mirror of http://gerrit.asterisk.org/asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90 65c4cc65-6c06-0410-ace0-fbb531ad65f31.0
Mark Spencer
26 years ago
parent
a8ffb8b553
commit
ca87930eb1
@ -0,0 +1,494 @@
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# Copyright 1992-1996 by Jutta Degener and Carsten Bormann, Technische
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# Universitaet Berlin. See the accompanying file "COPYRIGHT" for
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# details. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
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# Machine- or installation dependent flags you should configure to port
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SASR = -DSASR
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######### Define SASR if >> is a signed arithmetic shift (-1 >> 1 == -1)
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MULHACK = -DUSE_FLOAT_MUL
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######### Define this if your host multiplies floats faster than integers,
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######### e.g. on a SPARCstation.
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FAST = -DFAST
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######### Define together with USE_FLOAT_MUL to enable the GSM library's
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######### approximation option for incorrect, but good-enough results.
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# LTP_CUT = -DLTP_CUT
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LTP_CUT =
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######### Define to enable the GSM library's long-term correlation
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######### approximation option---faster, but worse; works for
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######### both integer and floating point multiplications.
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######### This flag is still in the experimental stage.
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WAV49 = -DWAV49
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#WAV49 =
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######### Define to enable the GSM library's option to pack GSM frames
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######### in the style used by the WAV #49 format. If you want to write
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######### a tool that produces .WAV files which contain GSM-encoded data,
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######### define this, and read about the GSM_OPT_WAV49 option in the
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######### manual page on gsm_option(3).
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# Choose a compiler. The code works both with ANSI and K&R-C.
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# Use -DNeedFunctionPrototypes to compile with, -UNeedFunctionPrototypes to
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# compile without, function prototypes in the header files.
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#
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# You can use the -DSTUPID_COMPILER to circumvent some compilers'
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# static limits regarding the number of subexpressions in a statement.
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# CC = cc
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# CCFLAGS = -c -DSTUPID_COMPILER
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# CC = /usr/lang/acc
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# CCFLAGS = -c -O
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CC = gcc -ansi -pedantic
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CCFLAGS += -c -DNeedFunctionPrototypes=1 -finline-functions -funroll-loops
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LD = $(CC)
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# LD = gcc
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# LDFLAGS =
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# If your compiler needs additional flags/libraries, regardless of
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# the source compiled, configure them here.
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# CCINC = -I/usr/gnu/lib/gcc-2.1/gcc-lib/sparc-sun-sunos4.1.2/2.1/include
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######### Includes needed by $(CC)
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# LDINC = -L/usr/gnu/lib/gcc-2.1/gcc-lib/sparc-sun-sunos4.1.2/2.1
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######### Library paths needed by $(LD)
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# LDLIB = -lgcc
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######### Additional libraries needed by $(LD)
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# Where do you want to install libraries, binaries, a header file
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# and the manual pages?
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#
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# Leave INSTALL_ROOT empty (or just don't execute "make install") to
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# not install gsm and toast outside of this directory.
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INSTALL_ROOT =
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# Where do you want to install the gsm library, header file, and manpages?
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#
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# Leave GSM_INSTALL_ROOT empty to not install the GSM library outside of
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# this directory.
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GSM_INSTALL_ROOT = $(INSTALL_ROOT)
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GSM_INSTALL_LIB = $(GSM_INSTALL_ROOT)/lib
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GSM_INSTALL_INC = $(GSM_INSTALL_ROOT)/inc
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GSM_INSTALL_MAN = $(GSM_INSTALL_ROOT)/man/man3
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# Where do you want to install the toast binaries and their manpage?
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#
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# Leave TOAST_INSTALL_ROOT empty to not install the toast binaries outside
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# of this directory.
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TOAST_INSTALL_ROOT = $(INSTALL_ROOT)
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TOAST_INSTALL_BIN = $(TOAST_INSTALL_ROOT)/bin
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TOAST_INSTALL_MAN = $(TOAST_INSTALL_ROOT)/man/man1
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# Other tools
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SHELL = /bin/sh
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LN = ln
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BASENAME = basename
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AR = ar
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ARFLAGS = cr
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RMFLAGS = -f
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FIND = find
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COMPRESS = compress
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COMPRESSFLAGS =
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# RANLIB = true
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RANLIB = ranlib
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#
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# You shouldn't have to configure below this line if you're porting.
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#
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# Local Directories
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ROOT = .
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ADDTST = $(ROOT)/add-test
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TST = $(ROOT)/tst
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MAN = $(ROOT)/man
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BIN = $(ROOT)/bin
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SRC = $(ROOT)/src
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LIB = $(ROOT)/lib
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TLS = $(ROOT)/tls
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INC = $(ROOT)/inc
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# Flags
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# DEBUG = -DNDEBUG
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######### Remove -DNDEBUG to enable assertions.
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CFLAGS = $(CCFLAGS) $(SASR) $(DEBUG) $(MULHACK) $(FAST) $(LTP_CUT) \
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$(WAV49) $(CCINC) -I$(INC)
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######### It's $(CC) $(CFLAGS)
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LFLAGS = $(LDFLAGS) $(LDINC)
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######### It's $(LD) $(LFLAGS)
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# Targets
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LIBGSM = $(LIB)/libgsm.a
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TOAST = $(BIN)/toast
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UNTOAST = $(BIN)/untoast
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TCAT = $(BIN)/tcat
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# Headers
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GSM_HEADERS = $(INC)/gsm.h
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HEADERS = $(INC)/proto.h \
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$(INC)/unproto.h \
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$(INC)/config.h \
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$(INC)/private.h \
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$(INC)/gsm.h \
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$(INC)/toast.h \
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$(TLS)/taste.h
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# Sources
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GSM_SOURCES = $(SRC)/add.c \
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$(SRC)/code.c \
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$(SRC)/debug.c \
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$(SRC)/decode.c \
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$(SRC)/long_term.c \
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$(SRC)/lpc.c \
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$(SRC)/preprocess.c \
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$(SRC)/rpe.c \
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$(SRC)/gsm_destroy.c \
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$(SRC)/gsm_decode.c \
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$(SRC)/gsm_encode.c \
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$(SRC)/gsm_explode.c \
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$(SRC)/gsm_implode.c \
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$(SRC)/gsm_create.c \
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$(SRC)/gsm_print.c \
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$(SRC)/gsm_option.c \
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$(SRC)/short_term.c \
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$(SRC)/table.c
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TOAST_SOURCES = $(SRC)/toast.c \
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$(SRC)/toast_lin.c \
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$(SRC)/toast_ulaw.c \
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$(SRC)/toast_alaw.c \
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$(SRC)/toast_audio.c
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SOURCES = $(GSM_SOURCES) \
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$(TOAST_SOURCES) \
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$(ADDTST)/add_test.c \
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$(TLS)/sour.c \
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$(TLS)/ginger.c \
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$(TLS)/sour1.dta \
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$(TLS)/sour2.dta \
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$(TLS)/bitter.c \
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$(TLS)/bitter.dta \
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$(TLS)/taste.c \
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$(TLS)/sweet.c \
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$(TST)/cod2lin.c \
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$(TST)/cod2txt.c \
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$(TST)/gsm2cod.c \
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$(TST)/lin2cod.c \
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$(TST)/lin2txt.c
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# Object files
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GSM_OBJECTS = $(SRC)/add.o \
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$(SRC)/code.o \
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$(SRC)/debug.o \
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$(SRC)/decode.o \
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$(SRC)/long_term.o \
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$(SRC)/lpc.o \
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$(SRC)/preprocess.o \
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$(SRC)/rpe.o \
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$(SRC)/gsm_destroy.o \
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$(SRC)/gsm_decode.o \
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$(SRC)/gsm_encode.o \
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$(SRC)/gsm_explode.o \
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$(SRC)/gsm_implode.o \
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$(SRC)/gsm_create.o \
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$(SRC)/gsm_print.o \
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$(SRC)/gsm_option.o \
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$(SRC)/short_term.o \
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$(SRC)/table.o
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TOAST_OBJECTS = $(SRC)/toast.o \
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$(SRC)/toast_lin.o \
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$(SRC)/toast_ulaw.o \
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$(SRC)/toast_alaw.o \
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$(SRC)/toast_audio.o
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OBJECTS = $(GSM_OBJECTS) $(TOAST_OBJECTS)
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# Manuals
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GSM_MANUALS = $(MAN)/gsm.3 \
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$(MAN)/gsm_explode.3 \
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$(MAN)/gsm_option.3 \
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$(MAN)/gsm_print.3
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TOAST_MANUALS = $(MAN)/toast.1
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MANUALS = $(GSM_MANUALS) $(TOAST_MANUALS) $(MAN)/bitter.1
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# Other stuff in the distribution
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STUFF = ChangeLog \
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INSTALL \
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MACHINES \
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MANIFEST \
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Makefile \
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README \
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$(ADDTST)/add_test.dta \
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$(TLS)/bitter.dta \
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$(TST)/run
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# Install targets
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GSM_INSTALL_TARGETS = \
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$(GSM_INSTALL_LIB)/libgsm.a \
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$(GSM_INSTALL_INC)/gsm.h \
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$(GSM_INSTALL_MAN)/gsm.3 \
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$(GSM_INSTALL_MAN)/gsm_explode.3 \
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$(GSM_INSTALL_MAN)/gsm_option.3 \
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$(GSM_INSTALL_MAN)/gsm_print.3
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TOAST_INSTALL_TARGETS = \
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$(TOAST_INSTALL_BIN)/toast \
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$(TOAST_INSTALL_BIN)/tcat \
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$(TOAST_INSTALL_BIN)/untoast \
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$(TOAST_INSTALL_MAN)/toast.1
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# Default rules
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.c.o:
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$(CC) $(CFLAGS) $?
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@-mv `$(BASENAME) $@` $@ > /dev/null 2>&1
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# Target rules
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all: $(LIBGSM) $(TOAST) $(TCAT) $(UNTOAST)
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@-echo $(ROOT): Done.
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tst: $(TST)/lin2cod $(TST)/cod2lin $(TOAST) $(TST)/test-result
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@-echo tst: Done.
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addtst: $(ADDTST)/add $(ADDTST)/add_test.dta
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$(ADDTST)/add < $(ADDTST)/add_test.dta > /dev/null
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@-echo addtst: Done.
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misc: $(TLS)/sweet $(TLS)/bitter $(TLS)/sour $(TLS)/ginger \
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$(TST)/lin2txt $(TST)/cod2txt $(TST)/gsm2cod
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@-echo misc: Done.
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install: toastinstall gsminstall
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@-echo install: Done.
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# The basic API: libgsm
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$(LIBGSM): $(LIB) $(GSM_OBJECTS)
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-rm $(RMFLAGS) $(LIBGSM)
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$(AR) $(ARFLAGS) $(LIBGSM) $(GSM_OBJECTS)
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$(RANLIB) $(LIBGSM)
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# Toast, Untoast and Tcat -- the compress-like frontends to gsm.
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$(TOAST): $(BIN) $(TOAST_OBJECTS) $(LIBGSM)
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$(LD) $(LFLAGS) -o $(TOAST) $(TOAST_OBJECTS) $(LIBGSM) $(LDLIB)
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$(UNTOAST): $(BIN) $(TOAST)
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-rm $(RMFLAGS) $(UNTOAST)
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$(LN) $(TOAST) $(UNTOAST)
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$(TCAT): $(BIN) $(TOAST)
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-rm $(RMFLAGS) $(TCAT)
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$(LN) $(TOAST) $(TCAT)
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# The local bin and lib directories
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$(BIN):
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if [ ! -d $(BIN) ] ; then mkdir $(BIN) ; fi
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$(LIB):
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if [ ! -d $(LIB) ] ; then mkdir $(LIB) ; fi
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# Installation
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gsminstall:
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-if [ x"$(GSM_INSTALL_ROOT)" != x ] ; then \
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make $(GSM_INSTALL_TARGETS) ; \
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fi
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toastinstall:
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-if [ x"$(TOAST_INSTALL_ROOT)" != x ]; then \
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make $(TOAST_INSTALL_TARGETS); \
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fi
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gsmuninstall:
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-if [ x"$(GSM_INSTALL_ROOT)" != x ] ; then \
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rm $(RMFLAGS) $(GSM_INSTALL_TARGETS) ; \
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fi
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toastuninstall:
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-if [ x"$(TOAST_INSTALL_ROOT)" != x ] ; then \
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rm $(RMFLAGS) $(TOAST_INSTALL_TARGETS); \
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fi
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$(TOAST_INSTALL_BIN)/toast: $(TOAST)
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-rm $@
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cp $(TOAST) $@
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chmod 755 $@
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$(TOAST_INSTALL_BIN)/untoast: $(TOAST_INSTALL_BIN)/toast
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-rm $@
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ln $? $@
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$(TOAST_INSTALL_BIN)/tcat: $(TOAST_INSTALL_BIN)/toast
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-rm $@
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ln $? $@
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$(TOAST_INSTALL_MAN)/toast.1: $(MAN)/toast.1
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_MAN)/gsm.3: $(MAN)/gsm.3
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_MAN)/gsm_option.3: $(MAN)/gsm_option.3
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_MAN)/gsm_explode.3: $(MAN)/gsm_explode.3
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_MAN)/gsm_print.3: $(MAN)/gsm_print.3
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_INC)/gsm.h: $(INC)/gsm.h
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-rm $@
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cp $? $@
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chmod 444 $@
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$(GSM_INSTALL_LIB)/libgsm.a: $(LIBGSM)
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-rm $@
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cp $? $@
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chmod 444 $@
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# Distribution
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dist: gsm-1.0.tar.Z
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@echo dist: Done.
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gsm-1.0.tar.Z: $(STUFF) $(SOURCES) $(HEADERS) $(MANUALS)
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( cd $(ROOT)/..; \
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tar cvf - `cat $(ROOT)/gsm-1.0/MANIFEST \
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| sed '/^#/d'` \
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) | $(COMPRESS) $(COMPRESSFLAGS) > $(ROOT)/gsm-1.0.tar.Z
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# Clean
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uninstall: toastuninstall gsmuninstall
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@-echo uninstall: Done.
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semi-clean:
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-rm $(RMFLAGS) */*.o \
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$(TST)/lin2cod $(TST)/lin2txt \
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$(TST)/cod2lin $(TST)/cod2txt \
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$(TST)/gsm2cod \
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$(TST)/*.*.*
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-$(FIND) . \( -name core -o -name foo \) \
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-print | xargs rm $(RMFLAGS)
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clean: semi-clean
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-rm $(RMFLAGS) $(LIBGSM) $(ADDTST)/add \
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$(TOAST) $(TCAT) $(UNTOAST) \
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$(ROOT)/gsm-1.0.tar.Z
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# Two tools that helped me generate gsm_encode.c and gsm_decode.c,
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# but aren't generally needed to port this.
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$(TLS)/sweet: $(TLS)/sweet.o $(TLS)/taste.o
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$(LD) $(LFLAGS) -o $(TLS)/sweet \
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$(TLS)/sweet.o $(TLS)/taste.o $(LDLIB)
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$(TLS)/bitter: $(TLS)/bitter.o $(TLS)/taste.o
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$(LD) $(LFLAGS) -o $(TLS)/bitter \
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$(TLS)/bitter.o $(TLS)/taste.o $(LDLIB)
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# A version of the same family that Jeff Chilton used to implement
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# the WAV #49 GSM format.
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$(TLS)/ginger: $(TLS)/ginger.o $(TLS)/taste.o
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$(LD) $(LFLAGS) -o $(TLS)/ginger \
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$(TLS)/ginger.o $(TLS)/taste.o $(LDLIB)
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$(TLS)/sour: $(TLS)/sour.o $(TLS)/taste.o
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$(LD) $(LFLAGS) -o $(TLS)/sour \
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$(TLS)/sour.o $(TLS)/taste.o $(LDLIB)
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# Run $(ADDTST)/add < $(ADDTST)/add_test.dta to make sure the
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# basic arithmetic functions work as intended.
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$(ADDTST)/add: $(ADDTST)/add_test.o
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$(LD) $(LFLAGS) -o $(ADDTST)/add $(ADDTST)/add_test.o $(LDLIB)
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# Various conversion programs between linear, text, .gsm and the code
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# format used by the tests we ran (.cod). We paid for the test data,
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# so I guess we can't just provide them with this package. Still,
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# if you happen to have them lying around, here's the code.
|
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#
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# You can use gsm2cod | cod2txt independently to look at what's
|
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# coded inside the compressed frames, although this shouldn't be
|
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# hard to roll on your own using the gsm_print() function from
|
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# the API.
|
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$(TST)/test-result: $(TST)/lin2cod $(TST)/cod2lin $(TOAST) $(TST)/run
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( cd $(TST); ./run )
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$(TST)/lin2txt: $(TST)/lin2txt.o $(LIBGSM)
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$(LD) $(LFLAGS) -o $(TST)/lin2txt \
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$(TST)/lin2txt.o $(LIBGSM) $(LDLIB)
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$(TST)/lin2cod: $(TST)/lin2cod.o $(LIBGSM)
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$(LD) $(LFLAGS) -o $(TST)/lin2cod \
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$(TST)/lin2cod.o $(LIBGSM) $(LDLIB)
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|
||||
$(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)
|
||||
@ -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
|
||||
@ -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
|
||||
@ -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
|
||||
@ -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
|
||||
|
||||
@ -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
|
||||
@ -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;
|
||||
}
|
||||
|
||||
Loading…
Reference in new issue