[svn-commits] branch rizzo/base r8864 - in /team/rizzo/base: ./
agi/ apps/ cdr/ channels/ c...
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Jan 31 14:20:23 MST 2006
Author: rizzo
Date: Sun Jan 29 08:57:22 2006
New Revision: 8864
URL: http://svn.digium.com/view/asterisk?rev=8864&view=rev
Log:
snapshot of all of my local changes.
The detailed list is very long, but a summary is as follows:
- various code cleanup to compile with -Werror
- massive cleanup to file.c
- show thread command
- fix cli completion
- remove macros in extension match code,
document it and prepare for possible extensions
- remove localized definition of struct localuser in
preparation for putting the usecount support in
the wrapper functions.
- cleanup of loader.c in preparation to loading modules
with RTLD_NOW | RTLD_LOCAL
- cleanup and simplification of the code in chan_sip.c,
including a per-peer register command, modified SIP packet
parsing/building code,, etc.
- massive cleanup of cli-related functions in pbx_config.c
removing redundant code
- improved algorithm in channel.c for ast_waitfor*()
- introduce a 'boost' command in chan_oss.c plus fix
to open the device on answer().
- rewrite of app_dial into app_dial2.c
- rewrite of say.c into say2.c and files in say/
(this might probably go away replaced by a better
implementation of say based on config files)
- extend app_playback to play numbers as well (experimental)
possibly replacing the entire say routines
- better algorithm to select a parking slot in
res_features
- wraparound detection code in include/asterisk/time.h
- experimental code for atomic fetchadd on i386 (very efficient)
and other platforms (lock based, same performance as now);
- a lot of comments everywhere
Added:
team/rizzo/base/apps/app_dial2.c
team/rizzo/base/say/
team/rizzo/base/say/say_cz.c
team/rizzo/base/say/say_da.c
team/rizzo/base/say/say_de.c
team/rizzo/base/say/say_en_GB.c
team/rizzo/base/say/say_es.c
team/rizzo/base/say/say_fr.c
team/rizzo/base/say/say_gr.c
team/rizzo/base/say/say_he.c
team/rizzo/base/say/say_it.c
team/rizzo/base/say/say_nl.c
team/rizzo/base/say/say_no.c
team/rizzo/base/say/say_pl.c
team/rizzo/base/say/say_pt.c
team/rizzo/base/say/say_ru.c
team/rizzo/base/say/say_ru_old.c
team/rizzo/base/say/say_se.c
team/rizzo/base/say/say_tw.c
team/rizzo/base/say2.c
Modified:
team/rizzo/base/Makefile
team/rizzo/base/agi/Makefile
team/rizzo/base/app.c
team/rizzo/base/apps/Makefile
team/rizzo/base/apps/app_directory.c
team/rizzo/base/apps/app_echo.c
team/rizzo/base/apps/app_externalivr.c
team/rizzo/base/apps/app_macro.c
team/rizzo/base/apps/app_meetme.c
team/rizzo/base/apps/app_playback.c
team/rizzo/base/apps/app_queue.c
team/rizzo/base/apps/app_sayunixtime.c
team/rizzo/base/apps/app_sms.c
team/rizzo/base/apps/app_voicemail.c
team/rizzo/base/asterisk.c
team/rizzo/base/autoservice.c
team/rizzo/base/callerid.c
team/rizzo/base/cdr.c
team/rizzo/base/cdr/Makefile
team/rizzo/base/channel.c
team/rizzo/base/channels/chan_agent.c
team/rizzo/base/channels/chan_features.c
team/rizzo/base/channels/chan_iax2.c
team/rizzo/base/channels/chan_local.c
team/rizzo/base/channels/chan_oss.c
team/rizzo/base/channels/chan_sip.c
team/rizzo/base/channels/chan_zap.c
team/rizzo/base/channels/iax2-provision.c
team/rizzo/base/channels/iax2-provision.h
team/rizzo/base/cli.c
team/rizzo/base/codecs/gsm/Makefile
team/rizzo/base/config.c
team/rizzo/base/dnsmgr.c
team/rizzo/base/file.c
team/rizzo/base/formats/format_pcm_alaw.c
team/rizzo/base/funcs/func_strings.c
team/rizzo/base/include/asterisk.h
team/rizzo/base/include/asterisk/cdr.h
team/rizzo/base/include/asterisk/channel.h
team/rizzo/base/include/asterisk/cli.h
team/rizzo/base/include/asterisk/config.h
team/rizzo/base/include/asterisk/lock.h
team/rizzo/base/include/asterisk/module.h
team/rizzo/base/include/asterisk/pbx.h
team/rizzo/base/include/asterisk/rtp.h
team/rizzo/base/include/asterisk/say.h
team/rizzo/base/include/asterisk/time.h
team/rizzo/base/include/asterisk/utils.h
team/rizzo/base/loader.c
team/rizzo/base/manager.c
team/rizzo/base/pbx.c
team/rizzo/base/pbx/pbx_config.c
team/rizzo/base/pbx/pbx_dundi.c
team/rizzo/base/pbx/pbx_loopback.c
team/rizzo/base/res/res_agi.c
team/rizzo/base/res/res_clioriginate.c
team/rizzo/base/res/res_features.c
team/rizzo/base/res/res_monitor.c
team/rizzo/base/rtp.c
team/rizzo/base/utils.c
Modified: team/rizzo/base/Makefile
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/Makefile?rev=8864&r1=8863&r2=8864&view=diff
==============================================================================
--- team/rizzo/base/Makefile (original)
+++ team/rizzo/base/Makefile Sun Jan 29 08:57:22 2006
@@ -13,6 +13,22 @@
.EXPORT_ALL_VARIABLES:
+# User-definable variables used here:
+#
+# CROSS_COMPILE path to the cross-compile tools
+# DESTDIR destination directory
+# WITHOUT_xxx build toggles
+# MAKECMDGOALS set of goals for the makefile (to be completed)
+# OVERWRITE overwrite config files on 'make samples'
+# DEBUG debug compiler flags
+# OPTIMIZE optimize compiler flags
+
+# you can also put initial values for these:
+#
+# CFLAGS compiler flags
+# SOFLAGS linker flags
+#
+
# Create OPTIONS variable
OPTIONS=
# If cross compiling, define these to suit
@@ -37,14 +53,14 @@
MAKETOPLEVEL?=$(MAKELEVEL)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
-######### More GSM codec optimization
-######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
-######### which support MMX instructions. This should be newer pentiums,
-######### ppro's, etc, as well as the AMD K6 and K7.
-#K6OPT = -DK6OPT
-
-#Tell gcc to optimize the code
-OPTIMIZE+=-O6
+ # More GSM codec optimization
+ # Uncomment to enable MMXTM optimizations for x86 architecture CPU's
+ # which support MMX instructions. This should be newer pentiums,
+ # ppro's, etc, as well as the AMD K6 and K7.
+ # K6OPT = -DK6OPT
+
+ # Tell gcc to optimize the code
+ OPTIMIZE?=-O2
else
# Stack backtraces, while useful for debugging, are incompatible with optimizations
ifeq (${OSARCH},Linux)
@@ -53,10 +69,10 @@
endif
#Overwite config files on "make samples"
-OVERWRITE=y
+OVERWRITE?=y
#Include debug and macro symbols in the executables (-g) and profiling info (-pg)
-DEBUG=-g3 #-pg
+DEBUG?=-g3 #-pg
#Set NOCRYPTO to yes if you do not want to have crypto support or
#dependencies
@@ -95,7 +111,7 @@
# For example, make DESTDIR=/tmp/asterisk woud put things in
# /tmp/asterisk/etc/asterisk
# XXX watch out, put no spaces or comments after the value
-DESTDIR?=
+DESTDIR?=/tmp/test_ast
#DESTDIR?=/tmp/asterisk
# Original busydetect routine
@@ -225,11 +241,13 @@
INCLUDE+=-Iinclude -I../include
ASTCFLAGS+=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
ASTCFLAGS+=$(OPTIMIZE)
+ASTCFLAGS+= -Werror -Wunused -Wl,--export-dynamic
ASTOBJ=-o asterisk
ifeq ($(findstring BSD,$(OSARCH)),BSD)
PROC=$(shell uname -m)
ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
+ SOLINK+= -L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
ifneq ($(PROC),ultrasparc)
@@ -252,6 +270,8 @@
BSDVERSION=$(shell make -V OSVERSION -f $(CROSS_COMPILE_TARGET)/usr/share/mk/bsd.port.subdir.mk)
ASTCFLAGS+=$(shell if test $(BSDVERSION) -lt 500016 ; then echo "-D_THREAD_SAFE"; fi)
LIBS+=$(shell if test $(BSDVERSION) -lt 502102 ; then echo "-lc_r"; else echo "-pthread"; fi)
+ # XXX libexecinfo
+ # LIBS+= -L/usr/local/lib -lexecinfo
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp),)
ASTCFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include/spandsp
endif
@@ -338,13 +358,13 @@
ASTCFLAGS+= $(BUSYDETECT)
ASTCFLAGS+= $(OPTIONS)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
-ASTCFLAGS+= -fomit-frame-pointer
+#ASTCFLAGS+= -fomit-frame-pointer
endif
SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
- translate.o file.o say.o pbx.o cli.o md5.o term.o \
+ translate.o file.o say2.o pbx.o cli.o md5.o term.o \
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
cdr.o tdd.o acl.o rtp.o udptl.o manager.o asterisk.o \
dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
@@ -352,6 +372,14 @@
utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \
cryptostub.o
+
+SAY_SRCS= say2.c \
+ say/say_cz.c say/say_da.c say/say_de.c \
+ say/say_en_GB.c say/say_es.c say/say_fr.c \
+ say/say_gr.c say/say_he.c say/say_it.c \
+ say/say_nl.c say/say_no.c say/say_pl.c \
+ say/say_pt.c say/say_ru.c say/say_se.c \
+ say/say_tw.c
ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/sys/poll.h),)
OBJS+= poll.o
@@ -380,7 +408,9 @@
else
#These are used for all but Darwin
ASTLINK=-Wl,-E
- SOLINK=-shared -Xlinker -x
+ # XXX added -Wl,-E to SOLINK for debugging
+ #SOLINK=-shared -Xlinker -x -Wl,-E
+ SOLINK=-shared -Wl,-E
ifeq ($(findstring BSD,$(OSARCH)),BSD)
SOLINK+=-L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
@@ -465,6 +495,14 @@
ifneq ($(wildcard .tags-depend),)
include .tags-depend
endif
+
+say2.o: $(SAY_SRCS)
+
+ast_expr2f.o:
+ $(CC) -c $(CFLAGS:-Werror=) $*.c # XXX gmake confused ?
+
+ast_expr2.o:
+ $(CC) -c $(CFLAGS:-Werror=) $*.c # XXX gmake confused ?
ast_expr2.c:
bison -d --name-prefix=ast_yy ast_expr2.y -o ast_expr2.c
Modified: team/rizzo/base/agi/Makefile
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/agi/Makefile?rev=8864&r1=8863&r2=8864&view=diff
==============================================================================
--- team/rizzo/base/agi/Makefile (original)
+++ team/rizzo/base/agi/Makefile Sun Jan 29 08:57:22 2006
@@ -18,10 +18,6 @@
LIBS=
ifeq ($(OSARCH),SunOS)
LIBS=-lsocket -lnsl ../strcompat.o
-endif
-
-ifeq ($(findstring BSD,${OSARCH}),BSD)
- CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
all: depend $(AGIS)
Modified: team/rizzo/base/app.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/app.c?rev=8864&r1=8863&r2=8864&view=diff
==============================================================================
--- team/rizzo/base/app.c (original)
+++ team/rizzo/base/app.c Sun Jan 29 08:57:22 2006
@@ -1546,14 +1546,13 @@
s = optstr;
while (*s) {
- curarg = *s++ & 0x7f;
+ curarg = *s++ & 0x7f; /* the array (in app.h) has 128 entries */
ast_set_flag(flags, options[curarg].flag);
argloc = options[curarg].arg_index;
if (*s == '(') {
/* Has argument */
arg = ++s;
- while (*s && (*s != ')'))
- s++;
+ s = strchr(s, ')');
if (*s) {
if (argloc)
args[argloc - 1] = arg;
Modified: team/rizzo/base/apps/Makefile
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/Makefile?rev=8864&r1=8863&r2=8864&view=diff
==============================================================================
--- team/rizzo/base/apps/Makefile (original)
+++ team/rizzo/base/apps/Makefile Sun Jan 29 08:57:22 2006
@@ -13,7 +13,7 @@
APPS=app_adsiprog.so app_alarmreceiver.so app_authenticate.so app_cdr.so \
app_chanisavail.so app_chanspy.so app_controlplayback.so app_db.so \
- app_dial.so app_dictate.so app_directed_pickup.so app_directory.so \
+ app_dial2.so app_dictate.so app_directed_pickup.so app_directory.so \
app_disa.so app_dumpchan.so app_echo.so app_exec.so app_externalivr.so \
app_festival.so app_forkcdr.so app_getcpeid.so app_hasnewvoicemail.so \
app_ices.so app_image.so app_lookupblacklist.so app_lookupcidname.so \
@@ -27,6 +27,7 @@
app_waitforring.so app_waitforsilence.so app_while.so app_zapateller.so \
app_morsecode.so
+# app_say_number.so
#
# Obsolete things...
#
@@ -51,7 +52,7 @@
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
-ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]]; then echo "OK" ; fi),)
+ifneq ($(shell if [ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907 ]; then echo "OK" ; fi),)
ifneq (${CURLLIBS},)
APPS+=app_curl.so
endif
Added: team/rizzo/base/apps/app_dial2.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/base/apps/app_dial2.c?rev=8864&view=auto
==============================================================================
--- team/rizzo/base/apps/app_dial2.c (added)
+++ team/rizzo/base/apps/app_dial2.c Sun Jan 29 08:57:22 2006
@@ -1,0 +1,1670 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
+ *
+ * \author Mark Spencer <markster at digium.com>
+ *
+ * \ingroup applications
+ */
+
+#include <stdlib.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <sys/time.h>
+#include <sys/signal.h>
+#include <netinet/in.h>
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 8379 $")
+
+#include "asterisk/lock.h"
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+#include "asterisk/translate.h"
+#include "asterisk/say.h"
+#include "asterisk/config.h"
+#include "asterisk/features.h"
+#include "asterisk/musiconhold.h"
+#include "asterisk/callerid.h"
+#include "asterisk/utils.h"
+#include "asterisk/app.h"
+#include "asterisk/causes.h"
+#include "asterisk/manager.h"
+#include "asterisk/privacy.h"
+#include "asterisk/rtp.h"
+
+static char *tdesc = "Dialing Application";
+
+static char *app = "Dial";
+
+static char *synopsis = "Place a call and connect to the current channel";
+
+static char *descrip =
+" Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):\n"
+"This applicaiton will place calls to one or more specified channels. As soon\n"
+"as one of the requested channels answers, the originating channel will be\n"
+"answered, if it has not already been answered. These two channels will then\n"
+"be active in a bridged call. All other channels that were requested will then\n"
+"be hung up.\n"
+" Unless there is a timeout specified, the Dial application will wait\n"
+"indefinitely until one of the called channels answers, the user hangs up, or\n"
+"if all of the called channels are busy or unavailable. Dialplan executing will\n"
+"continue if no requested channels can be called, or if the timeout expires.\n\n"
+" This application sets the following channel variables upon completion:\n"
+" DIALEDTIME - This is the time from dialing a channel until when it\n"
+" is disconnected.\n"
+" ANSWEREDTIME - This is the amount of time for actual call.\n"
+" DIALSTATUS - This is the status of the call:\n"
+" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
+" DONTCALL | TORTURE\n"
+" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
+"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
+"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
+"wants to send the caller to the 'torture' script.\n"
+" This application will report normal termination if the originating channel\n"
+"hangs up, or if the call is bridged and either of the parties in the bridge\n"
+"ends the call.\n"
+" The optional URL will be sent to the called party if the channel supports it.\n"
+" If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
+"application will be put into that group (as in Set(GROUP()=...).\n\n"
+" Options:\n"
+" A(x) - Play an announcement to the called party, using 'x' as the file.\n"
+" C - Reset the CDR for this call.\n"
+" d - Allow the calling user to dial a 1 digit extension while waiting for\n"
+" a call to be answered. Exit to that extension if it exists in the\n"
+" current context, or the context defined in the EXITCONTEXT variable,\n"
+" if it exists.\n"
+" D([called][:calling]) - Send the specified DTMF strings *after* the called\n"
+" party has answered, but before the call gets bridged. The 'called'\n"
+" DTMF string is sent to the called party, and the 'calling' DTMF\n"
+" string is sent to the calling party. Both parameters can be used\n"
+" alone.\n"
+" f - Force the callerid of the *calling* channel to be set as the\n"
+" extension associated with the channel using a dialplan 'hint'.\n"
+" For example, some PSTNs do not allow CallerID to be set to anything\n"
+" other than the number assigned to the caller.\n"
+" g - Proceed with dialplan execution at the current extension if the\n"
+" destination channel hangs up.\n"
+" G(context^exten^pri) - If the call is answered, transfer both parties to\n"
+" the specified priority. Optionally, an extension, or extension and\n"
+" context may be specified. Otherwise, the current extension is used.\n"
+" h - Allow the called party to hang up by sending the '*' DTMF digit.\n"
+" H - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
+" j - Jump to priority n+101 if all of the requested channels were busy.\n"
+" L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are\n"
+" left. Repeat the warning every 'z' ms. The following special\n"
+" variables can be used with this option:\n"
+" * LIMIT_PLAYAUDIO_CALLER yes|no (default yes)\n"
+" Play sounds to the caller.\n"
+" * LIMIT_PLAYAUDIO_CALLEE yes|no\n"
+" Play sounds to the callee.\n"
+" * LIMIT_TIMEOUT_FILE File to play when time is up.\n"
+" * LIMIT_CONNECT_FILE File to play when call begins.\n"
+" * LIMIT_WARNING_FILE File to play as warning if 'y' is defined.\n"
+" The default is to say the time remaining.\n"
+" m([class]) - Provide hold music to the calling party until a requested\n"
+" channel answers. A specific MusicOnHold class can be\n"
+" specified.\n"
+" M(x[^arg]) - Execute the Macro for the *called* channel before connecting\n"
+" to the calling channel. Arguments can be specified to the Macro\n"
+" using '^' as a delimeter. The Macro can set the variable\n"
+" MACRO_RESULT to specify the following actions after the Macro is\n"
+" finished executing.\n"
+" * ABORT Hangup both legs of the call.\n"
+" * CONGESTION Behave as if line congestion was encountered.\n"
+" * BUSY Behave as if a busy signal was encountered. This will also\n"
+" have the application jump to priority n+101 if the\n"
+" 'j' option is set.\n"
+" * CONTINUE Hangup the called party and allow the calling party\n"
+" to continue dialplan execution at the next priority.\n"
+" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
+" specified priority. Optionally, an extension, or\n"
+" extension and priority can be specified.\n"
+" n - This option is a modifier for the screen/privacy mode. It specifies\n"
+" that no introductions are to be saved in the priv-callerintros\n"
+" directory.\n"
+" N - This option is a modifier for the screen/privacy mode. It specifies\n"
+" that if callerID is present, do not screen the call.\n"
+" o - Specify that the CallerID that was present on the *calling* channel\n"
+" be set as the CallerID on the *called* channel. This was the\n"
+" behavior of Asterisk 1.0 and earlier.\n"
+" p - This option enables screening mode. This is basically Privacy mode\n"
+" without memory.\n"
+" P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if\n"
+" it is provided. The current extension is used if a database\n"
+" family/key is not specified.\n"
+" r - Indicate ringing to the calling party. Pass no audio to the calling\n"
+" party until the called channel has answered.\n"
+" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
+" answered the call.\n"
+" t - Allow the called party to transfer the calling party by sending the\n"
+" DTMF sequence defined in features.conf.\n"
+" T - Allow the calling party to transfer the called party by sending the\n"
+" DTMF sequence defined in features.conf.\n"
+" w - Allow the called party to enable recording of the call by sending\n"
+" the DTMF sequence defined for one-touch recording in features.conf.\n"
+" W - Allow the calling party to enable recording of the call by sending\n"
+" the DTMF sequence defined for one-touch recording in features.conf.\n";
+
+/* RetryDial App by Anthony Minessale II <anthmct at yahoo.com> Jan/2005 */
+static char *rapp = "RetryDial";
+static char *rsynopsis = "Place a call, retrying on failure allowing optional exit extension.";
+static char *rdescrip =
+" RetryDial(announce|sleep|retries|dialargs): This application will attempt to\n"
+"place a call using the normal Dial application. If no channel can be reached,\n"
+"the 'announce' file will be played. Then, it will wait 'sleep' number of\n"
+"seconds before retying the call. After 'retires' number of attempts, the\n"
+"calling channel will continue at the next priority in the dialplan. If the\n"
+"'retries' setting is set to 0, this application will retry endlessly.\n"
+" While waiting to retry a call, a 1 digit extension may be dialed. If that\n"
+"extension exists in either the context defined in ${EXITCONTEXT} or the current\n"
+"one, The call will jump to that extension immediately.\n"
+" The 'dialargs' are specified in the same format that arguments are provided\n"
+"to the Dial application.\n";
+
+enum {
+ OPT_ANNOUNCE = (1 << 0),
+ OPT_RESETCDR = (1 << 1),
+ OPT_DTMF_EXIT = (1 << 2),
+ OPT_SENDDTMF = (1 << 3),
+ OPT_FORCECLID = (1 << 4),
+ OPT_GO_ON = (1 << 5),
+ OPT_CALLEE_HANGUP = (1 << 6),
+ OPT_CALLER_HANGUP = (1 << 7),
+ OPT_PRIORITY_JUMP = (1 << 8),
+ OPT_DURATION_LIMIT = (1 << 9),
+ OPT_MUSICBACK = (1 << 10),
+ OPT_CALLEE_MACRO = (1 << 11),
+ OPT_SCREEN_NOINTRO = (1 << 12),
+ OPT_SCREEN_NOCLID = (1 << 13),
+ OPT_ORIGINAL_CLID = (1 << 14),
+ OPT_SCREENING = (1 << 15),
+ OPT_PRIVACY = (1 << 16),
+ OPT_RINGBACK = (1 << 17),
+ OPT_DURATION_STOP = (1 << 18),
+ OPT_CALLEE_TRANSFER = (1 << 19),
+ OPT_CALLER_TRANSFER = (1 << 20),
+ OPT_CALLEE_MONITOR = (1 << 21),
+ OPT_CALLER_MONITOR = (1 << 22),
+ OPT_GOTO = (1 << 23),
+} dial_exec_option_flags;
+
+#define DIAL_STILLGOING (1 << 30)
+#define DIAL_NOFORWARDHTML (1 << 31)
+
+enum {
+ OPT_ARG_ANNOUNCE = 0,
+ OPT_ARG_SENDDTMF,
+ OPT_ARG_GOTO,
+ OPT_ARG_DURATION_LIMIT,
+ OPT_ARG_MUSICBACK,
+ OPT_ARG_CALLEE_MACRO,
+ OPT_ARG_PRIVACY,
+ OPT_ARG_DURATION_STOP,
+ /* note: this entry _MUST_ be the last one in the enum */
+ OPT_ARG_ARRAY_SIZE,
+} dial_exec_option_args;
+
+AST_APP_OPTIONS(dial_exec_options, {
+ AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
+ AST_APP_OPTION('C', OPT_RESETCDR),
+ AST_APP_OPTION('d', OPT_DTMF_EXIT),
+ AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
+ AST_APP_OPTION('f', OPT_FORCECLID),
+ AST_APP_OPTION('g', OPT_GO_ON),
+ AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
+ AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
+ AST_APP_OPTION('H', OPT_CALLER_HANGUP),
+ AST_APP_OPTION('j', OPT_PRIORITY_JUMP),
+ AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
+ AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
+ AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
+ AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
+ AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
+ AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
+ AST_APP_OPTION('p', OPT_SCREENING),
+ AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
+ AST_APP_OPTION('r', OPT_RINGBACK),
+ AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
+ AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
+ AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
+ AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
+ AST_APP_OPTION('W', OPT_CALLER_MONITOR),
+});
+
+/* We define a custom "local user" structure because we
+ use it not only for keeping track of what is in use but
+ also for keeping track of who we're dialing. */
+
+struct dial_localuser {
+ struct ast_channel *chan;
+ unsigned int flags;
+ int forwards;
+ struct dial_localuser *next;
+};
+
+LOCAL_USER_DECL;
+
+static void hanguptree(struct dial_localuser *outgoing, struct ast_channel *exception)
+{
+ /* Hang up a tree of stuff */
+ struct dial_localuser *oo;
+ while (outgoing) {
+ /* Hangup any existing lines we have open */
+ if (outgoing->chan && (outgoing->chan != exception))
+ ast_hangup(outgoing->chan);
+ oo = outgoing;
+ outgoing=outgoing->next;
+ free(oo);
+ }
+}
+
+#define AST_MAX_FORWARDS 8
+
+#define AST_MAX_WATCHERS 256
+
+/*
+ * argument to handle_cause() and other functions.
+ */
+struct cause_args {
+ struct ast_channel *chan;
+ int busy;
+ int congestion;
+ int nochan;
+};
+
+static void handle_cause(int cause, struct cause_args *num)
+{
+ 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_UNREGISTERED:
+ if (cdr)
+ ast_cdr_failed(cdr);
+ num->nochan++;
+ break;
+ default:
+ num->nochan++;
+ break;
+ }
+}
+
+
+/* return first string if not empty, otherwise second */
+#define S_OR(a, b) (!ast_strlen_zero(a) ? (a) : (b))
+
+/* free the buffer if allocated, and set the pointer to the second arg */
+#define S_REPLACE(s, new) \
+ do { \
+ if (s) \
+ free(s); \
+ s = (new); \
+ } while (0)
+
+
+static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
+{
+ char rexten[2] = { exten, '\0' };
+
+ if (context) {
+ if (!ast_goto_if_exists(chan, context, rexten, pri))
+ return 1;
+ } else {
+ if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
+ return 1;
+ else if (!ast_strlen_zero(chan->macrocontext)) {
+ if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
+ return 1;
+ }
+ }
+ return 0;
+}
+
+
+static char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
+{
+ char *context = S_OR(chan->macrocontext, chan->context);
+ char *exten = S_OR(chan->macroexten, chan->exten);
+ if (ast_get_hint(NULL, 0, name, namelen, chan, context, exten))
+ return name;
+ else
+ return "";
+}
+
+static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
+{
+ manager_event(EVENT_FLAG_CALL, "Dial",
+ "Source: %s\r\n"
+ "Destination: %s\r\n"
+ "CallerID: %s\r\n"
+ "CallerIDName: %s\r\n"
+ "SrcUniqueID: %s\r\n"
+ "DestUniqueID: %s\r\n",
+ src->name, dst->name, src->cid.cid_num ? src->cid.cid_num : "<unknown>",
+ src->cid.cid_name ? src->cid.cid_name : "<unknown>", src->uniqueid,
+ dst->uniqueid);
+}
+
+static void do_forward_base(struct dial_localuser *o, int *cause, struct cause_args *num, struct ast_flags *peerflags, char *tmpchan, int tmpchanlen)
+{
+ char *stuff;
+ char *tech;
+ struct ast_channel *winner = o->chan; /* the winner */
+ struct ast_channel *in = num->chan; /* the input channel */
+
+ ast_copy_string(tmpchan, winner->call_forward, tmpchanlen);
+ if ((stuff = strchr(tmpchan, '/'))) {
+ *stuff++ = '\0';
+ tech = tmpchan;
+ } else {
+ snprintf(tmpchan, tmpchanlen, "%s@%s", winner->call_forward, winner->context);
+ stuff = tmpchan;
+ tech = "Local";
+ }
+ /* Before processing channel, go ahead and check for forwarding */
+ o->forwards++;
+ if (o->forwards < AST_MAX_FORWARDS) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, winner->name);
+ /* Setup parameters */
+ o->chan = ast_request(tech, in->nativeformats, stuff, cause);
+ if (!o->chan)
+ ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, *cause);
+ } else {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", o->chan->name);
+ *cause = AST_CAUSE_CONGESTION;
+ o->chan = NULL;
+ }
+}
+
+static void do_forward(struct dial_localuser *o, int *cause, struct cause_args *num, struct ast_flags *peerflags)
+{
+ char tmpchan[256];
+ struct ast_channel *winner = o->chan; /* the winner */
+ struct ast_channel *in = num->chan; /* the input channel */
+
+ do_forward_base(o, cause, num, peerflags, tmpchan, sizeof(tmpchan));
+
+ if (!o->chan) {
+ ast_clear_flag(o, DIAL_STILLGOING);
+ handle_cause(*cause, num);
+ } else {
+ struct ast_channel *c = o->chan;
+ char *new_cid_num;
+ char *new_cid_name;
+ struct ast_channel *src;
+
+ ast_rtp_make_compatible(o->chan, in);
+ if (ast_test_flag(o, OPT_FORCECLID)) {
+ new_cid_num = S_OR(in->macroexten, in->exten);
+ new_cid_name = NULL; /* XXX no name ? */
+ src = winner;
+ } else {
+ new_cid_num = in->cid.cid_num;
+ new_cid_name = in->cid.cid_name;
+ src = in;
+ }
+
+ S_REPLACE(c->cid.cid_num, new_cid_num);
+ S_REPLACE(c->cid.cid_name, new_cid_name);
+ 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)));
+ ast_copy_string(c->accountcode, src->accountcode, sizeof(c->accountcode));
+ c->cdrflags = src->cdrflags;
+
+ if (ast_call(c, tmpchan, 0)) {
+ ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
+ ast_clear_flag(o, DIAL_STILLGOING);
+ ast_hangup(c);
+ o->chan = NULL;
+ num->nochan++;
+ } else {
+ senddialevent(in, c);
+ /* After calling, set callerid to extension */
+ if (!ast_test_flag(peerflags, OPT_ORIGINAL_CLID)) {
+ char cidname[AST_MAX_EXTENSION];
+ ast_set_callerid(c, ast_strlen_zero(in->macroexten) ? in->exten : in->macroexten,
+ get_cid_name(cidname, sizeof(cidname), in), NULL);
+ }
+ }
+ }
+ /* Hangup the original channel now, in case we needed it */
+ ast_hangup(winner);
+}
+
+/* argument used for some functions. */
+struct privacy_args {
+ int sentringing;
+ int privdb_val;
+ char privcid[256];
+ char privintro[1024];
+ char status[256];
+};
+
+static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_localuser *outgoing, int *to, struct ast_flags *peerflags,
+ struct privacy_args *pa, const struct cause_args *num_in, int priority_jump, int *result)
+{
+ struct cause_args num = *num_in;
+ int prestart = num.busy + num.congestion + num.nochan;
+ int cause;
+ int orig = *to;
+ struct ast_channel *peer = NULL;
+ struct ast_channel *watchers[AST_MAX_WATCHERS];
+ int single;
+
+ single = (outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK));
+
+ if (single) {
+ /* Turn off hold music, etc */
+ ast_deactivate_generator(in);
+ /* If we are calling a single channel, make them compatible for in-band tone purpose */
+ ast_channel_make_compatible(outgoing->chan, in);
+ }
+
+
+ while (*to && !peer) {
+ struct ast_channel *winner;
+ int pos = 0;
+ int numlines = prestart;
+ struct dial_localuser *o;
+
+ watchers[pos++] = in;
+ for (o = outgoing; o; o = o->next) {
+ /* Keep track of important channels */
+ if (ast_test_flag(o, DIAL_STILLGOING) && o->chan)
+ watchers[pos++] = o->chan;
+ numlines++;
+ }
+ if (pos == 1) { /* only the input channel available */
+ if (numlines == (num.busy + num.congestion + num.nochan)) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "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");
+ if (ast_opt_priority_jumping || priority_jump)
+ ast_goto_if_exists(in, in->context, in->exten, in->priority + 101);
+ } else {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, 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_flag(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
+ if (!peer) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", c->name, in->name);
+ peer = c;
+ ast_copy_flags(peerflags, o,
+ OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
+ OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
+ OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
+ DIAL_NOFORWARDHTML);
+ }
+ continue;
+ }
+ if (c != winner)
+ continue;
+ if (!ast_strlen_zero(c->call_forward)) {
+ do_forward(o, &cause, &num, peerflags);
+ continue;
+ }
+ f = ast_read(winner);
+ if (f == NULL) {
+ in->hangupcause = o->chan->hangupcause;
+ ast_hangup(o->chan);
+ o->chan = NULL;
+ ast_clear_flag(o, DIAL_STILLGOING);
+ continue;
+ }
+ if (f->frametype == AST_FRAME_CONTROL) {
+ switch(f->subclass) {
+ case AST_CONTROL_ANSWER:
+ /* This is our guy if someone answered. */
+ if (!peer) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s answered %s\n", o->chan->name, in->name);
+ peer = o->chan;
+ ast_copy_flags(peerflags, o,
+ OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
+ OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
+ OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
+ DIAL_NOFORWARDHTML);
+ }
+ /* If call has been answered, then the eventual hangup is likely to be normal hangup */
+ in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
+ o->chan->hangupcause = AST_CAUSE_NORMAL_CLEARING;
+ break;
+ case AST_CONTROL_BUSY:
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s is busy\n", o->chan->name);
+ in->hangupcause = o->chan->hangupcause;
+ ast_hangup(o->chan);
+ o->chan = NULL;
+ ast_clear_flag(o, DIAL_STILLGOING);
+ handle_cause(AST_CAUSE_BUSY, &num);
+ break;
+ case AST_CONTROL_CONGESTION:
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s is circuit-busy\n", o->chan->name);
+ in->hangupcause = o->chan->hangupcause;
+ ast_hangup(o->chan);
+ o->chan = NULL;
+ ast_clear_flag(o, DIAL_STILLGOING);
+ handle_cause(AST_CAUSE_CONGESTION, &num);
+ break;
+ case AST_CONTROL_RINGING:
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", o->chan->name);
+ if (!pa->sentringing && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
+ ast_indicate(in, AST_CONTROL_RINGING);
+ pa->sentringing++;
+ }
+ break;
+ case AST_CONTROL_PROGRESS:
+ if (option_verbose > 2)
+ ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", o->chan->name,in->name);
+ if (!ast_test_flag(outgoing, OPT_RINGBACK))
+ ast_indicate(in, AST_CONTROL_PROGRESS);
+ break;
+ case AST_CONTROL_VIDUPDATE:
+ if (option_verbose > 2)
+ ast_verbose (VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
+ ast_indicate(in, AST_CONTROL_VIDUPDATE);
+ break;
+ case AST_CONTROL_PROCEEDING:
+ if (option_verbose > 2)
+ ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
+ if (!ast_test_flag(outgoing, OPT_RINGBACK))
+ ast_indicate(in, AST_CONTROL_PROCEEDING);
+ break;
+ case AST_CONTROL_HOLD:
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Call on %s placed on hold\n", o->chan->name);
+ ast_indicate(in, AST_CONTROL_HOLD);
+ break;
+ case AST_CONTROL_UNHOLD:
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Call on %s left from hold\n", o->chan->name);
+ ast_indicate(in, AST_CONTROL_UNHOLD);
+ break;
+ case AST_CONTROL_OFFHOOK:
+ case AST_CONTROL_FLASH:
+ /* Ignore going off hook and flash */
+ break;
+ case -1:
+ if (!ast_test_flag(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "%s stopped sounds\n", o->chan->name);
+ ast_indicate(in, -1);
+ pa->sentringing = 0;
+ }
+ break;
+ default:
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Dunno what to do with control type %d\n", f->subclass);
+ }
+ } else if (single) {
+ if (f->frametype == AST_FRAME_VOICE &&
+ !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
+ if (ast_write(in, f))
+ ast_log(LOG_DEBUG, "Unable to forward voice frame\n");
+ } else if (f->frametype == AST_FRAME_IMAGE &&
+ !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
+ if (ast_write(in, f))
+ ast_log(LOG_DEBUG, "Unable to forward image\n");
+ } else if (f->frametype == AST_FRAME_TEXT &&
+ !ast_test_flag(outgoing, OPT_RINGBACK|OPT_MUSICBACK)) {
+ if (ast_write(in, f))
+ ast_log(LOG_DEBUG, "Unable to send text\n");
+ } else if (f->frametype == AST_FRAME_HTML && !ast_test_flag(outgoing, DIAL_NOFORWARDHTML))
+ if (ast_channel_sendhtml(in, f->subclass, f->data, f->datalen) == -1)
+ ast_log(LOG_WARNING, "Unable to send URL\n");
+ }
+
+ ast_frfree(f);
+ } /* end for */
+ if (winner == in) {
+ struct ast_frame *f = ast_read(in);
+#if 0
+ if (f && (f->frametype != AST_FRAME_VOICE))
+ printf("Frame type: %d, %d\n", f->frametype, f->subclass);
+ else if (!f || (f->frametype != AST_FRAME_VOICE))
+ printf("Hangup received on %s\n", in->name);
+#endif
+ if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
+ /* Got hung up */
+ *to = -1;
+ strcpy(pa->status, "CANCEL");
+ if (f)
+ ast_frfree(f);
+ return NULL;
+ }
+ /* now f is guaranteed good */
+ if (f->frametype == AST_FRAME_DTMF) {
+ int done = 0;
+ if (ast_test_flag(peerflags, OPT_DTMF_EXIT)) {
+ const char *context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
+ if (onedigit_goto(in, context, (char) f->subclass, 1)) {
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
+ *result = f->subclass;
+ done = 1;
+ }
+ }
[... 25701 lines stripped ...]
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