Version 0.1.0 from FTP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90 f38db490-d61c-443f-a65b-d21fe96a405b

7 files changed, 1051 insertions, 0 deletions

diff --git a/codecs/gsm/Makefile b/codecs/gsm/Makefile
new file mode 100755
index 000000000..4af3788f6
--- /dev/null
+++ b/codecs/gsm/Makefile
@@ -0,0 +1,494 @@
+# Copyright 1992-1996 by Jutta Degener and Carsten Bormann, Technische
+# Universitaet Berlin.  See the accompanying file "COPYRIGHT" for
+# details.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
+
+# Machine- or installation dependent flags you should configure to port
+
+SASR	= -DSASR
+######### Define SASR if >> is a signed arithmetic shift (-1 >> 1 == -1)
+
+MULHACK = -DUSE_FLOAT_MUL
+######### Define this if your host multiplies floats faster than integers,
+######### e.g. on a SPARCstation.
+
+FAST	= -DFAST
+######### Define together with USE_FLOAT_MUL to enable the GSM library's
+######### approximation option for incorrect, but good-enough results.
+
+# LTP_CUT	= -DLTP_CUT
+LTP_CUT	=
+######### Define to enable the GSM library's long-term correlation 
+######### approximation option---faster, but worse; works for
+######### both integer and floating point multiplications.
+######### This flag is still in the experimental stage.
+
+WAV49	= -DWAV49
+#WAV49	=
+######### Define to enable the GSM library's option to pack GSM frames 
+######### in the style used by the WAV #49 format.  If you want to write
+######### a tool that produces .WAV files which contain GSM-encoded data,
+######### define this, and read about the GSM_OPT_WAV49 option in the
+######### manual page on gsm_option(3).
+
+# Choose a compiler.  The code works both with ANSI and K&R-C.
+# Use -DNeedFunctionPrototypes to compile with, -UNeedFunctionPrototypes to
+# compile without, function prototypes in the header files.
+#
+# You can use the -DSTUPID_COMPILER to circumvent some compilers'
+# static limits regarding the number of subexpressions in a statement.
+
+# CC		= cc
+# CCFLAGS 	= -c -DSTUPID_COMPILER
+
+# CC		= /usr/lang/acc
+# CCFLAGS 	= -c -O
+
+CC		= gcc -ansi -pedantic
+CCFLAGS 	+= -c -DNeedFunctionPrototypes=1 -finline-functions -funroll-loops
+
+LD 		= $(CC)
+
+# LD		= gcc
+# LDFLAGS 	=
+
+
+# If your compiler needs additional flags/libraries, regardless of
+# the source compiled, configure them here.
+
+# CCINC	= -I/usr/gnu/lib/gcc-2.1/gcc-lib/sparc-sun-sunos4.1.2/2.1/include
+######### Includes needed by $(CC)
+
+# LDINC	= -L/usr/gnu/lib/gcc-2.1/gcc-lib/sparc-sun-sunos4.1.2/2.1
+######### Library paths needed by $(LD)
+
+# LDLIB	= -lgcc
+######### Additional libraries needed by $(LD)
+
+
+# Where do you want to install libraries, binaries, a header file
+# and the manual pages?
+#
+# Leave INSTALL_ROOT empty (or just don't execute "make install") to
+# not install gsm and toast outside of this directory.
+
+INSTALL_ROOT	=
+
+# Where do you want to install the gsm library, header file, and manpages?
+#
+# Leave GSM_INSTALL_ROOT empty to not install the GSM library outside of
+# this directory.
+
+GSM_INSTALL_ROOT = $(INSTALL_ROOT)
+GSM_INSTALL_LIB = $(GSM_INSTALL_ROOT)/lib
+GSM_INSTALL_INC = $(GSM_INSTALL_ROOT)/inc
+GSM_INSTALL_MAN = $(GSM_INSTALL_ROOT)/man/man3
+
+
+# Where do you want to install the toast binaries and their manpage?
+#
+# Leave TOAST_INSTALL_ROOT empty to not install the toast binaries outside
+# of this directory.
+
+TOAST_INSTALL_ROOT	  = $(INSTALL_ROOT)
+TOAST_INSTALL_BIN = $(TOAST_INSTALL_ROOT)/bin
+TOAST_INSTALL_MAN = $(TOAST_INSTALL_ROOT)/man/man1
+
+#  Other tools
+
+SHELL		= /bin/sh
+LN		= ln
+BASENAME 	= basename
+AR		= ar
+ARFLAGS		= cr
+RMFLAGS		= -f
+FIND		= find
+COMPRESS 	= compress
+COMPRESSFLAGS 	= 
+# RANLIB 	= true
+RANLIB	 	= ranlib
+
+#
+#    You shouldn't have to configure below this line if you're porting.
+# 
+
+
+# Local Directories
+
+ROOT	= .
+ADDTST	= $(ROOT)/add-test
+TST	= $(ROOT)/tst
+MAN	= $(ROOT)/man
+BIN	= $(ROOT)/bin
+SRC	= $(ROOT)/src
+LIB	= $(ROOT)/lib
+TLS	= $(ROOT)/tls
+INC	= $(ROOT)/inc
+
+# Flags
+
+# DEBUG	= -DNDEBUG
+######### Remove -DNDEBUG to enable assertions.
+
+CFLAGS	= $(CCFLAGS) $(SASR) $(DEBUG) $(MULHACK) $(FAST) $(LTP_CUT) \
+	$(WAV49) $(CCINC) -I$(INC)
+######### It's $(CC) $(CFLAGS)
+
+LFLAGS	= $(LDFLAGS) $(LDINC)
+######### It's $(LD) $(LFLAGS)
+
+
+# Targets
+
+LIBGSM	= $(LIB)/libgsm.a
+
+TOAST	= $(BIN)/toast
+UNTOAST	= $(BIN)/untoast
+TCAT	= $(BIN)/tcat
+
+# Headers
+
+GSM_HEADERS =	$(INC)/gsm.h
+
+HEADERS	=	$(INC)/proto.h		\
+		$(INC)/unproto.h	\
+		$(INC)/config.h		\
+		$(INC)/private.h	\
+		$(INC)/gsm.h		\
+		$(INC)/toast.h		\
+		$(TLS)/taste.h
+
+# Sources
+
+GSM_SOURCES =	$(SRC)/add.c		\
+		$(SRC)/code.c		\
+		$(SRC)/debug.c		\
+		$(SRC)/decode.c		\
+		$(SRC)/long_term.c	\
+		$(SRC)/lpc.c		\
+		$(SRC)/preprocess.c	\
+		$(SRC)/rpe.c		\
+		$(SRC)/gsm_destroy.c	\
+		$(SRC)/gsm_decode.c	\
+		$(SRC)/gsm_encode.c	\
+		$(SRC)/gsm_explode.c	\
+		$(SRC)/gsm_implode.c	\
+		$(SRC)/gsm_create.c	\
+		$(SRC)/gsm_print.c	\
+		$(SRC)/gsm_option.c	\
+		$(SRC)/short_term.c	\
+		$(SRC)/table.c
+
+TOAST_SOURCES = $(SRC)/toast.c 		\
+		$(SRC)/toast_lin.c	\
+		$(SRC)/toast_ulaw.c	\
+		$(SRC)/toast_alaw.c	\
+		$(SRC)/toast_audio.c
+
+SOURCES	=	$(GSM_SOURCES)		\
+		$(TOAST_SOURCES)	\
+		$(ADDTST)/add_test.c	\
+		$(TLS)/sour.c		\
+		$(TLS)/ginger.c		\
+		$(TLS)/sour1.dta	\
+		$(TLS)/sour2.dta	\
+		$(TLS)/bitter.c		\
+		$(TLS)/bitter.dta	\
+		$(TLS)/taste.c		\
+		$(TLS)/sweet.c		\
+		$(TST)/cod2lin.c	\
+		$(TST)/cod2txt.c	\
+		$(TST)/gsm2cod.c	\
+		$(TST)/lin2cod.c	\
+		$(TST)/lin2txt.c
+
+# Object files
+
+GSM_OBJECTS =	$(SRC)/add.o		\
+		$(SRC)/code.o		\
+		$(SRC)/debug.o		\
+		$(SRC)/decode.o		\
+		$(SRC)/long_term.o	\
+		$(SRC)/lpc.o		\
+		$(SRC)/preprocess.o	\
+		$(SRC)/rpe.o		\
+		$(SRC)/gsm_destroy.o	\
+		$(SRC)/gsm_decode.o	\
+		$(SRC)/gsm_encode.o	\
+		$(SRC)/gsm_explode.o	\
+		$(SRC)/gsm_implode.o	\
+		$(SRC)/gsm_create.o	\
+		$(SRC)/gsm_print.o	\
+		$(SRC)/gsm_option.o	\
+		$(SRC)/short_term.o	\
+		$(SRC)/table.o
+
+TOAST_OBJECTS =	$(SRC)/toast.o 		\
+		$(SRC)/toast_lin.o	\
+		$(SRC)/toast_ulaw.o	\
+		$(SRC)/toast_alaw.o	\
+		$(SRC)/toast_audio.o
+
+OBJECTS =	 $(GSM_OBJECTS) $(TOAST_OBJECTS)
+
+# Manuals
+
+GSM_MANUALS =	$(MAN)/gsm.3		\
+		$(MAN)/gsm_explode.3	\
+		$(MAN)/gsm_option.3	\
+		$(MAN)/gsm_print.3
+
+TOAST_MANUALS =	$(MAN)/toast.1
+
+MANUALS	= 	$(GSM_MANUALS) $(TOAST_MANUALS) $(MAN)/bitter.1
+
+# Other stuff in the distribution
+
+STUFF = 	ChangeLog			\
+		INSTALL			\
+		MACHINES		\
+		MANIFEST		\
+		Makefile		\
+		README			\
+		$(ADDTST)/add_test.dta	\
+		$(TLS)/bitter.dta	\
+		$(TST)/run
+
+
+# Install targets
+
+GSM_INSTALL_TARGETS =	\
+		$(GSM_INSTALL_LIB)/libgsm.a		\
+		$(GSM_INSTALL_INC)/gsm.h		\
+		$(GSM_INSTALL_MAN)/gsm.3		\
+		$(GSM_INSTALL_MAN)/gsm_explode.3	\
+		$(GSM_INSTALL_MAN)/gsm_option.3		\
+		$(GSM_INSTALL_MAN)/gsm_print.3
+
+TOAST_INSTALL_TARGETS =	\
+		$(TOAST_INSTALL_BIN)/toast		\
+		$(TOAST_INSTALL_BIN)/tcat		\
+		$(TOAST_INSTALL_BIN)/untoast		\
+		$(TOAST_INSTALL_MAN)/toast.1
+
+
+# Default rules
+
+.c.o:
+		$(CC) $(CFLAGS) $?
+		@-mv `$(BASENAME) $@` $@ > /dev/null 2>&1
+
+# Target rules
+
+all:		$(LIBGSM) $(TOAST) $(TCAT) $(UNTOAST)
+		@-echo $(ROOT): Done.
+
+tst:		$(TST)/lin2cod $(TST)/cod2lin $(TOAST) $(TST)/test-result
+		@-echo tst: Done.
+
+addtst:		$(ADDTST)/add $(ADDTST)/add_test.dta
+		$(ADDTST)/add < $(ADDTST)/add_test.dta > /dev/null
+		@-echo addtst: Done.
+
+misc:		$(TLS)/sweet $(TLS)/bitter $(TLS)/sour $(TLS)/ginger 	\
+			$(TST)/lin2txt $(TST)/cod2txt $(TST)/gsm2cod
+		@-echo misc: Done.
+
+install:	toastinstall gsminstall
+		@-echo install: Done.
+
+
+# The basic API: libgsm
+
+$(LIBGSM):	$(LIB) $(GSM_OBJECTS)
+		-rm $(RMFLAGS) $(LIBGSM)
+		$(AR) $(ARFLAGS) $(LIBGSM) $(GSM_OBJECTS)
+		$(RANLIB) $(LIBGSM)
+
+
+# Toast, Untoast and Tcat -- the compress-like frontends to gsm.
+
+$(TOAST):	$(BIN) $(TOAST_OBJECTS) $(LIBGSM)
+		$(LD) $(LFLAGS) -o $(TOAST) $(TOAST_OBJECTS) $(LIBGSM) $(LDLIB)
+
+$(UNTOAST):	$(BIN) $(TOAST)
+		-rm $(RMFLAGS) $(UNTOAST)
+		$(LN) $(TOAST) $(UNTOAST)
+
+$(TCAT):	$(BIN) $(TOAST)
+		-rm $(RMFLAGS) $(TCAT)
+		$(LN) $(TOAST) $(TCAT)
+
+
+# The local bin and lib directories
+
+$(BIN):
+		if [ ! -d $(BIN) ] ; then mkdir $(BIN) ; fi
+
+$(LIB):
+		if [ ! -d $(LIB) ] ; then mkdir $(LIB) ; fi
+
+
+# Installation
+
+gsminstall:
+		-if [ x"$(GSM_INSTALL_ROOT)" != x ] ; then	\
+			make $(GSM_INSTALL_TARGETS) ;	\
+		fi
+
+toastinstall:
+		-if [ x"$(TOAST_INSTALL_ROOT)" != x ]; then	\
+			make $(TOAST_INSTALL_TARGETS);	\
+		fi
+
+gsmuninstall:
+		-if [ x"$(GSM_INSTALL_ROOT)" != x ] ; then	\
+			rm $(RMFLAGS) $(GSM_INSTALL_TARGETS) ;	\
+		fi
+
+toastuninstall:
+		-if [ x"$(TOAST_INSTALL_ROOT)" != x ] ; then 	\
+			rm $(RMFLAGS) $(TOAST_INSTALL_TARGETS);	\
+		fi
+
+$(TOAST_INSTALL_BIN)/toast:	$(TOAST)
+		-rm $@
+		cp $(TOAST) $@
+		chmod 755 $@
+
+$(TOAST_INSTALL_BIN)/untoast:	$(TOAST_INSTALL_BIN)/toast
+		-rm $@
+		ln $? $@
+
+$(TOAST_INSTALL_BIN)/tcat:	$(TOAST_INSTALL_BIN)/toast
+		-rm $@
+		ln $? $@
+
+$(TOAST_INSTALL_MAN)/toast.1:	$(MAN)/toast.1
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_MAN)/gsm.3:	$(MAN)/gsm.3
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_MAN)/gsm_option.3:	$(MAN)/gsm_option.3
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_MAN)/gsm_explode.3:	$(MAN)/gsm_explode.3
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_MAN)/gsm_print.3:	$(MAN)/gsm_print.3
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_INC)/gsm.h:	$(INC)/gsm.h
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+$(GSM_INSTALL_LIB)/libgsm.a:	$(LIBGSM)
+		-rm $@
+		cp $? $@
+		chmod 444 $@
+
+
+# Distribution
+
+dist:		gsm-1.0.tar.Z
+		@echo dist: Done.
+
+gsm-1.0.tar.Z:	$(STUFF) $(SOURCES) $(HEADERS) $(MANUALS)
+		(	cd $(ROOT)/..;				\
+			tar cvf - `cat $(ROOT)/gsm-1.0/MANIFEST	\
+				| sed '/^#/d'`			\
+		) | $(COMPRESS) $(COMPRESSFLAGS) > $(ROOT)/gsm-1.0.tar.Z
+
+# Clean
+
+uninstall:	toastuninstall gsmuninstall
+		@-echo uninstall: Done.
+
+semi-clean:
+		-rm $(RMFLAGS)  */*.o			\
+			$(TST)/lin2cod $(TST)/lin2txt	\
+			$(TST)/cod2lin $(TST)/cod2txt	\
+			$(TST)/gsm2cod 			\
+			$(TST)/*.*.*
+		-$(FIND) . \( -name core -o -name foo \) \
+			-print | xargs rm $(RMFLAGS)
+
+clean:	semi-clean
+		-rm $(RMFLAGS) $(LIBGSM) $(ADDTST)/add		\
+			$(TOAST) $(TCAT) $(UNTOAST)	\
+			$(ROOT)/gsm-1.0.tar.Z
+
+
+# Two tools that helped me generate gsm_encode.c and gsm_decode.c,
+# but aren't generally needed to port this.
+
+$(TLS)/sweet:	$(TLS)/sweet.o $(TLS)/taste.o
+		$(LD) $(LFLAGS) -o $(TLS)/sweet \
+			$(TLS)/sweet.o $(TLS)/taste.o $(LDLIB)
+
+$(TLS)/bitter:	$(TLS)/bitter.o $(TLS)/taste.o
+		$(LD) $(LFLAGS) -o $(TLS)/bitter \
+			$(TLS)/bitter.o $(TLS)/taste.o $(LDLIB)
+
+# A version of the same family that Jeff Chilton used to implement
+# the WAV #49 GSM format.
+
+$(TLS)/ginger:	$(TLS)/ginger.o $(TLS)/taste.o
+		$(LD) $(LFLAGS) -o $(TLS)/ginger \
+			$(TLS)/ginger.o $(TLS)/taste.o $(LDLIB)
+
+$(TLS)/sour:	$(TLS)/sour.o $(TLS)/taste.o
+		$(LD) $(LFLAGS) -o $(TLS)/sour \
+			$(TLS)/sour.o $(TLS)/taste.o $(LDLIB)
+
+# Run $(ADDTST)/add < $(ADDTST)/add_test.dta to make sure the
+# basic arithmetic functions work as intended.
+
+$(ADDTST)/add:	$(ADDTST)/add_test.o
+		$(LD) $(LFLAGS) -o $(ADDTST)/add $(ADDTST)/add_test.o $(LDLIB)
+
+
+# Various conversion programs between linear, text, .gsm and the code
+# format used by the tests we ran (.cod).  We paid for the test data,
+# so I guess we can't just provide them with this package.  Still,
+# if you happen to have them lying around, here's the code.
+# 
+# You can use gsm2cod | cod2txt independently to look at what's
+# coded inside the compressed frames, although this shouldn't be
+# hard to roll on your own using the gsm_print() function from
+# the API.
+
+
+$(TST)/test-result:	$(TST)/lin2cod $(TST)/cod2lin $(TOAST) $(TST)/run
+			( cd $(TST); ./run ) 
+
+$(TST)/lin2txt:		$(TST)/lin2txt.o $(LIBGSM)
+			$(LD) $(LFLAGS) -o $(TST)/lin2txt \
+				$(TST)/lin2txt.o $(LIBGSM) $(LDLIB)
+
+$(TST)/lin2cod:		$(TST)/lin2cod.o $(LIBGSM)
+			$(LD) $(LFLAGS) -o $(TST)/lin2cod \
+				$(TST)/lin2cod.o $(LIBGSM) $(LDLIB)
+
+$(TST)/gsm2cod:		$(TST)/gsm2cod.o $(LIBGSM)
+			$(LD) $(LFLAGS) -o $(TST)/gsm2cod \
+				$(TST)/gsm2cod.o $(LIBGSM) $(LDLIB)
+
+$(TST)/cod2txt:		$(TST)/cod2txt.o $(LIBGSM)
+			$(LD) $(LFLAGS) -o $(TST)/cod2txt \
+				$(TST)/cod2txt.o $(LIBGSM) $(LDLIB)
+
+$(TST)/cod2lin:		$(TST)/cod2lin.o $(LIBGSM)
+			$(LD) $(LFLAGS) -o $(TST)/cod2lin \
+				$(TST)/cod2lin.o $(LIBGSM) $(LDLIB)
diff --git a/configs/adtranvofr.conf.sample b/configs/adtranvofr.conf.sample
new file mode 100755
index 000000000..df19e094e
--- /dev/null
+++ b/configs/adtranvofr.conf.sample
@@ -0,0 +1,37 @@
+;
+; Voice over Frame Relay (Adtran style)
+;
+; Configuration file
+;
+[interfaces]
+;
+; Lines for which we are the user termination.  They accept incoming
+; and outgoing calls.
+;
+;user=voice00
+;user=voice01
+;user=voice02
+;user=voice03
+;user=voice04
+;user=voice05
+;user=voice06
+;user=voice07
+context=default
+user=voice13
+user=voice14
+user=voice15
+; Calls on 16 and 17 come from the outside world, so they get
+; a little bit special treatment
+context=remote
+user=voice16
+user=voice17
+;
+; Next we have lines which we only accept calls on, and typically
+; do not send outgoing calls on (i.e. these are where we are the
+; network termination)
+;
+;network=voice08
+;network=voice09
+;network=voice10
+;network=voice11
+;network=voice12
diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample
new file mode 100755
index 000000000..e654c4597
--- /dev/null
+++ b/configs/extensions.conf.sample
@@ -0,0 +1,118 @@
+;
+; Static extension configuration files, used by
+; the pbx_config module.
+;
+; The "General" category is for certain variables.  All other categories
+; are interpreted as extension contexts
+;
+[general]
+;
+; If static is set to no, or omitted, then the pbx_config will rewrite
+; this file when extensions are modified.  Remember that all comments
+; made in the file will be lost when that happens.
+;
+static=yes
+
+; Remote things always ring all phones first.
+[remote]
+exten=s,1,Dial,AdtranVoFR/4200&AdtranVoFR/4151&AdtranVoFR/4300|15
+exten=s,2,Goto,default|s|2
+
+; Local stuff
+[local]
+exten=s,1,Goto,defaults|s|2
+; Special extension for local phone numbers, long distance, etc, going
+; out via the Frame Relay interface.  Patterns are prefixed with "_", which
+; is ignored.
+exten=_9NXXXXXX,1,Dial,AdtranVoFR/BYEXTENSION
+exten=_91NXXNXXXXXX,1,Dial,AdtranVoFR/BYEXTENSION
+exten=_9911,1,Dial,AdtranVoFR/BYEXTENSION
+
+[default]
+exten=s,1,Wait,0
+exten=s,2,Answer
+exten=s,3,DigitTimeout,5
+exten=s,4,ResponseTimeout,10
+exten=s,5,BackGround,welcome
+exten=*,1,Directory,default
+exten=*,2,Goto,s|4
+exten=#,1,Playback,goodbye
+exten=#,2,Hangup
+exten=100,1,Goto,other|s|1
+exten=200,1,Intercom
+exten=400,1,MP3Player,song8.mp3
+exten=401,1,MP3Player,sample.mp3
+exten=402,1,MP3Player,sunscreen.mp3
+exten=403,1,MP3Player,http://trode.vergenet.net:8000
+exten=404,1,MP3Player,http://216.32.166.94:14900
+exten=405,1,Playback,sample
+;
+; Here's the template for a typical extension, carefully broken apart
+; for analysis.  The others are pretty much the same, but not as well
+; documented.
+;
+; Step 1:  Play back a "Please hold while I try that extension" message
+exten=4300,1,Playback,transfer
+; Step 2:  Dial the numbers where Ben is likely to be.  Try for no more
+;          than 15 seconds.
+exten=4300,2,Dial,AdtranVoFR/4300|15
+; Step 3:  If there is no answer, play back a message stating that Ben is
+;          unavailable.  Alternatively, we could have rung an operator first.
+exten=4300,3,Playback,vm/4300/unavail
+; Step 4:  Send them to voicemail.
+exten=4300,4,Voicemail,4300
+; Step 5:  If they return from voicemail, go back to the top
+exten=4300,5,Goto,s|4
+; Step 103: If the Dialing is busy, it will try here first.  We'll play a
+;           special "I'm busy" message...
+exten=4300,103,Playback,vm/4300/busy
+; Step 104: And then continue as if it had been busy in the first place.
+exten=4300,104,Goto,4
+; Exten. 4301: Provide a short-circuit so we can transfer striaght to 
+;              voicemail.
+exten=4301,1,Goto,4300|3
+; Exten. 4302: Provide a way to ring a given phone indefinitely
+exten=4302,1,Dial,AdtranVoFR/4300
+
+exten=4200,1,Playback,transfer
+exten=4200,2,Dial,AdtranVoFR/4200|15
+exten=4200,3,Playback,vm/4200/unavail
+exten=4200,4,Voicemail,4200
+exten=4200,5,Goto,s|4
+exten=4200,103,Playback,vm/4200/busy
+exten=4200,104,Goto,4
+exten=4201,1,Goto,4200|3
+exten=4202,1,Dial,AdtranVoFR/4200
+
+exten=4230,1,Dial,PhoneJack/ixj0
+
+exten=4110,1,Playback,transfer
+;exten=4110,2,Dial,AdtranVoFR/4110|15
+exten=4110,2,Wait,5
+exten=4110,3,Playback,vm/4110/unavail
+exten=4110,4,Voicemail,4110
+exten=4110,5,Goto,s|4
+exten=4110,103,Playback,vm/4110/busy
+exten=4110,104,Goto,4
+exten=4111,1,Goto,4110|3
+exten=4112,1,Dial,AdtranVoFR/4110
+exten=4113,1,Voicemail,s4110
+
+exten=8500,1,VoicemailMain
+exten=8500,2,Goto,s|4
+exten=762,1,Playback,somepeople
+exten=762,2,Wait,4
+exten=762,3,Goto,s|4
+
+; Timeout stuff...  We could send to an operator, or just ditch them.
+exten=t,1,Goto,#|1
+exten=i,1,BackGround,invalid
+
+[other]
+exten=s,1,Playback,digits/9
+exten=s,2,Playback,digits/8
+exten=s,3,Playback,digits/7
+exten=s,4,Goto,100|1
+exten=100,1,Playback,digits/6
+exten=100,2,Playback,digits/5
+exten=100,3,Goto,default|s|4
diff --git a/configs/ixj.conf.sample b/configs/ixj.conf.sample
new file mode 100755
index 000000000..ed6be96c1
--- /dev/null
+++ b/configs/ixj.conf.sample
@@ -0,0 +1,19 @@
+;
+; Internet Phone Jack
+;
+; Configuration file
+;
+[interfaces]
+;
+; Select a mode, either the line jack provides dialtone, reads digits,
+; then starts PBX with the given extension (dialtone mode), or 
+; immediately provides the PBX without reading any digits or providing 
+; any dialtone (this is the immediate mode, the default)
+;
+;mode=immediate
+mode=dialtone
+;
+; List all devices we can use.
+;
+context=local
+device=/dev/ixj0
diff --git a/configs/modules.conf.sample b/configs/modules.conf.sample
new file mode 100755
index 000000000..2fe03093b
--- /dev/null
+++ b/configs/modules.conf.sample
@@ -0,0 +1,14 @@
+;
+; Asterisk configuration file
+;
+; Module Loader configuration file
+;
+[modules]
+autoload=yes
+;load=pbx_gtkconsole.so
+noload=pbx_gtkconsole.so
+noload=pbx_kdeconsole.so
+noload=app_intercom.so
+;load=chan_vofr.so
+;load=chan_h323.so 
+
diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample
new file mode 100755
index 000000000..2d30f3fa0
--- /dev/null
+++ b/configs/voicemail.conf.sample
@@ -0,0 +1,14 @@
+;
+; Voicemail Configuration
+;
+[general]
+; Default format for writing Voicemail
+; format=g723sf|rawgsm|mp3|wav
+format=g723sf|wav
+
+[default]
+4200=2345,Mark Spencer,markster@linux-support.net
+4300=2345,Ben Rigas,ben@american-computer.net
+4310=2345,Sales,sales@marko.net
+4069=2345,Matt Brooks,matt@marko.net
+4110=1379,Rob Flynn,rflynn@blueridge.net
diff --git a/formats/format_wav.c b/formats/format_wav.c
new file mode 100755
index 000000000..e3265d379
--- /dev/null
+++ b/formats/format_wav.c
@@ -0,0 +1,355 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Microsoft WAV File Format using libaudiofile 
+ * 
+ * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#include <asterisk/channel.h>
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/sched.h>
+#include <asterisk/module.h>
+#include <arpa/inet.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <pthread.h>
+#include <audiofile.h>
+
+
+/* Read 320 samples at a time, max */ 
+#define WAV_MAX_SIZE 320
+
+/* Fudge in milliseconds */
+#define WAV_FUDGE 2
+
+struct ast_filestream {
+	/* First entry MUST be reserved for the channel type */
+	void *reserved[AST_RESERVED_POINTERS];
+	/* This is what a filestream means to us */
+	int fd; /* Descriptor */
+	/* Audio File */
+	AFfilesetup afs;
+	AFfilehandle af;
+	int lasttimeout;
+	struct ast_channel *owner;
+	struct ast_filestream *next;
+	struct ast_frame fr;				/* Frame information */
+	char waste[AST_FRIENDLY_OFFSET];	/* Buffer for sending frames, etc */
+	short samples[WAV_MAX_SIZE];
+};
+
+
+static struct ast_filestream *glist = NULL;
+static pthread_mutex_t wav_lock = PTHREAD_MUTEX_INITIALIZER;
+static int glistcnt = 0;
+
+static char *name = "wav";
+static char *desc = "Microsoft WAV format (PCM/16, 8000Hz mono)";
+static char *exts = "wav";
+
+static struct ast_filestream *wav_open(int fd)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	int notok = 0;
+	int fmt, width;
+	double rate;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		tmp->afs = afNewFileSetup();
+		if (!tmp->afs) {
+			ast_log(LOG_WARNING, "Unable to create file setup\n");
+			free(tmp);
+			return NULL;
+		}
+		afInitFileFormat(tmp->afs, AF_FILE_WAVE);
+		tmp->af = afOpenFD(fd, "r", tmp->afs);
+		if (!tmp->af) {
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to open file descriptor\n");
+			free(tmp);
+			return NULL;
+		}
+#if 0
+		afGetFileFormat(tmp->af, &version);
+		if (version != AF_FILE_WAVE) {
+			ast_log(LOG_WARNING, "This is not a wave file (%d)\n", version);
+			notok++;
+		}
+#endif
+		/* Read the format and make sure it's exactly what we seek. */
+		if (afGetChannels(tmp->af, AF_DEFAULT_TRACK) != 1) {
+			ast_log(LOG_WARNING, "Invalid number of channels %d.  Should be mono (1)\n", afGetChannels(tmp->af, AF_DEFAULT_TRACK));
+			notok++;
+		}
+		afGetSampleFormat(tmp->af, AF_DEFAULT_TRACK, &fmt, &width);
+		if (fmt != AF_SAMPFMT_TWOSCOMP) {
+			ast_log(LOG_WARNING, "Input file is not signed\n");
+			notok++;
+		}
+		rate = afGetRate(tmp->af, AF_DEFAULT_TRACK);
+		if ((rate < 7900) || (rate > 8100)) {
+			ast_log(LOG_WARNING, "Rate %f is not close enough to 8000 Hz\n", rate);
+			notok++;
+		}
+		if (width != 16) {
+			ast_log(LOG_WARNING, "Input file is not 16-bit\n");
+			notok++;
+		}
+		if (notok) {
+			afCloseFile(tmp->af);
+			afFreeFileSetup(tmp->afs);
+			free(tmp);
+			return NULL;
+		}
+		if (pthread_mutex_lock(&wav_lock)) {
+			afCloseFile(tmp->af);
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to lock wav list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->fr.data = tmp->samples;
+		tmp->fr.frametype = AST_FRAME_VOICE;
+		tmp->fr.subclass = AST_FORMAT_SLINEAR;
+		/* datalen will vary for each frame */
+		tmp->fr.src = name;
+		tmp->fr.mallocd = 0;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&wav_lock);
+		ast_update_use_count();
+	}
+	return tmp;
+}
+
+static struct ast_filestream *wav_rewrite(int fd, char *comment)
+{
+	/* We don't have any header to read or anything really, but
+	   if we did, it would go here.  We also might want to check
+	   and be sure it's a valid file.  */
+	struct ast_filestream *tmp;
+	if ((tmp = malloc(sizeof(struct ast_filestream)))) {
+		tmp->afs = afNewFileSetup();
+		if (!tmp->afs) {
+			ast_log(LOG_WARNING, "Unable to create file setup\n");
+			free(tmp);
+			return NULL;
+		}
+		/* WAV format */
+		afInitFileFormat(tmp->afs, AF_FILE_WAVE);
+		/* Mono */
+		afInitChannels(tmp->afs, AF_DEFAULT_TRACK, 1);
+		/* Signed linear, 16-bit */
+		afInitSampleFormat(tmp->afs, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+		/* 8000 Hz */
+		afInitRate(tmp->afs, AF_DEFAULT_TRACK, (double)8000.0);
+		tmp->af = afOpenFD(fd, "w", tmp->afs);
+		if (!tmp->af) {
+			afFreeFileSetup(tmp->afs);
+			ast_log(LOG_WARNING, "Unable to open file descriptor\n");
+			free(tmp);
+			return NULL;
+		}
+		if (pthread_mutex_lock(&wav_lock)) {
+			ast_log(LOG_WARNING, "Unable to lock wav list\n");
+			free(tmp);
+			return NULL;
+		}
+		tmp->next = glist;
+		glist = tmp;
+		tmp->fd = fd;
+		tmp->owner = NULL;
+		tmp->lasttimeout = -1;
+		glistcnt++;
+		pthread_mutex_unlock(&wav_lock);
+		ast_update_use_count();
+	} else
+		ast_log(LOG_WARNING, "Out of memory\n");
+	return tmp;
+}
+
+static struct ast_frame *wav_read(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+static void wav_close(struct ast_filestream *s)
+{
+	struct ast_filestream *tmp, *tmpl = NULL;
+	if (pthread_mutex_lock(&wav_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock wav list\n");
+		return;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp == s) {
+			if (tmpl)
+				tmpl->next = tmp->next;
+			else
+				glist = tmp->next;
+			break;
+		}
+		tmpl = tmp;
+		tmp = tmp->next;
+	}
+	glistcnt--;
+	if (s->owner) {
+		s->owner->stream = NULL;
+		if (s->owner->streamid > -1)
+			ast_sched_del(s->owner->sched, s->owner->streamid);
+		s->owner->streamid = -1;
+	}
+	pthread_mutex_unlock(&wav_lock);
+	ast_update_use_count();
+	if (!tmp) 
+		ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
+	afCloseFile(tmp->af);
+	afFreeFileSetup(tmp->afs);
+	close(s->fd);
+	free(s);
+}
+
+static int ast_read_callback(void *data)
+{
+	u_int32_t delay = -1;
+	int retval = 0;
+	int res;
+	struct ast_filestream *s = data;
+	/* Send a frame from the file to the appropriate channel */
+
+	if ((res = afReadFrames(s->af, AF_DEFAULT_TRACK, s->samples, sizeof(s->samples)/2)) < 1) {
+		if (res)
+			ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
+		s->owner->streamid = -1;
+		return 0;
+	}
+	/* Per 8 samples, one milisecond */
+	delay = res / 8;
+	s->fr.frametype = AST_FRAME_VOICE;
+	s->fr.subclass = AST_FORMAT_SLINEAR;
+	s->fr.offset = AST_FRIENDLY_OFFSET;
+	s->fr.datalen = res * 2;
+	s->fr.data = s->samples;
+	s->fr.mallocd = 0;
+	s->fr.timelen = delay;
+	/* Unless there is no delay, we're going to exit out as soon as we
+	   have processed the current frame. */
+	/* If there is a delay, lets schedule the next event */
+	if (delay != s->lasttimeout) {
+		/* We'll install the next timeout now. */
+		s->owner->streamid = ast_sched_add(s->owner->sched, 
+											  delay, 
+											  ast_read_callback, s);
+		
+		s->lasttimeout = delay;
+	} else {
+		/* Just come back again at the same time */
+		retval = -1;
+	}
+	/* Lastly, process the frame */
+	if (ast_write(s->owner, &s->fr)) {
+		ast_log(LOG_WARNING, "Failed to write frame\n");
+		s->owner->streamid = -1;
+		return 0;
+	}
+	
+	return retval;
+}
+
+static int wav_apply(struct ast_channel *c, struct ast_filestream *s)
+{
+	/* Select our owner for this stream, and get the ball rolling. */
+	s->owner = c;
+	ast_read_callback(s);
+	return 0;
+}
+
+static int wav_write(struct ast_filestream *fs, struct ast_frame *f)
+{
+	int res;
+	if (f->frametype != AST_FRAME_VOICE) {
+		ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
+		return -1;
+	}
+	if (f->subclass != AST_FORMAT_SLINEAR) {
+		ast_log(LOG_WARNING, "Asked to write non-signed linear frame (%d)!\n", f->subclass);
+		return -1;
+	}
+	if ((res = afWriteFrames(fs->af, AF_DEFAULT_TRACK, f->data, f->datalen/2)) != f->datalen/2) {
+		ast_log(LOG_WARNING, "Unable to write frame: res=%d (%s)\n", res, strerror(errno));
+		return -1;
+	}	
+	return 0;
+}
+
+char *wav_getcomment(struct ast_filestream *s)
+{
+	return NULL;
+}
+
+int load_module()
+{
+	return ast_format_register(name, exts, AST_FORMAT_SLINEAR,
+								wav_open,
+								wav_rewrite,
+								wav_apply,
+								wav_write,
+								wav_read,
+								wav_close,
+								wav_getcomment);								
+								
+								
+}
+
+int unload_module()
+{
+	struct ast_filestream *tmp, *tmpl;
+	if (pthread_mutex_lock(&wav_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock wav list\n");
+		return -1;
+	}
+	tmp = glist;
+	while(tmp) {
+		if (tmp->owner)
+			ast_softhangup(tmp->owner);
+		tmpl = tmp;
+		tmp = tmp->next;
+		free(tmpl);
+	}
+	pthread_mutex_unlock(&wav_lock);
+	return ast_format_unregister(name);
+}	
+
+int usecount()
+{
+	int res;
+	if (pthread_mutex_lock(&wav_lock)) {
+		ast_log(LOG_WARNING, "Unable to lock wav list\n");
+		return -1;
+	}
+	res = glistcnt;
+	pthread_mutex_unlock(&wav_lock);
+	return res;
+}
+
+char *description()
+{
+	return desc;
+}
+