FORMAT, ast_cdr_reset(), ast_cdr_setdestchan(), ast_channel_datastore_add(), ast_channel_datastore_find(), ast_channel_datastore_inherit(), ast_channel_datastore_remove(), ast_channel_early_bridge(), ast_channel_inherit_variables(), ast_channel_lock, ast_channel_make_compatible(), ast_channel_sendurl(), ast_channel_setoption(), ast_channel_supports_html(), ast_channel_unlock, ast_check_hangup(), AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_copy_flags64, ast_copy_string(), ast_datastore_alloc, ast_datastore_free(), ast_deactivate_generator(), ast_debug, AST_DECLARE_APP_ARGS, AST_DIGIT_ANY, ast_dtmf_stream(), ast_exists_extension(), AST_FEATURE_AUTOMIXMON, AST_FEATURE_AUTOMON, AST_FEATURE_DISCONNECT, AST_FEATURE_NO_H_EXTEN, AST_FEATURE_PARKCALL, AST_FEATURE_REDIRECT, AST_FLAG_IN_AUTOLOOP, ast_free, ast_hangup(), ast_indicate(), AST_LIST_HEAD, AST_LIST_HEAD_INIT, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, ast_log(), AST_MAX_EXTENSION, ast_moh_start(), ast_moh_stop(), AST_OPTION_OPRMODE, ast_parseable_goto(), AST_PBX_INCOMPLETE, ast_pbx_run_args(), ast_pbx_start(), AST_PRIVACY_UNKNOWN, ast_request(), ast_rtp_make_compatible(), ast_senddigit(), ast_set2_flag, ast_set2_flag64, ast_set_callerid(), ast_set_flag, ast_set_flag64, ast_spawn_extension(), AST_STANDARD_APP_ARGS, AST_STATE_UP, ast_strdup, ast_strdupa, ast_streamfile(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_test_flag64, ast_tvadd(), ast_tvnow(), ast_tvzero(), ast_verb, ast_waitstream(), ast_channel::cdr, ast_channel::cdrflags, chanlist::chan, ast_channel::cid, ast_callerid::cid_ani, ast_callerid::cid_ani2, ast_callerid::cid_name, ast_callerid::cid_num, ast_callerid::cid_pres, ast_callerid::cid_rdnis, ast_callerid::cid_tns, ast_callerid::cid_ton, ast_channel::context, ast_channel::data, ast_datastore::data, DATASTORE_INHERIT_FOREVER, di, dial_exec_options, DIAL_NOFORWARDHTML, DIAL_STILLGOING, dialcontext, dialed_interface_info, do_timelimit(), end_bridge_callback(), ast_bridge_config::end_bridge_callback, ast_bridge_config::end_bridge_callback_data, end_bridge_callback_data_fixup(), ast_bridge_config::end_bridge_callback_data_fixup, ast_bridge_config::end_sound, errno, ast_channel::exten, ast_bridge_config::features_callee, ast_bridge_config::features_caller, ast_flags64::flags, get_cid_name(), handle_cause(), ast_channel::hangupcause, hanguptree(), ast_datastore::inheritance, ast_dialed_interface::interface, LOG_ERROR, LOG_WARNING, ast_channel::macrocontext, ast_channel::macroexten, oprmode::mode, moh, musicclass, ast_channel::nativeformats, chanlist::next, ast_pbx_args::no_hangup_chan, OPT_ANNOUNCE, OPT_ARG_ANNOUNCE, OPT_ARG_ARRAY_SIZE, OPT_ARG_CALLEE_GO_ON, OPT_ARG_CALLEE_GOSUB, OPT_ARG_CALLEE_MACRO, OPT_ARG_DURATION_LIMIT, OPT_ARG_DURATION_STOP, OPT_ARG_GOTO, OPT_ARG_OPERMODE, OPT_ARG_PRIVACY, OPT_ARG_SENDDTMF, OPT_CALLEE_GO_ON, OPT_CALLEE_GOSUB, OPT_CALLEE_HANGUP, OPT_CALLEE_MACRO, OPT_CALLEE_MIXMONITOR, OPT_CALLEE_MONITOR, OPT_CALLEE_PARK, OPT_CALLEE_TRANSFER, OPT_CALLER_HANGUP, OPT_CALLER_MIXMONITOR, OPT_CALLER_MONITOR, OPT_CALLER_PARK, OPT_CALLER_TRANSFER, OPT_CANCEL_ELSEWHERE, OPT_DTMF_EXIT, OPT_DURATION_LIMIT, OPT_DURATION_STOP, OPT_FORCECLID, OPT_GO_ON, OPT_GOTO, OPT_IGNORE_FORWARDING, OPT_MUSICBACK, OPT_OPERMODE, OPT_ORIGINAL_CLID, OPT_PEER_H, OPT_PRIVACY, OPT_RESETCDR, OPT_RINGBACK, OPT_SCREENING, OPT_SENDDTMF, parse(), pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), pbx_exec(), pbx_findapp(), oprmode::peer, ast_channel::priority, privacy_args::privdb_val, replace_macro_delimiter(), S_OR, S_REPLACE, senddialendevent(), senddialevent(), privacy_args::sentringing, setup_privacy_args(), ast_bridge_config::start_sound, privacy_args::status, strsep(), ast_channel::tech, ast_channel::transfercapability, ast_channel_tech::type, url, wait_for_answer(), ast_bridge_config::warning_sound, and ast_channel::whentohangup.

Referenced by dial_exec(), and retrydial_exec().

{
   int res = -1; /* default: error */
   char *rest, *cur; /* scan the list of destinations */
   struct chanlist *outgoing = NULL; /* list of destinations */
   struct ast_channel *peer;
   int to; /* timeout */
   struct cause_args num = { chan, 0, 0, 0 };
   int cause;
   char numsubst[256];
   char cidname[AST_MAX_EXTENSION] = "";

   struct ast_bridge_config config = { { 0, } };
   struct timeval calldurationlimit = { 0, };
   char *dtmfcalled = NULL, *dtmfcalling = 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;
   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];
   struct ast_datastore *datastore = NULL;
   int fulldial = 0, num_dialed = 0;

   /* 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", "");

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

   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])) {
      dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
      dtmfcalled = strsep(&dtmfcalling, ":");
   }

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

   if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
      ast_cdr_reset(chan->cdr, NULL);
   if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
      opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);

   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 (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
      __ast_answer(chan, 0, 0);
   }

   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);  
   ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING);

   /* loop through the list of dial destinations */
   rest = args.peers;
   while ((cur = strsep(&rest, "&")) ) {
      struct chanlist *tmp;
      struct ast_channel *tc; /* channel for this destination */
      /* 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;
      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_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);
         ast_set2_flag64(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_datastore_alloc(&dialed_interface_info, NULL))) {
            ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
            ast_free(tmp);
            goto out;
         }

         datastore->inheritance = DATASTORE_INHERIT_FOREVER;

         if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
            ast_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++;
         ast_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);
            ast_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);
      }

      tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
      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",
            tech, cause, ast_cause2str(cause));
         handle_cause(cause, &num);
         if (!rest) /* we are on the last destination */
            chan->hangupcause = cause;
         ast_free(tmp);
         continue;
      }
      pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);

      /* Setup outgoing SDP to match incoming one */
      ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
      
      /* Inherit specially named variables from parent channel */
      ast_channel_inherit_variables(chan, tc);
      ast_channel_datastore_inherit(chan, tc);

      tc->appl = "AppDial";
      tc->data = "(Outgoing Line)";
      memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));

      S_REPLACE(tc->cid.cid_num, ast_strdup(chan->cid.cid_num));
      S_REPLACE(tc->cid.cid_name, ast_strdup(chan->cid.cid_name));
      S_REPLACE(tc->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
      S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
      
      ast_string_field_set(tc, accountcode, chan->accountcode);
      tc->cdrflags = chan->cdrflags;
      if (ast_strlen_zero(tc->musicclass))
         ast_string_field_set(tc, musicclass, chan->musicclass);
      /* Pass callingpres, type of number, tns, ADSI CPE, transfer capability */
      tc->cid.cid_pres = chan->cid.cid_pres;
      tc->cid.cid_ton = chan->cid.cid_ton;
      tc->cid.cid_tns = chan->cid.cid_tns;
      tc->cid.cid_ani2 = chan->cid.cid_ani2;
      tc->adsicpe = chan->adsicpe;
      tc->transfercapability = chan->transfercapability;

      /* If we have an outbound group, set this peer channel to it */
      if (outbound_group)
         ast_app_group_set_channel(tc, outbound_group);

      /* Inherit context and extension */
      ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
      if (!ast_strlen_zero(chan->macroexten))
         ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
      else
         ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));

      res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */

      /* Save the info in cdr's that we called them */
      if (chan->cdr)
         ast_cdr_setdestchan(chan->cdr, tc->name);

      /* 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", numsubst);
         if (tc->hangupcause) {
            chan->hangupcause = tc->hangupcause;
         }
         ast_hangup(tc);
         tc = NULL;
         ast_free(tmp);
         continue;
      } else {
         senddialevent(chan, tc, numsubst);
         ast_verb(3, "Called %s\n", numsubst);
         if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID))
            ast_set_callerid(tc, 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_flag64(tmp, DIAL_STILLGOING);
      tmp->chan = tc;
      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'. Setting timeout to infinite\n", args.timeout);
         to = -1;
      }
   }

   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(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_flag64(outgoing, OPT_RINGBACK)) {
         ast_indicate(chan, AST_CONTROL_RINGING);
         sentringing++;
      }
   }

   peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &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_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;
      }

      /* SIP, in particular, sends back this error code to indicate an
       * overlap dialled number needs more digits. */
      if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
         res = AST_PBX_INCOMPLETE;
      }

      /* almost done, although the 'else' block is 400 lines */
   } else {
      const char *number;

      strcpy(pa.status, "ANSWER");
      pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.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, 1);
      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);
      
      ast_channel_lock(peer);
      number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER"); 
      if (!number)
         number = numsubst;
      pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
      ast_channel_unlock(peer);

      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)) {
            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;
         /* 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, 0);
         else
            res = digit;

      }

      if (chan && peer && ast_test_flag64(&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, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
         if (continue_exec)
            *continue_exec = 1;
         res = 0;
         goto done;
      }

      if (ast_test_flag64(&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 ? */
            /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
            ast_copy_string(peer->context, chan->context, sizeof(peer->context));
            ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));

            replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
            res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
            ast_debug(1, "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) {
            res = -1;
         }

         ast_channel_lock(peer);

         if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
            char *macro_transfer_dest;

            if (!strcasecmp(macro_result, "BUSY")) {
               ast_copy_string(pa.status, macro_result, sizeof(pa.status));
               ast_set_flag64(peerflags, OPT_GO_ON);
               res = -1;
            } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
               ast_copy_string(pa.status, macro_result, sizeof(pa.status));
               ast_set_flag64(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_flag64(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_flag64(peerflags, OPT_GO_ON);
               }
            }
         }

         ast_channel_unlock(peer);
      }

      if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
         struct ast_app *theapp;
         const char *gosub_result;
         char *gosub_args, *gosub_argstart;
         int res9 = -1;

         res9 = ast_autoservice_start(chan);
         if (res9) {
            ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
            res9 = -1;
         }

         theapp = pbx_findapp("Gosub");

         if (theapp && !res9) {
            replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);

            /* Set where we came from */
            ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
            ast_copy_string(peer->exten, "s", sizeof(peer->exten));
            peer->priority = 0;

            gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
            if (gosub_argstart) {
               *gosub_argstart = 0;
               if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
                  ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
                  gosub_args = NULL;
               }
               *gosub_argstart = ',';
            } else {
               if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
                  ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
                  gosub_args = NULL;
               }
            }

            if (gosub_args) {
               res9 = pbx_exec(peer, theapp, gosub_args);
               if (!res9) {
                  struct ast_pbx_args args;
                  /* A struct initializer fails to compile for this case ... */
                  memset(&args, 0, sizeof(args));
                  args.no_hangup_chan = 1;
                  ast_pbx_run_args(peer, &args);
               }
               ast_free(gosub_args);
               ast_debug(1, "Gosub exited with status %d\n", res9);
            } else {
               ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
            }

         } else if (!res9) {
            ast_log(LOG_ERROR, "Could not find application Gosub\n");
            res9 = -1;
         }

         if (ast_autoservice_stop(chan) < 0) {
            ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
            res9 = -1;
         }
         
         ast_channel_lock(peer);

         if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
            char *gosub_transfer_dest;

            if (!strcasecmp(gosub_result, "BUSY")) {
               ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
               ast_set_flag64(peerflags, OPT_GO_ON);
               res9 = -1;
            } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
               ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
               ast_set_flag64(peerflags, OPT_GO_ON);
               res9 = -1;
            } else if (!strcasecmp(gosub_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_flag64(peerflags, OPT_GO_ON);
               res9 = -1;
            } else if (!strcasecmp(gosub_result, "ABORT")) {
               /* Hangup both ends unless the caller has the g flag */
               res9 = -1;
            } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
               res9 = -1;
               /* perform a transfer to a new extension */
               if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
                  replace_macro_delimiter(gosub_transfer_dest);
                  if (!ast_parseable_goto(chan, gosub_transfer_dest))
                     ast_set_flag64(peerflags, OPT_GO_ON);
               }
            }
         }

         ast_channel_unlock(peer);  
      }

      if (!res) {
         if (!ast_tvzero(calldurationlimit)) {
            struct timeval whentohangup = calldurationlimit;
            peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
         }
         if (!ast_strlen_zero(dtmfcalled)) {
            ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
            res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
         }
         if (!ast_strlen_zero(dtmfcalling)) {
            ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
            res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
         }
      }

      if (res) { /* some error */
         res = -1;
      } else {
         if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
            ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
         if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
            ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
         if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
            ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
         if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
            ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
         if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
            ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
         if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
            ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
         if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
            ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
         if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
            ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
         if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
            ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
         if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
            ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
         if (ast_test_flag64(peerflags, OPT_GO_ON))
            ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);

         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 && !strncmp(chan->tech->type, "DAHDI", 5) && !strncmp(peer->name, "DAHDI", 5)) {
            /* what's this special handling for dahdi <-> dahdi ?
             * A: dahdi to dahdi calls are natively bridged at the kernel driver
             * level, so we need to ensure that this mode gets propagated
             * all the way down. */
            struct oprmode oprmode;

            oprmode.peer = peer;
            oprmode.mode = opermode;

            ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
         }
         res = ast_bridge_call(chan, peer, &config);
      }

      strcpy(peer->context, chan->context);

      if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
         int autoloopflag;
         int found;
         int res9;
         
         strcpy(peer->exten, "h");
         peer->priority = 1;
         autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
         ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);

         while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
            peer->priority++;

         if (found && res9) {
            /* Something bad happened, or a hangup has been requested. */
            ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
            ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
         }
         ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP);  /* set it back the way it was */
      }
      if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {      
         replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
         ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
         ast_pbx_start(peer);
      } else {
         if (!ast_check_hangup(chan))
            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_channel_early_bridge(chan, NULL);
   hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
   pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
   senddialendevent(chan, pa.status);
   ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
   
   if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
      if (!ast_tvzero(calldurationlimit))
         memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
      res = 0;
   }

done:
   if (config.warning_sound) {
      ast_free((char *)config.warning_sound);
   }
   if (config.end_sound) {
      ast_free((char *)config.end_sound);
   }
   if (config.start_sound) {
      ast_free((char *)config.start_sound);
   }
   return res;
}

static void do_forward	(	struct chanlist *	o,
		struct cause_args *	num,
		struct ast_flags64 *	peerflags,
		int	single
	)			[static]

helper function for wait_for_answer()

XXX this code is highly suspicious, as it essentially overwrites the outgoing channel without properly deleting it.

Definition at line 491 of file app_dial.c.

References accountcode, ast_call(), AST_CAUSE_BUSY, ast_channel_datastore_inherit(), ast_channel_inherit_variables(), ast_channel_lock, ast_channel_make_compatible(), ast_channel_unlock, ast_clear_flag64, ast_copy_string(), ast_hangup(), ast_indicate(), ast_log(), AST_MAX_EXTENSION, ast_request(), ast_rtp_make_compatible(), ast_set_callerid(), ast_strdup, ast_string_field_set, ast_strlen_zero(), ast_test_flag64, ast_verb, ast_channel::cdrflags, cause_args::chan, chanlist::chan, ast_channel::cid, ast_callerid::cid_ani, ast_callerid::cid_name, ast_callerid::cid_num, ast_callerid::cid_rdnis, ast_channel::context, DIAL_STILLGOING, ast_channel::exten, get_cid_name(), handle_cause(), LOG_NOTICE, ast_channel::macroexten, ast_channel::nativeformats, cause_args::nochan, OPT_FORCECLID, OPT_IGNORE_FORWARDING, OPT_ORIGINAL_CLID, pbx_builtin_getvar_helper(), S_OR, S_REPLACE, senddialevent(), and ast_channel::tech.

Referenced by wait_for_answer().

{
   char tmpchan[256];
   struct ast_channel *original = o->chan;
   struct ast_channel *c = o->chan; /* the winner */
   struct ast_channel *in = num->chan; /* the input channel */
   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;
      ast_channel_lock(c);
      forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
      if (ast_strlen_zero(forward_context)) {
         forward_context = NULL;
      }
      snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
      ast_channel_unlock(c);
      stuff = tmpchan;
      tech = "Local";
   }
   /* Before processing channel, go ahead and check for forwarding */
   ast_verb(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_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
      ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
      c = o->chan = NULL;
      cause = AST_CAUSE_BUSY;
   } else {
      /* Setup parameters */
      c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
      if (c) {
         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_flag64(o, DIAL_STILLGOING);
      handle_cause(cause, num);
      ast_hangup(original);
   } else {
      char *new_cid_num, *new_cid_name;
      struct ast_channel *src;

      ast_rtp_make_compatible(c, in, single);
      if (ast_test_flag64(o, OPT_FORCECLID)) {
         new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
         new_cid_name = NULL; /* XXX no name ? */
         src = c; /* XXX possible bug in previous code, which used 'winner' ? it may have changed */
      } else {
         new_cid_num = ast_strdup(in->cid.cid_num);
         new_cid_name = ast_strdup(in->cid.cid_name);
         src = in;
      }
      ast_string_field_set(c, accountcode, src->accountcode);
      c->cdrflags = src->cdrflags;
      S_REPLACE(c->cid.cid_num, new_cid_num);
      S_REPLACE(c->cid.cid_name, new_cid_name);

      if (in->cid.cid_ani) { /* XXX or maybe unconditional ? */
         S_REPLACE(c->cid.cid_ani, ast_strdup(in->cid.cid_ani));
      }
      S_REPLACE(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_flag64(o, DIAL_STILLGOING);
         ast_hangup(original);
         ast_hangup(c);
         c = o->chan = NULL;
         num->nochan++;
      } else {
         senddialevent(in, c, stuff);
         /* After calling, set callerid to extension */
         if (!ast_test_flag64(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(original);
      }
      if (single) {
         ast_indicate(in, -1);
      }
   }
}

static int do_timelimit	(	struct ast_channel *	chan,
		struct ast_bridge_config *	config,
		char *	parse,
		struct timeval *	calldurationlimit
	)			[static]

Definition at line 938 of file app_dial.c.

References ast_channel_lock, ast_channel_unlock, AST_FEATURE_PLAY_WARNING, ast_log(), ast_set_flag, ast_strdup, ast_strdupa, ast_strlen_zero(), ast_true(), ast_verb, ast_bridge_config::end_sound, ast_bridge_config::features_callee, ast_bridge_config::features_caller, LOG_WARNING, pbx_builtin_getvar_helper(), ast_bridge_config::play_warning, S_OR, ast_bridge_config::start_sound, strsep(), ast_bridge_config::timelimit, var, ast_bridge_config::warning_freq, and ast_bridge_config::warning_sound.

Referenced by dial_exec_full().

{
   char *stringp = ast_strdupa(parse);
   char *limit_str, *warning_str, *warnfreq_str;
   const char *var;
   int play_to_caller = 0, play_to_callee = 0;
   int delta;

   limit_str = strsep(&stringp, ":");
   warning_str = strsep(&stringp, ":");
   warnfreq_str = strsep(&stringp, ":");

   config->timelimit = atol(limit_str);
   if (warning_str)
      config->play_warning = atol(warning_str);
   if (warnfreq_str)
      config->warning_freq = atol(warnfreq_str);

   if (!config->timelimit) {
      ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
      config->timelimit = config->play_warning = config->warning_freq = 0;
      config->warning_sound = NULL;
      return -1; /* error */
   } else if ( (delta = config->play_warning - config->timelimit) > 0) {
      int w = config->warning_freq;

      /* 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.
         Graphically:
              timelim->|    delta        |<-playwarning
         0__________________|_________________|
                | w  |    |    |    |

         so the number of intervals to cut is 1+(delta-1)/w
      */

      if (w == 0) {
         config->play_warning = 0;
      } else {
         config->play_warning -= w * ( 1 + (delta-1)/w );
         if (config->play_warning < 1)
            config->play_warning = config->warning_freq = 0;
      }
   }
   
   ast_channel_lock(chan);

   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");
   config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");

   /* The code looking at config wants a NULL, not just "", to decide
    * that the message should not be played, so we replace "" with NULL.
    * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
    * not found.
    */

   var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
   config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;

   var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
   config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;

   ast_channel_unlock(chan);

   /* undo effect of S(x) in case they are both used */
   calldurationlimit->tv_sec = 0;
   calldurationlimit->tv_usec = 0;

   /* more efficient to do it like S(x) does since no advanced opts */
   if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
      calldurationlimit->tv_sec = config->timelimit / 1000;
      calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
      ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
         calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
      config->timelimit = play_to_caller = play_to_callee =
      config->play_warning = config->warning_freq = 0;
   } else {
      ast_verb(3, "Limit Data for this call:\n");
      ast_verb(4, "timelimit      = %ld\n", config->timelimit);
      ast_verb(4, "play_warning   = %ld\n", config->play_warning);
      ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
      ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
      ast_verb(4, "warning_freq   = %ld\n", config->warning_freq);
      ast_verb(4, "start_sound    = %s\n", S_OR(config->start_sound, ""));
      ast_verb(4, "warning_sound  = %s\n", config->warning_sound);
      ast_verb(4, "end_sound      = %s\n", S_OR(config->end_sound, ""));
   }
   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);
   return 0;
}

static void end_bridge_callback

(

void *

data

)

[static]

Definition at line 1281 of file app_dial.c.

References ast_cdr::answer, ast_channel_lock, ast_channel_unlock, buf, ast_channel::cdr, pbx_builtin_setvar_helper(), and ast_cdr::start.

Referenced by dial_exec_full().

{
   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
	)			[static]

Definition at line 1306 of file app_dial.c.

References ast_bridge_config::end_bridge_callback_data.

Referenced by dial_exec_full().

                                                                                                                                              {
   bconfig->end_bridge_callback_data = originator;
}

static const char* get_cid_name	(	char *	name,
		int	namelen,
		struct ast_channel *	chan
	)			[static]

Definition at line 451 of file app_dial.c.

References ast_get_hint(), ast_channel::context, context, ast_channel::exten, exten, ast_channel::macrocontext, ast_channel::macroexten, and S_OR.

Referenced by dial_exec_full(), and do_forward().

{
   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 handle_cause	(	int	cause,
		struct cause_args *	num
	)			[static]

Definition at line 386 of file app_dial.c.

References AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NORMAL_CLEARING, AST_CAUSE_UNREGISTERED, ast_cdr_busy(), ast_cdr_failed(), ast_cdr_noanswer(), cause_args::busy, ast_channel::cdr, cause_args::chan, cause_args::congestion, and cause_args::nochan.

Referenced by dial_exec_full(), do_forward(), and wait_for_answer().

{
   struct ast_cdr *cdr = num->chan->cdr;

   switch(cause) {
   case AST_CAUSE_BUSY:
      if (cdr)
         ast_cdr_busy(cdr);
      num->busy++;
      break;

   case AST_CAUSE_CONGESTION:
      if (cdr)
         ast_cdr_failed(cdr);
      num->congestion++;
      break;

   case AST_CAUSE_NO_ROUTE_DESTINATION:
   case AST_CAUSE_UNREGISTERED:
      if (cdr)
         ast_cdr_failed(cdr);
      num->nochan++;
      break;

   case AST_CAUSE_NO_ANSWER:
      if (cdr) {
         ast_cdr_noanswer(cdr);
      }
      break;
   case AST_CAUSE_NORMAL_CLEARING:
      break;

   default:
      num->nochan++;
      break;
   }
}

static void hanguptree	(	struct chanlist *	outgoing,
		struct ast_channel *	exception,
		int	answered_elsewhere
	)			[static]

Definition at line 357 of file app_dial.c.

References AST_FLAG_ANSWERED_ELSEWHERE, ast_free, ast_hangup(), ast_set_flag, chanlist::chan, and chanlist::next.

Referenced by dial_exec_full().

{
   /* Hang up a tree of stuff */
   struct chanlist *oo;
   while (outgoing) {
      /* Hangup any existing lines we have open */
      if (outgoing->chan && (outgoing->chan != exception)) {
         if (answered_elsewhere)
            ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
         ast_hangup(outgoing->chan);
      }
      oo = outgoing;
      outgoing = outgoing->next;
      ast_free(oo);
   }
}

static int load_module

(

void

)

[static]

Definition at line 2187 of file app_dial.c.

References ast_add_extension2(), ast_context_find_or_create(), ast_free_ptr(), ast_log(), ast_register_application, ast_strdup, dial_exec(), LOG_ERROR, and retrydial_exec().

{
   int res;
   struct ast_context *con;

   con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
   if (!con)
      ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
   else
      ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");

   res = ast_register_application(app, dial_exec, synopsis, descrip);
   res |= ast_register_application(rapp, retrydial_exec, rsynopsis, rdescrip);

   return res;
}

static int onedigit_goto	(	struct ast_channel *	chan,
		const char *	context,
		char	exten,
		int	pri
	)			[static]

Definition at line 432 of file app_dial.c.

References ast_goto_if_exists(), ast_strlen_zero(), ast_channel::context, and ast_channel::macrocontext.

Referenced by retrydial_exec(), and wait_for_answer().

{
   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 void replace_macro_delimiter

(

char *

s

)

[static]

Definition at line 919 of file app_dial.c.

Referenced by dial_exec_full().

{
   for (; *s; s++)
      if (*s == '^')
         *s = ',';
}

static int retrydial_exec	(	struct ast_channel *	chan,
		void *	data
	)			[static]

Definition at line 2063 of file app_dial.c.

References AST_APP_ARG, ast_channel_lock, ast_channel_unlock, AST_DECLARE_APP_ARGS, AST_DIGIT_ANY, ast_fileexists(), AST_FLAG_MOH, ast_log(), ast_moh_start(), ast_moh_stop(), AST_PBX_INCOMPLETE, AST_STANDARD_APP_ARGS, ast_strdupa, ast_streamfile(), ast_strlen_zero(), ast_test_flag, ast_test_flag64, ast_waitfordigit(), ast_waitstream(), context, ast_channel::data, dial_exec_full(), LOG_ERROR, LOG_WARNING, onedigit_goto(), OPT_DTMF_EXIT, parse(), and pbx_builtin_getvar_helper().

Referenced by load_module().

{
   char *parse;
   const char *context = NULL;
   int sleepms = 0, loops = 0, res = -1;
   struct ast_flags64 peerflags = { 0, };
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(announce);
      AST_APP_ARG(sleep);
      AST_APP_ARG(retries);
      AST_APP_ARG(dialdata);
   );

   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
      return -1;
   }

   parse = ast_strdupa(data);
   AST_STANDARD_APP_ARGS(args, parse);

   if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
      sleepms *= 1000;

   if (!ast_strlen_zero(args.retries)) {
      loops = atoi(args.retries);
   }

   if (!args.dialdata) {
      ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
      goto done;
   }

   if (sleepms < 1000)
      sleepms = 10000;

   if (!loops)
      loops = -1; /* run forever */

   ast_channel_lock(chan);
   context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
   context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
   ast_channel_unlock(chan);

   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, args.dialdata, &peerflags, &continue_exec);
      if (continue_exec)
         break;

      if (res == 0) {
         if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
            if (!ast_strlen_zero(args.announce)) {
               if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
                  if (!(res = ast_streamfile(chan, args.announce, chan->language)))
                     ast_waitstream(chan, AST_DIGIT_ANY);
               } else
                  ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
            }
            if (!res && sleepms) {
               if (!ast_test_flag(chan, AST_FLAG_MOH))
                  ast_moh_start(chan, NULL, NULL);
               res = ast_waitfordigit(chan, sleepms);
            }
         } else {
            if (!ast_strlen_zero(args.announce)) {
               if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
                  if (!(res = ast_streamfile(chan, args.announce, chan->language)))
                     res = ast_waitstream(chan, "");
               } else
                  ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
            }
            if (sleepms) {
               if (!ast_test_flag(chan, AST_FLAG_MOH))
                  ast_moh_start(chan, NULL, NULL);
               if (!res)
                  res = ast_waitfordigit(chan, sleepms);
            }
         }
      }

      if (res < 0 || res == AST_PBX_INCOMPLETE) {
         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:
   return res;
}

static void senddialendevent	(	const struct ast_channel *	src,
		const char *	dialstatus
	)			[static]

Definition at line 475 of file app_dial.c.

References EVENT_FLAG_CALL, and manager_event.

Referenced by dial_exec_full().

{
   manager_event(EVENT_FLAG_CALL, "Dial",
      "SubEvent: End\r\n"
      "Channel: %s\r\n"
      "UniqueID: %s\r\n"
      "DialStatus: %s\r\n",
      src->name, src->uniqueid, dialstatus);
}

static void senddialevent	(	struct ast_channel *	src,
		struct ast_channel *	dst,
		const char *	dialstring
	)			[static]

Definition at line 459 of file app_dial.c.

References ast_channel::cid, ast_callerid::cid_name, ast_callerid::cid_num, EVENT_FLAG_CALL, manager_event, and S_OR.

Referenced by dial_exec_full(), and do_forward().

{
   manager_event(EVENT_FLAG_CALL, "Dial",
      "SubEvent: Begin\r\n"
      "Channel: %s\r\n"
      "Destination: %s\r\n"
      "CallerIDNum: %s\r\n"
      "CallerIDName: %s\r\n"
      "UniqueID: %s\r\n"
      "DestUniqueID: %s\r\n"
      "Dialstring: %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, dialstring ? dialstring : "");
}

static int setup_privacy_args	(	struct privacy_args *	pa,
		struct ast_flags64 *	opts,
		char *	opt_args[],
		struct ast_channel *	chan
	)			[static]

returns 1 if successful, 0 or <0 if the caller should 'goto out'

Definition at line 1180 of file app_dial.c.

References ast_answer(), ast_config_AST_DATA_DIR, ast_copy_string(), ast_dsp_get_threshold_from_settings(), ast_filedelete(), ast_fileexists(), ast_log(), ast_mkdir(), ast_play_and_record(), AST_PRIVACY_ALLOW, ast_privacy_check(), AST_PRIVACY_DENY, AST_PRIVACY_KILL, AST_PRIVACY_TORTURE, AST_PRIVACY_UNKNOWN, ast_shrink_phone_number(), ast_strdupa, ast_streamfile(), ast_strlen_zero(), ast_test_flag64, ast_verb, ast_waitstream(), ast_channel::cid, ast_callerid::cid_num, ast_channel::exten, LOG_NOTICE, LOG_WARNING, OPT_ARG_PRIVACY, OPT_PRIVACY, OPT_SCREEN_NOCLID, privacy_args::privcid, privacy_args::privdb_val, privacy_args::privintro, silencethreshold, privacy_args::status, and THRESHOLD_SILENCE.

Referenced by dial_exec_full().

{
   char callerid[60];
   int res;
   char *l;
   int silencethreshold;

   if (!ast_strlen_zero(chan->cid.cid_num)) {
      l = ast_strdupa(chan->cid.cid_num);
      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(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 == '/')  /* any other chars to be afraid of? */
            *tn2 = '=';
      }
      ast_verb(3, "Privacy-- callerid is empty\n");

      snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, 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_NOCLID)) {
      /* if callerid is set and OPT_SCREEN_NOCLID 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_NOCLID) && 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:"

         */
         silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
         ast_answer(chan);
         res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, 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", chan->language) )
            ast_waitstream(chan, "");
      }
   }
   return 1; /* success */
}

static int unload_module

(

void

)

[static]

Definition at line 2171 of file app_dial.c.

References ast_context_destroy(), ast_context_find(), ast_context_remove_extension2(), and ast_unregister_application().

{
   int res;
   struct ast_context *con;

   res = ast_unregister_application(app);
   res |= ast_unregister_application(rapp);

   if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
      ast_context_remove_extension2(con, "s", 1, NULL, 0);
      ast_context_destroy(con, "app_dial"); /* leave nothing behind */
   }

   return res;
}

static int valid_priv_reply	(	struct ast_flags64 *	opts,
		int	res
	)			[static]

Definition at line 927 of file app_dial.c.

References ast_test_flag64, OPT_PRIVACY, and OPT_SCREENING.

{
   if (res < '1')
      return 0;
   if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
      return 1;
   if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
      return 1;
   return 0;
}

static struct ast_channel* wait_for_answer	(	struct ast_channel *	in,
		struct chanlist *	outgoing,
		int *	to,
		struct ast_flags64 *	peerflags,
		struct privacy_args *	pa,
		const struct cause_args *	num_in,
		int *	result
	)			[static, read]

Definition at line 595 of file app_dial.c.

References ast_channel::_state, ast_cdr::answer, AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NORMAL_CLEARING, AST_CDR_ANSWERED, ast_cdr_noanswer(), ast_channel_early_bridge(), ast_channel_lock, ast_channel_make_compatible(), ast_channel_sendhtml(), ast_channel_unlock, ast_check_hangup(), ast_clear_flag64, AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, AST_CONTROL_SRCUPDATE, AST_CONTROL_UNHOLD, AST_CONTROL_VIDUPDATE, ast_copy_flags64, ast_copy_string(), ast_deactivate_generator(), ast_debug, AST_FRAME_CONTROL, AST_FRAME_DTMF, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IMAGE, AST_FRAME_TEXT, AST_FRAME_VOICE, ast_frfree, ast_hangup(), ast_indicate(), ast_indicate_data(), ast_log(), AST_MAX_WATCHERS, ast_poll_channel_add(), ast_poll_channel_del(), ast_read(), AST_STATE_UP, ast_str_alloca, ast_string_field_set, ast_strlen_zero(), ast_test_flag64, ast_tvnow(), ast_verb, ast_waitfor_n(), ast_write(), cause_args::busy, CAN_EARLY_BRIDGE, ast_channel::cdr, chanlist::chan, cause_args::congestion, context, ast_frame::data, ast_frame::datalen, detect_disconnect(), DIAL_NOFORWARDHTML, DIAL_STILLGOING, dialcontext, ast_cdr::disposition, do_forward(), ast_channel::exten, f, FEATURE_MAX_LEN, ast_frame::frametype, handle_cause(), ast_channel::hangupcause, LOG_WARNING, chanlist::next, cause_args::nochan, onedigit_goto(), OPT_CALLEE_HANGUP, OPT_CALLEE_MIXMONITOR, OPT_CALLEE_MONITOR, OPT_CALLEE_PARK, OPT_CALLEE_TRANSFER, OPT_CALLER_HANGUP, OPT_CALLER_MIXMONITOR, OPT_CALLER_MONITOR, OPT_CALLER_PARK, OPT_CALLER_TRANSFER, OPT_DTMF_EXIT, OPT_MUSICBACK, OPT_RINGBACK, pbx_builtin_getvar_helper(), ast_frame::ptr, privacy_args::sentringing, privacy_args::status, ast_frame::subclass, and ast_frame::uint32.

Referenced by dial_exec_full().

{
   struct cause_args num = *num_in;
   int prestart = num.busy + num.congestion + num.nochan;
   int orig = *to;
   struct ast_channel *peer = NULL;
   /* single is set if only one destination is enabled */
   int single = outgoing && !outgoing->next && !ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
#ifdef HAVE_EPOLL
   struct chanlist *epollo;
#endif
   struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
   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);
   }

#ifdef HAVE_EPOLL
   for (epollo = outgoing; epollo; epollo = epollo->next)
      ast_poll_channel_add(in, epollo->chan);
#endif

   while (*to && !peer) {
      struct chanlist *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_flag64(o, DIAL_STILLGOING) && o->chan)
            watchers[pos++] = o->chan;
         numlines++;
      }
      if (pos == 1) { /* only the input channel is available */
         if (numlines == (num.busy + num.congestion + num.nochan)) {
            ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
            if (num.busy)
               strcpy(pa->status, "BUSY");
            else if (num.congestion)
               strcpy(pa->status, "CONGESTION");
            else if (num.nochan)
               strcpy(pa->status, "CHANUNAVAIL");
         } else {
            ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
         }
         *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_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
            if (!peer) {
               ast_verb(3, "%s answered %s\n", c->name, in->name);
               peer = c;
               ast_copy_flags64(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 |
                  OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
                  DIAL_NOFORWARDHTML);
               ast_string_field_set(c, dialcontext, "");
               ast_copy_string(c->exten, "", sizeof(c->exten));
            }
            continue;
         }
         if (c != winner)
            continue;
         /* here, o->chan == c == winner */
         if (!ast_strlen_zero(c->call_forward)) {
            do_forward(o, &num, peerflags, single);
            continue;
         }
         f = ast_read(winner);
         if (!f) {
            in->hangupcause = c->hangupcause;
#ifdef HAVE_EPOLL
            ast_poll_channel_del(in, c);
#endif
            ast_hangup(c);
            c = o->chan = NULL;
            ast_clear_flag64(o, DIAL_STILLGOING);
            handle_cause(in->hangupcause, &num);
            continue;
         }
         if (f->frametype == AST_FRAME_CONTROL) {
            switch(f->subclass) {
            case AST_CONTROL_ANSWER:
               /* This is our guy if someone answered. */
               if (!peer) {
                  ast_verb(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_flags64(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 |
                     OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
                     DIAL_NOFORWARDHTML);
                  ast_string_field_set(c, dialcontext, "");
                  ast_copy_string(c->exten, "", sizeof(c->exten));
                  if (CAN_EARLY_BRIDGE(peerflags, in, peer))
                     /* Setup early bridge if appropriate */
                     ast_channel_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:
               ast_verb(3, "%s is busy\n", c->name);
               in->hangupcause = c->hangupcause;
               ast_hangup(c);
               c = o->chan = NULL;
               ast_clear_flag64(o, DIAL_STILLGOING);
               handle_cause(AST_CAUSE_BUSY, &num);
               break;
            case AST_CONTROL_CONGESTION:
               ast_verb(3, "%s is circuit-busy\n", c->name);
               in->hangupcause = c->hangupcause;
               ast_hangup(c);
               c = o->chan = NULL;
               ast_clear_flag64(o, DIAL_STILLGOING);
               handle_cause(AST_CAUSE_CONGESTION, &num);
               break;
            case AST_CONTROL_RINGING:
               ast_verb(3, "%s is ringing\n", c->name);
               /* Setup early media if appropriate */
               if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
                  ast_channel_early_bridge(in, c);
               if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
                  ast_indicate(in, AST_CONTROL_RINGING);
                  pa->sentringing++;
               }
               break;
            case AST_CONTROL_PROGRESS:
               ast_verb(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_channel_early_bridge(in, c);
               if (!ast_test_flag64(outgoing, OPT_RINGBACK))
                  ast_indicate(in, AST_CONTROL_PROGRESS);
               break;
            case AST_CONTROL_VIDUPDATE:
               ast_verb(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:
               ast_verb(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:
               ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
               if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
                  ast_channel_early_bridge(in, c);
               if (!ast_test_flag64(outgoing, OPT_RINGBACK))
                  ast_indicate(in, AST_CONTROL_PROCEEDING);
               break;
            case AST_CONTROL_HOLD:
               ast_verb(3, "Call on %s placed on hold\n", c->name);
               ast_indicate(in, AST_CONTROL_HOLD);
               break;
            case AST_CONTROL_UNHOLD:
               ast_verb(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_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
                  ast_verb(3, "%s stopped sounds\n", c->name);
                  ast_indicate(in, -1);
                  pa->sentringing = 0;
               }
               break;
            default:
               ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
            }
         } else if (single) {
            switch (f->frametype) {
               case AST_FRAME_VOICE:
               case AST_FRAME_IMAGE:
               case AST_FRAME_TEXT:
                  if (ast_write(in, f)) {
                     ast_log(LOG_WARNING, "Unable to write frame\n");
                  }
                  break;
               case AST_FRAME_HTML:
                  if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass, f->data.ptr, f->datalen) == -1) {
                     ast_log(LOG_WARNING, "Unable to send URL\n");
                  }
                  break;
               default:
                  break;
            }
         }
         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;
            strcpy(pa->status, "CANCEL");
            ast_cdr_noanswer(in->cdr);
            if (f) {
               if (f->data.uint32) {
                  in->hangupcause = f->data.uint32;
               }
               ast_frfree(f);
            }
            return NULL;
         }

         /* now f is guaranteed non-NULL */
         if (f->frametype == AST_FRAME_DTMF) {
            if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
               const char *context;
               ast_channel_lock(in);
               context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
               if (onedigit_goto(in, context, (char) f->subclass, 1)) {
                  ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
                  *to = 0;
                  ast_cdr_noanswer(in->cdr);
                  *result = f->subclass;
                  strcpy(pa->status, "CANCEL");
                  ast_frfree(f);
                  ast_channel_unlock(in);
                  return NULL;
               }
               ast_channel_unlock(in);
            }

            if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
               detect_disconnect(in, f->subclass, featurecode)) {
               ast_verb(3, "User requested call disconnect.\n");
               *to = 0;
               strcpy(pa->status, "CANCEL");
               ast_cdr_noanswer(in->cdr);
               ast_frfree(f);
               return NULL;
            }
         }

         /* Forward HTML stuff */
         if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
            if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data.ptr, 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))) {
            ast_verb(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.ptr, f->datalen);
         }
         ast_frfree(f);
      }
      if (!*to)
         ast_verb(3, "Nobody picked up in %d ms\n", orig);
      if (!*to || ast_check_hangup(in))
         ast_cdr_noanswer(in->cdr);
   }

#ifdef HAVE_EPOLL
   for (epollo = outgoing; epollo; epollo = epollo->next) {
      if (epollo->chan)
         ast_poll_channel_del(in, epollo->chan);
   }
#endif

   return peer;
}

---

Variable Documentation

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Dialing Application" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "a9c98e5d177805051735cb5b0b16b0a0" , .load = load_module, .unload = unload_module, } [static]

Definition at line 2204 of file app_dial.c.

char* app = "Dial" [static]

Definition at line 65 of file app_dial.c.

struct ast_module_info* ast_module_info = &__mod_info [static]

Definition at line 2204 of file app_dial.c.

char* descrip [static]

Definition at line 69 of file app_dial.c.

struct ast_app_option dial_exec_options[128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = ((uint64_t)1 << 33) }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'e' ] = { .flag = ((uint64_t)1 << 34) }, [ 'f' ] = { .flag = OPT_FORCECLID }, [ 'F' ] = { .flag = ((uint64_t)1 << 35) , .arg_index = OPT_ARG_CALLEE_GO_ON + 1 }, [ 'g' ] = { .flag = OPT_GO_ON }, [ 'G' ] = { .flag = OPT_GOTO , .arg_index = OPT_ARG_GOTO + 1 }, [ 'h' ] = { .flag = OPT_CALLEE_HANGUP }, [ 'H' ] = { .flag = OPT_CALLER_HANGUP }, [ 'i' ] = { .flag = OPT_IGNORE_FORWARDING }, [ 'k' ] = { .flag = OPT_CALLEE_PARK }, [ 'K' ] = { .flag = OPT_CALLER_PARK }, [ 'L' ] = { .flag = OPT_DURATION_LIMIT , .arg_index = OPT_ARG_DURATION_LIMIT + 1 }, [ 'm' ] = { .flag = OPT_MUSICBACK , .arg_index = OPT_ARG_MUSICBACK + 1 }, [ 'M' ] = { .flag = OPT_CALLEE_MACRO , .arg_index = OPT_ARG_CALLEE_MACRO + 1 }, [ 'n' ] = { .flag = OPT_SCREEN_NOINTRO }, [ 'N' ] = { .flag = OPT_SCREEN_NOCLID }, [ 'o' ] = { .flag = OPT_ORIGINAL_CLID }, [ 'O' ] = { .flag = OPT_OPERMODE , .arg_index = OPT_ARG_OPERMODE + 1 }, [ 'p' ] = { .flag = OPT_SCREENING }, [ 'P' ] = { .flag = OPT_PRIVACY , .arg_index = OPT_ARG_PRIVACY + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'U' ] = { .flag = OPT_CALLEE_GOSUB , .arg_index = OPT_ARG_CALLEE_GOSUB + 1 }, [ 'w' ] = { .flag = OPT_CALLEE_MONITOR }, [ 'W' ] = { .flag = OPT_CALLER_MONITOR }, [ 'x' ] = { .flag = OPT_CALLEE_MIXMONITOR }, [ 'X' ] = { .flag = OPT_CALLER_MIXMONITOR }, } [static]

Definition at line 339 of file app_dial.c.

Referenced by dial_exec_full().

char* rapp = "RetryDial" [static]

Definition at line 235 of file app_dial.c.

char* rdescrip [static]

Definition at line 237 of file app_dial.c.

char* rsynopsis = "Place a call, retrying on failure allowing optional exit extension." [static]

Definition at line 236 of file app_dial.c.

char* synopsis = "Place a call and connect to the current channel" [static]

Definition at line 67 of file app_dial.c.