From cf38740db3247ad69d6c6ad5a2b9693aadefca02 Mon Sep 17 00:00:00 2001 From: markster Date: Sun, 5 Dec 1999 07:09:27 +0000 Subject: Version 0.1.0 from FTP git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90 f38db490-d61c-443f-a65b-d21fe96a405b --- codecs/gsm/Makefile | 494 +++++++++++++++++++++++++++++++++++++++++ configs/adtranvofr.conf.sample | 37 +++ configs/extensions.conf.sample | 118 ++++++++++ configs/ixj.conf.sample | 19 ++ configs/modules.conf.sample | 14 ++ configs/voicemail.conf.sample | 14 ++ formats/format_wav.c | 355 +++++++++++++++++++++++++++++ 7 files changed, 1051 insertions(+) create mode 100755 codecs/gsm/Makefile create mode 100755 configs/adtranvofr.conf.sample create mode 100755 configs/extensions.conf.sample create mode 100755 configs/ixj.conf.sample create mode 100755 configs/modules.conf.sample create mode 100755 configs/voicemail.conf.sample create mode 100755 formats/format_wav.c 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 + * + * This program is free software, distributed under the terms of + * the GNU General Public License + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + + +/* 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; +} + -- cgit v1.2.3