chan_vpb: Remove deprecated module.

ASTERISK-29597

Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1
pull/24/head
Joshua C. Colp 4 years ago
parent 1eb2d85c99
commit 20b2741232

@ -67,7 +67,6 @@ TONEZONE=@PBX_TONEZONE@
UNBOUND=@PBX_UNBOUND@
UNIXODBC=@PBX_UNIXODBC@
VORBIS=@PBX_VORBIS@
VPB=@PBX_VPB@
WINARCH=@PBX_WINARCH@
ZLIB=@PBX_ZLIB@
TIMERFD=@PBX_TIMERFD@

File diff suppressed because it is too large Load Diff

@ -1,248 +0,0 @@
;
; Voicetronix Voice Processing Board (VPB) telephony interface
;
; Configuration file
;
[general]
;
; Total number of Voicetronix cards in this machine
;
cards=0
;
; Which indication functions to use
; 1 = use Asterisk functions
; 0 = use VPB functions
;
indication=1
;
; Echo Canceller suppression threshold
; 0 = no suppression threshold
; 2048 = -18dB
; 4096 = -24dB
;
;ecsuppthres=0
;
; Inter-digit delay timeout, used when collecting DTMF tones for dialling
; from a station port. Measured in milliseconds.
;
dtmfidd=3000
;
; How to play DTMF tones
; any value = use Asterisk functions
; commented out = use VPB functions
;
;ast-dtmf=1
;
; How to detect DTMF tones
; any value = use Asterisk functions
; commented out = use VPB functions
;
; NOTE: this setting is currently broken, and uncommenting it will
; stop dialling from working. Any volunteers to fix it?
;ast-dtmf-det=1
;
; Use relaxed DTMF detection (ignored unless ast-dtmf-det is set)
;
relaxdtmf=1
;
; When we do a native bridge between two VPB channels:
; yes = only break the connection for '#' and '*'
; no = break the connection for any DTMF
;
; NOTE: this is currently broken, and setting to no will segfault
; Asterisk while dialling. Any volunteers to fix it?
;
break-for-dtmf=yes
;
; The maximum period between received rings. Measures in milliseconds.
;
timer_period_ring=4000
[interfaces]
;
; Default language
;
language=en
;
; Default context
;
context=public
;
; Echo cancellation
; off = no not use echo cancellation
; on = use echo cancellation
;
echocancel=off
;
; Caller ID routines/signalling
; For FXO ports, select one of:
; on = collect caller ID between 1st/2nd rings using VPB routines
; off = do not use caller ID
; bell = bell202 as used in US, using Asterisk's caller ID routines
; v23 = v23 as used in the UK, using Asterisk's caller ID routines
; For FXS ports, set the channel's CID in '"name" <number>' format
;
; NOTE that other caller ID standards are supported in Asterisk, but are
; not yet active in chan_vpb. It should be reasonably trivial to add
; support for the other standards (see the default chan_dahdi.conf for a
; list of them) that Asterisk already handles.
;
callerid=bell
;
; Use a polarity reversal as the trigger for the start of caller ID,
; rather than triggering after the first ring.
;
usepolaritycid=0
;
; Use loop drop to detect the end of a call. On by default, but if you
; experience unexpected hangups, try turning it off.
;
useloopdrop=1
;
; Use in-kernel bridging. This will generally give lower delay audio if
; bridging between two VPB channels. It will not affect bridging
; between VPB channels and other technologies.
;
usenativebridge=1
;
; Software transmit and receive gain. Adjusting these will change the
; volume of audio files that are played (tx) and recorded (rx). It will
; _not_ affect audio between channels in a native bridge. It will,
; however, affect the volume of audio between VPB channels and channels
; using other technologies (such as VoIP channels). Usually it's best to
; leave these as they are. If you're looking to get rid of echo, the
; first thing to do is match your line impedance with the bal1/bal2/bal3
; settings.
;
;txgain=0.0
;rxgain=0.0
;
; Hardware transmit and receive gain. Adjusting these will change the
; volume of all audio on a channel. The allowed range of settings is
; -12.0 to 12.0 (measured in dB).
;
;txhwgain=0.0
;rxhwgain=0.0
;
; Balance register settings, for matching the impedance of the card to
; that of the connected equipment. Only relevant for OpenLine and
; OpenSwitch series cards. Values should be in the range 0 - 255.
;
; We (Voicetronix) have determined the best codec balance values for
; standard interfaces based on their US, Australian and European
; specifications, shown below.
;
; US (600 ohm)
;bal1=0xf8
;bal2=0x1a
;bal3=0x0c
;
; Australia (complex impedance)
;bal1=0xf0
;bal2=0x5d
;bal3=0x79
;
; Europe (CTR-21)
;bal1=0xf0
;bal2=0x6e
;bal3=0x75
;
; Logical groups can be assigned to allow outgoing rollover. Groups range
; from 0 to 63, and multiple groups can be specified.
;
group=1
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is
; ringing and it is a member of a group which is one of your pickup
; groups, then you can answer it by picking up and dialling *8#. For
; simple offices, just make these both the same. Groups range from 0 to
; 63.
;
callgroup=1
pickupgroup=1
;
; If we haven't had a "grunt" (voice activity detection) for this many
; seconds, then we hang up the line due to inactivity. Default is one
; hour.
;
grunttimeout=3600
;
; Type of line and line handling. This setting will usually be overridden
; on a per channel basis. Valid settings are:
; fxo = this is an FXO port
; immediate = this is an FXS port, with no dialtone or dialling
; required (ie it is a "hotline")
; dialtone = this is an FXS port, providing dialtone and dialling
;
mode=immediate
; ------------------------------------------------------------------------
; Channel definitions
;
; Each channel inherits the settings specified above, unless the are
; overridden. As a minimum, the board number and channel number must be
; set, starting from 0 for the first board, and for the channels on each
; board. For example, board 0, channels 0 to 11, then board 1, channels
; 0 to 11 for two OpenSwitch12 cards.
;
;
; First board is an OpenSwitch12 card (jumpers at factory defaults)
;
;board=0
;
;mode=dialtone
;context=from-handset
;group=1
;channel=0
;channel=1
;channel=2
;channel=3
;channel=4
;channel=5
;channel=6
;channel=7
;
;mode=fxo
;context=from-pstn
;group=2
;channel=8
;channel=9
;channel=10
;channel=11
;
; Second board is an OpenLine4
;
;board=1
;
;mode=fxo
;group=2
;context=from-pstn
;channel=0
;channel=1
;channel=2
;channel=3

157
configure vendored

@ -720,10 +720,6 @@ PBX_X11
X11_DIR
X11_INCLUDE
X11_LIB
PBX_VPB
VPB_DIR
VPB_INCLUDE
VPB_LIB
PBX_VORBIS
VORBIS_DIR
VORBIS_INCLUDE
@ -1426,7 +1422,6 @@ with_tonezone
with_unbound
with_unixodbc
with_vorbis
with_vpb
with_x11
with_z
enable_xmldoc
@ -2196,7 +2191,6 @@ Optional Packages:
--with-unbound=PATH use unbound files in PATH
--with-unixodbc=PATH use unixODBC files in PATH
--with-vorbis=PATH use Vorbis files in PATH
--with-vpb=PATH use Voicetronix API files in PATH
--with-x11=PATH use X11 files in PATH
--with-z=PATH use zlib compression files in PATH
@ -3057,52 +3051,6 @@ rm -f conftest.val
as_fn_set_status $ac_retval
} # ac_fn_c_compute_int
# ac_fn_cxx_try_link LINENO
# -------------------------
# Try to link conftest.$ac_ext, and return whether this succeeded.
ac_fn_cxx_try_link ()
{
as_lineno=${as_lineno-"$1"} as_lineno_stack=as_lineno_stack=$as_lineno_stack
rm -f conftest.$ac_objext conftest$ac_exeext
if { { ac_try="$ac_link"
case "(($ac_try" in
*\"* | *\`* | *\\*) ac_try_echo=\$ac_try;;
*) ac_try_echo=$ac_try;;
esac
eval ac_try_echo="\"\$as_me:${as_lineno-$LINENO}: $ac_try_echo\""
$as_echo "$ac_try_echo"; } >&5
(eval "$ac_link") 2>conftest.err
ac_status=$?
if test -s conftest.err; then
grep -v '^ *+' conftest.err >conftest.er1
cat conftest.er1 >&5
mv -f conftest.er1 conftest.err
fi
$as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5
test $ac_status = 0; } && {
test -z "$ac_cxx_werror_flag" ||
test ! -s conftest.err
} && test -s conftest$ac_exeext && {
test "$cross_compiling" = yes ||
test -x conftest$ac_exeext
}; then :
ac_retval=0
else
$as_echo "$as_me: failed program was:" >&5
sed 's/^/| /' conftest.$ac_ext >&5
ac_retval=1
fi
# Delete the IPA/IPO (Inter Procedural Analysis/Optimization) information
# created by the PGI compiler (conftest_ipa8_conftest.oo), as it would
# interfere with the next link command; also delete a directory that is
# left behind by Apple's compiler. We do this before executing the actions.
rm -rf conftest.dSYM conftest_ipa8_conftest.oo
eval $as_lineno_stack; ${as_lineno_stack:+:} unset as_lineno
as_fn_set_status $ac_retval
} # ac_fn_cxx_try_link
cat >config.log <<_ACEOF
This file contains any messages produced by compilers while
running configure, to aid debugging if configure makes a mistake.
@ -12491,38 +12439,6 @@ fi
VPB_DESCRIP="Voicetronix API"
VPB_OPTION="vpb"
PBX_VPB=0
# Check whether --with-vpb was given.
if test "${with_vpb+set}" = set; then :
withval=$with_vpb;
case ${withval} in
n|no)
USE_VPB=no
# -1 is a magic value used by menuselect to know that the package
# was disabled, other than 'not found'
PBX_VPB=-1
;;
y|ye|yes)
ac_mandatory_list="${ac_mandatory_list} VPB"
;;
*)
VPB_DIR="${withval}"
ac_mandatory_list="${ac_mandatory_list} VPB"
;;
esac
fi
X11_DESCRIP="X11"
X11_OPTION="x11"
PBX_X11=0
@ -32358,79 +32274,6 @@ rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
fi
ac_ext=cpp
ac_cpp='$CXXCPP $CPPFLAGS'
ac_compile='$CXX -c $CXXFLAGS $CPPFLAGS conftest.$ac_ext >&5'
ac_link='$CXX -o conftest$ac_exeext $CXXFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5'
ac_compiler_gnu=$ac_cv_cxx_compiler_gnu
if test "${USE_VPB}" != "no"; then
{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for vpb_open in -lvpb" >&5
$as_echo_n "checking for vpb_open in -lvpb... " >&6; }
saved_libs="${LIBS}"
saved_cppflags="${CPPFLAGS}"
if test "x${VPB_DIR}" != "x"; then
if test -d ${VPB_DIR}/lib; then
vpblibdir=${VPB_DIR}/lib
else
vpblibdir=${VPB_DIR}
fi
LIBS="${LIBS} -L${vpblibdir}"
CPPFLAGS="${CPPFLAGS} -I${VPB_DIR}/include"
fi
LIBS="${PTHREAD_LIBS} ${LIBS} -lvpb"
CPPFLAGS="${CPPFLAGS} ${PTHREAD_CFLAGS}"
cat confdefs.h - <<_ACEOF >conftest.$ac_ext
/* end confdefs.h. */
#include <vpbapi.h>
int
main ()
{
int q = vpb_open(0,0);
;
return 0;
}
_ACEOF
if ac_fn_cxx_try_link "$LINENO"; then :
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
$as_echo "yes" >&6; }
ac_cv_lib_vpb_vpb_open="yes"
else
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
$as_echo "no" >&6; }
ac_cv_lib_vpb_vpb_open="no"
fi
rm -f core conftest.err conftest.$ac_objext \
conftest$ac_exeext conftest.$ac_ext
LIBS="${saved_libs}"
CPPFLAGS="${saved_cppflags}"
if test "${ac_cv_lib_vpb_vpb_open}" = "yes"; then
VPB_LIB="-lvpb"
if test "${VPB_DIR}" != ""; then
VPB_LIB="-L${vpblibdir} ${VPB_LIB}"
VPB_INCLUDE="-I${VPB_DIR}/include"
fi
PBX_VPB=1
$as_echo "#define HAVE_VPB 1" >>confdefs.h
fi
fi
ac_ext=c
ac_cpp='$CPP $CPPFLAGS'
ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5'
ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5'
ac_compiler_gnu=$ac_cv_c_compiler_gnu
if test "x${PBX_ZLIB}" != "x1" -a "${USE_ZLIB}" != "no"; then
pbxlibdir=""

@ -598,7 +598,6 @@ AST_EXT_LIB_SETUP([TONEZONE], [tonezone], [tonezone])
AST_EXT_LIB_SETUP([UNBOUND], [unbound], [unbound])
AST_EXT_LIB_SETUP([UNIXODBC], [unixODBC], [unixodbc])
AST_EXT_LIB_SETUP([VORBIS], [Vorbis], [vorbis])
AST_EXT_LIB_SETUP([VPB], [Voicetronix API], [vpb])
AST_EXT_LIB_SETUP([X11], [X11], [x11])
AST_EXT_LIB_SETUP([ZLIB], [zlib compression], [z])
@ -2715,51 +2714,6 @@ AST_EXT_LIB_CHECK([TONEZONE], [tonezone], [tone_zone_find], [dahdi/tonezone.h],
AST_EXT_LIB_CHECK([VORBIS], [vorbis], [vorbis_info_init], [vorbis/codec.h], [-lm -lvorbisenc -lvorbisfile])
AST_C_DECLARE_CHECK([VORBIS_OPEN_CALLBACKS], [OV_CALLBACKS_NOCLOSE], [vorbis/vorbisfile.h])
AC_LANG_PUSH(C++)
if test "${USE_VPB}" != "no"; then
AC_MSG_CHECKING(for vpb_open in -lvpb)
saved_libs="${LIBS}"
saved_cppflags="${CPPFLAGS}"
if test "x${VPB_DIR}" != "x"; then
if test -d ${VPB_DIR}/lib; then
vpblibdir=${VPB_DIR}/lib
else
vpblibdir=${VPB_DIR}
fi
LIBS="${LIBS} -L${vpblibdir}"
CPPFLAGS="${CPPFLAGS} -I${VPB_DIR}/include"
fi
LIBS="${PTHREAD_LIBS} ${LIBS} -lvpb"
CPPFLAGS="${CPPFLAGS} ${PTHREAD_CFLAGS}"
AC_LINK_IFELSE(
[
AC_LANG_PROGRAM(
[#include <vpbapi.h>],
[int q = vpb_open(0,0);])
],
[ AC_MSG_RESULT(yes)
ac_cv_lib_vpb_vpb_open="yes"
],
[ AC_MSG_RESULT(no)
ac_cv_lib_vpb_vpb_open="no"
]
)
LIBS="${saved_libs}"
CPPFLAGS="${saved_cppflags}"
if test "${ac_cv_lib_vpb_vpb_open}" = "yes"; then
VPB_LIB="-lvpb"
if test "${VPB_DIR}" != ""; then
VPB_LIB="-L${vpblibdir} ${VPB_LIB}"
VPB_INCLUDE="-I${VPB_DIR}/include"
fi
PBX_VPB=1
AC_DEFINE([HAVE_VPB], 1, [Define if your system has the VoiceTronix API libraries.])
fi
fi
AC_LANG_POP
AST_EXT_LIB_CHECK([ZLIB], [z], [compress], [zlib.h])
if test "x${PBX_UNIXODBC}" = "x1" -o "x${PBX_IODBC}" = "x1"; then

@ -25,7 +25,7 @@ PACKAGES_DEBIAN="$PACKAGES_DEBIAN libedit-dev libjansson-dev libsqlite3-dev uuid
# Asterisk: for addons:
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libspeex-dev libspeexdsp-dev libogg-dev libvorbis-dev libasound2-dev portaudio19-dev libcurl4-openssl-dev xmlstarlet bison flex"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libpq-dev unixodbc-dev libneon27-dev libgmime-2.6-dev libgmime-3.0-dev liblua5.2-dev liburiparser-dev libxslt1-dev libssl-dev"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libvpb-dev libmysqlclient-dev libbluetooth-dev libradcli-dev freetds-dev libosptk-dev libjack-jackd2-dev bash libcap-dev"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libmysqlclient-dev libbluetooth-dev libradcli-dev freetds-dev libosptk-dev libjack-jackd2-dev bash libcap-dev"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libsnmp-dev libiksemel-dev libcorosync-common-dev libcpg-dev libcfg-dev libnewt-dev libpopt-dev libical-dev libspandsp-dev"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libresample1-dev libc-client2007e-dev binutils-dev libsrtp0-dev libsrtp2-dev libgsm1-dev doxygen graphviz zlib1g-dev libldap2-dev"
PACKAGES_DEBIAN="$PACKAGES_DEBIAN libcodec2-dev libfftw3-dev libsndfile1-dev libunbound-dev"

@ -0,0 +1,6 @@
Subject: chan_vpb
Master-Only: True
This module was deprecated in Asterisk 16
and is now being removed in accordance with
the Asterisk Module Deprecation policy.

@ -1239,9 +1239,6 @@
/* Define if your system has OV_CALLBACKS_NOCLOSE declared. */
#undef HAVE_VORBIS_OPEN_CALLBACKS
/* Define if your system has the VoiceTronix API libraries. */
#undef HAVE_VPB
/* Define to 1 if you have the `vprintf' function. */
#undef HAVE_VPRINTF

@ -134,7 +134,7 @@
* \li libsqlite3.so.0
* \li libss7.so.1
* \li libssl.so.0.9.8 (chan_h323)
* \li libstdc++.so (chan_h323, chan_vpb)
* \li libstdc++.so (chan_h323)
* \li libsuppserv.so
* \li libsybdb.so.5
* \li libsysfs.so.2
@ -144,7 +144,6 @@
* \li libtonezone.so.1.0
* \li libvorbis.so.0
* \li libvorbisenc.so.2
* \li libvpb.a (chan_vpb)
* \li libwrap.so.0
* \li libxcb-xlib.so.0
* \li libxcb.so.1

@ -322,9 +322,6 @@ UUID_LIB=@UUID_LIB@
VORBIS_INCLUDE=@VORBIS_INCLUDE@
VORBIS_LIB=@VORBIS_LIB@
VPB_INCLUDE=@VPB_INCLUDE@
VPB_LIB=@VPB_LIB@
HAVE_DAHDI=@PBX_DAHDI@
DAHDI_INCLUDE=@DAHDI_INCLUDE@

@ -135,7 +135,7 @@ if [ $NO_MENUSELECT -eq 0 ] ; then
fi
runner menuselect/menuselect `gen_cats enable $cat_enables` menuselect.makeopts
mod_disables="res_digium_phone chan_vpb"
mod_disables="res_digium_phone"
if [ $TESTED_ONLY -eq 1 ] ; then
# These modules are not tested at all. They are loaded but nothing is ever done
# with them, no testsuite tests depend on them.

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