chan_vpb: Remove deprecated module.

ASTERISK-29597

Change-Id: I19bb39eed0257ddfef453eb2df5646d073d50fe1

build_tools/menuselect-deps.in

@@ -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@

configs/samples/vpb.conf.sample

@@ -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

configure

@@ -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=""

configure.ac

@@ -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

contrib/scripts/install_prereq

@@ -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"

doc/UPGRADE-staging/chan_vpb_removal.txt

@@ -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.

include/asterisk/autoconfig.h.in

@@ -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
 

include/asterisk/doxygen/licensing.h

@@ -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

makeopts.in

@@ -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@
 

tests/CI/buildAsterisk.sh

@@ -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.