remove unused header and channel module

use auto-build for channels


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

3 files changed, 37 insertions, 1274 deletions

diff --git a/channels/Makefile b/channels/Makefile
index 417046177..8425ff61c 100644
--- a/channels/Makefile
+++ b/channels/Makefile
@@ -3,7 +3,7 @@
 # 
 # Makefile for channel drivers
 #
-# Copyright (C) 1999-2005, Mark Spencer
+# Copyright (C) 1999-2006, Digium, Inc.
 #
 # Mark Spencer <markster@digium.com>
 #
@@ -14,7 +14,7 @@
 # the GNU General Public License
 #
 
-CHANNEL_LIBS=chan_sip.so chan_agent.so chan_mgcp.so chan_iax2.so chan_local.so chan_skinny.so chan_features.so
+MODS:=$(patsubst %.c,%.so,$(wildcard chan_*.c))
 
 ifeq (${OSARCH},OpenBSD)
   PTLIB=-lpt_OpenBSD_x86_r
@@ -28,9 +28,9 @@ ifeq (${OSARCH},Linux)
 endif
 
 ifeq (${OSARCH},CYGWIN)
-CYGSOLINK=-Wl,--out-implib=lib$@.a -Wl,--export-all-symbols
-CYGSOLIB=-L.. -L. -L../res -lasterisk.dll -lres_features.so
-CYG_CHAN_AGENT=-lres_monitor.so
+  CYGSOLINK=-Wl,--out-implib=lib$@.a -Wl,--export-all-symbols
+  CYGSOLIB=-L.. -L. -L../res -lasterisk.dll -lres_features.so
+  CYG_CHAN_AGENT=-lres_monitor.so
 endif
 
 ifeq ($(PROC),sparc64)
@@ -49,52 +49,51 @@ ifeq (${OSARCH},NetBSD)
   H323LIB=-lh323_NetBSD_x86_r
 endif
 
-ifneq (${OSARCH},Darwin)
-  ifneq (${OSARCH},SunOS)
-    ifneq (${OSARCH},CYGWIN)
-       CHANNEL_LIBS+=chan_oss.so
-    endif
-  endif
+ifeq (${OSARCH},Darwin)
+  MODS:=$(filter-out chan_oss.so,$(MODS))
 endif
 
 ifeq (${OSARCH},SunOS)
+  MODS:=$(filter-out chan_oss.so,$(MODS))
   SOLINK+=-lrt
 endif
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/ixjuser.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/ixjuser.h),)
-  CHANNEL_LIBS+=chan_phone.so
+ifeq (${OSARCH},CYGWIN)
+  MODS:=$(filter-out chan_oss.so,$(MODS))
+endif
+
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/ixjuser.h $(CROSS_COMPILE_TARGET)/usr/local/include/ixjuser.h),)
+  MODS:=$(filter-out chan_phone.so,$(MODS))
 endif
 
-#
-# Asterisk SMDI integration
-#
 ifeq (${WITH_SMDI},1)
 CFLAGS+=-DWITH_SMDI
 endif
 
-ifneq ($(wildcard h323/libchanh323.a),)
-  CHANNEL_LIBS+=chan_h323.so
+ifeq ($(wildcard h323/libchanh323.a),)
+  MODS:=$(filter-out chan_h323.so,$(MODS))
 endif
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/mISDNuser/mISDNlib.h),)
-  CHANNEL_LIBS+=chan_misdn.so
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/mISDNuser/mISDNlib.h),)
+  MODS:=$(filter-out chan_misdn.so,$(MODS))
+else
   CFLAGS+=-Imisdn 
 endif
 
 CFLAGS+=-Wno-missing-prototypes -Wno-missing-declarations
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/alsa/asoundlib.h),)
-  CHANNEL_LIBS+=chan_alsa.so
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/alsa/asoundlib.h),)
+  MODS:=$(filter-out chan_alsa.so,$(MODS))
 endif
 
 ifndef WITHOUT_PRI
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libpri.so.1)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/libpri.so.1),)
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libpri.so.1 $(CROSS_COMPILE_TARGET)/usr/local/lib/libpri.so.1),)
   CFLAGS+=-DZAPATA_PRI
   ZAPPRI=-lpri
 endif
 endif # WITHOUT_PRI
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libmfcr2.so.1)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/lib/libmfcr2.so.1),)
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/lib/libmfcr2.so.1 $(CROSS_COMPILE_TARGET)/usr/local/lib/libmfcr2.so.1),)
   CFLAGS+=-DZAPATA_R2
   ZAPR2=-lmfcr2
 endif
@@ -107,7 +106,12 @@ ifneq ($(wildcard alsa-monitor.h),)
 endif
 
 ifndef WITHOUT_ZAPTEL
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/pkg/include/zaptel.h),)
+ZAPAVAIL:=$(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h $(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h)
+endif
+
+ifeq (${ZAPAVAIL},)
+  MODS:=$(filter-out chan_zap.so,$(MODS))
+else
   ifeq (${OSARCH},NetBSD)
     SOLINK+=-L$(CROSS_COMPILE_TARGET)/usr/pkg/lib
   endif
@@ -115,27 +119,26 @@ ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard
     SOLINK+=-L$(CROSS_COMPILE_TARGET)/usr/local/lib
   endif
   CFLAGS+=-DIAX_TRUNKING
-  CHANNEL_LIBS+=chan_zap.so
 endif
-endif # WITHOUT_ZAPTEL
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/vpbapi.h),)
-  CHANNEL_LIBS+=chan_vpb.so
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/vpbapi.h),)
+  MODS:=$(filter-out chan_vpb.so,$(MODS))
+else
   CFLAGS+=-DLINUX
 endif
 
 CFLAGS+=-DCRYPTO
 
 ifneq ($(OSARCH),CYGWIN)
-CFLAGS+=-fPIC
+  CFLAGS+=-fPIC
 endif
 
 CFLAGS+=#-DVOFRDUMPER
 
 ZAPDIR=/usr/lib
 
-ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/nbs.h),)
-  CHANNEL_LIBS+=chan_nbs.so
+ifeq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/nbs.h),)
+  MODS:=$(filter-out chan_nbs.so,$(MODS))
 endif
 
 ifndef OPENH323DIR
@@ -146,9 +149,7 @@ ifndef PWLIBDIR
   PWLIBDIR=$(HOME)/pwlib
 endif
 
-#CFLAGS+=$(shell [ -f $(ZAPDIR)/libzap.a ] && echo "-I$(ZAPDIR)")
-
-all: depend $(CHANNEL_LIBS) 
+all: depend $(MODS) 
 
 clean:
 	rm -f *.so *.o .depend
@@ -245,8 +246,7 @@ chan_misdn_config.o: chan_misdn_config.c misdn/chan_misdn_config.h
 
 
 install: all
-	for x in $(CHANNEL_LIBS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
-	if ! [ -f chan_iax.so ]; then rm -f $(DESTDIR)$(MODULES_DIR)/chan_iax.so ; fi
+	for x in $(MODS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ; done
 
 uninstall:
 
diff --git a/channels/adtranvofr.h b/channels/adtranvofr.h
deleted file mode 100644
index 88fd428ee..000000000
--- a/channels/adtranvofr.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Asterisk -- A telephony toolkit for Linux.
- *
- * Implementation of Voice over Frame Relay, Adtran Style
- * 
- * Copyright (C) 1999, Mark Spencer
- *
- * Mark Spencer <markster@linux-support.net>
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License
- */
-
-#ifndef _ADTRANVOFR_H
-#define _ADTRANVOFR_H
-
-#define VOFR_CONTROL_ADTRAN		0x0
-#define VOFR_CONTROL_VOICE   		0x1
-#define VOFR_CONTROL_RFC1490		0x3
-
-#define VOFR_TYPE_SIGNAL		0x0
-#define VOFR_TYPE_VOICE			0x1
-#define VOFR_TYPE_ANSWER		0x2
-#define VOFR_TYPE_FAX			0x3
-#define VOFR_TYPE_DTMF			0x4
-
-#define VOFR_CARD_TYPE_UNSPEC		0x0
-#define VOFR_CARD_TYPE_FXS		0x1
-#define VOFR_CARD_TYPE_FXO		0x2
-#define VOFR_CARD_TYPE_ENM		0x3
-#define VOFR_CARD_TYPE_VCOM		0x4
-#define VOFR_CARD_TYPE_ASTERISK		0xf
-
-#define VOFR_MODULATION_SINGLE		0x0
-#define VOFR_MODULATION_V21		0x1
-#define VOFR_MODULATION_V27ter_2	0x2
-#define VOFR_MODULATION_V27ter_4	0x3
-#define VOFR_MODULATION_V29_7		0x4
-#define VOFR_MODULATION_V29_9		0x5
-#define VOFR_MODULATION_V33_12		0x6
-#define VOFR_MODULATION_V33_14		0x7
-
-#define VOFR_ROUTE_NONE			0x0
-#define VOFR_ROUTE_LOCAL		0x1
-#define VOFR_ROUTE_VOICE		0x2
-#define VOFR_ROUTE_DTE1			0x4
-#define VOFR_ROUTE_DTE2			0x8
-#define VOFR_ROUTE_DTE			0xC
-
-#define VOFR_MASK_EI			0x80
-#define VOFR_MASK_LI			0x40
-#define VOFR_MASK_CONTROL		0x3F
-
-#define VOFR_SIGNAL_ON_HOOK		0x00
-#define VOFR_SIGNAL_OFF_HOOK		0x01
-#define VOFR_SIGNAL_RING		0x40
-#define VOFR_SIGNAL_SWITCHED_DIAL	0x08
-#define VOFR_SIGNAL_BUSY		0x02
-#define VOFR_SIGNAL_TRUNK_BUSY		0x04
-#define VOFR_SIGNAL_UNKNOWN		0x10
-#define VOFR_SIGNAL_OFFHOOK		0x81
-
-#define VOFR_TRACE_SIGNAL		1 << 0
-#define VOFR_TRACE_VOICE		1 << 1
-
-#define VOFR_MAX_PKT_SIZE		1500
-
-/*
- * Wire level protocol 
- */
-
-struct vofr_hdr {
-	u_int8_t control;		/* Also contains unused EI and LI bits */
-#if __BYTE_ORDER == __LITTLE_ENDIAN
-	u_int8_t dtype:4;		/* Data type */
-	u_int8_t ctag:4;		/* Connect tag */
-	u_int8_t dlcih:4;		/* Hi 2 bits of DLCI x-ref */
-	u_int8_t vflags:4;		/* Voice Routing Flags */
-	u_int8_t dlcil;			/* Lo 8 bits of DLCI x-ref */
-	u_int8_t cid;			/* Channel ID */
-	u_int8_t mod:4;			/* Modulation */
-	u_int8_t remid:4;		/* Remote ID */
-#elif __BYTE_ORDER == __BIG_ENDIAN
-	u_int8_t ctag:4;		/* Connect tag */
-	u_int8_t dtype:4;		/* Data type */
-	u_int8_t vflags:4;		/* Voice Routing Flags */
-	u_int8_t dlcih:4;		/* Hi 2 bits of DLCI x-ref */
-	u_int8_t dlcil;			/* Lo 8 bits of DLCI x-ref */
-	u_int8_t cid;			/* Channel ID */
-	u_int8_t remid:4;		/* Remote ID or Relay CMD*/
-	u_int8_t mod:4;			/* Modulation */
-#else
-#error	"Please fix <bytesex.h>"
-#endif
-#ifdef __GNUC__
-	u_int8_t data[0];		/* Data */
-#endif
-};
-
-#define VOFR_HDR_SIZE 6
-
-/* Number of milliseconds to fudge -- experimentally derived */
-#define VOFR_FUDGE 2
-
-#endif
diff --git a/channels/chan_oss_old.c b/channels/chan_oss_old.c
deleted file mode 100644
index c1613d3b6..000000000
--- a/channels/chan_oss_old.c
+++ /dev/null
@@ -1,1132 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
-
-/*
- * Use /dev/dsp as a channel, and the console to command it :).
- *
- * The full-duplex "simulation" is pretty weak.  This is generally a 
- * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
- * writing a driver.
- *
- * \ingroup channel_drivers
- */
-
-#include <unistd.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <sys/ioctl.h>
-#include <sys/time.h>
-#include <string.h>
-#include <stdlib.h>
-#include <stdio.h>
-
-#ifdef __linux
-#include <linux/soundcard.h>
-#elif defined(__FreeBSD__)
-#include <sys/soundcard.h>
-#else
-#include <soundcard.h>
-#endif
-
-#include "asterisk.h"
-
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
-#include "asterisk/lock.h"
-#include "asterisk/frame.h"
-#include "asterisk/logger.h"
-#include "asterisk/channel.h"
-#include "asterisk/module.h"
-#include "asterisk/options.h"
-#include "asterisk/pbx.h"
-#include "asterisk/config.h"
-#include "asterisk/cli.h"
-#include "asterisk/utils.h"
-#include "asterisk/causes.h"
-#include "asterisk/endian.h"
-
-#include "busy.h"
-#include "ringtone.h"
-#include "ring10.h"
-#include "answer.h"
-
-/* Which device to use */
-#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
-#define DEV_DSP "/dev/audio"
-#else
-#define DEV_DSP "/dev/dsp"
-#endif
-
-/* Lets use 160 sample frames, just like GSM.  */
-#define FRAME_SIZE 160
-
-/* When you set the frame size, you have to come up with
-   the right buffer format as well. */
-/* 5 64-byte frames = one frame */
-#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
-
-/* Don't switch between read/write modes faster than every 300 ms */
-#define MIN_SWITCH_TIME 600
-
-static struct timeval lasttime;
-
-static int usecnt;
-static int silencesuppression = 0;
-static int silencethreshold = 1000;
-static int playbackonly = 0;
-
-
-AST_MUTEX_DEFINE_STATIC(usecnt_lock);
-
-static const char type[] = "Console";
-static const char desc[] = "OSS Console Channel Driver";
-static const char tdesc[] = "OSS Console Channel Driver";
-static const char config[] = "oss.conf";
-
-static char context[AST_MAX_CONTEXT] = "default";
-static char language[MAX_LANGUAGE] = "";
-static char exten[AST_MAX_EXTENSION] = "s";
-
-static int hookstate=0;
-
-static short silence[FRAME_SIZE] = {0, };
-
-struct sound {
-	int ind;
-	short *data;
-	int datalen;
-	int samplen;
-	int silencelen;
-	int repeat;
-};
-
-static struct sound sounds[] = {
-	{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
-	{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
-	{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
-	{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
-	{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
-};
-
-/* Sound command pipe */
-static int sndcmd[2];
-
-static struct chan_oss_pvt {
-	/* We only have one OSS structure -- near sighted perhaps, but it
-	   keeps this driver as simple as possible -- as it should be. */
-	struct ast_channel *owner;
-	char exten[AST_MAX_EXTENSION];
-	char context[AST_MAX_CONTEXT];
-} oss;
-
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
-static int oss_digit(struct ast_channel *c, char digit);
-static int oss_text(struct ast_channel *c, const char *text);
-static int oss_hangup(struct ast_channel *c);
-static int oss_answer(struct ast_channel *c);
-static struct ast_frame *oss_read(struct ast_channel *chan);
-static int oss_call(struct ast_channel *c, char *dest, int timeout);
-static int oss_write(struct ast_channel *chan, struct ast_frame *f);
-static int oss_indicate(struct ast_channel *chan, int cond);
-static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
-
-static const struct ast_channel_tech oss_tech = {
-	.type = type,
-	.description = tdesc,
-	.capabilities = AST_FORMAT_SLINEAR,
-	.requester = oss_request,
-	.send_digit = oss_digit,
-	.send_text = oss_text,
-	.hangup = oss_hangup,
-	.answer = oss_answer,
-	.read = oss_read,
-	.call = oss_call,
-	.write = oss_write,
-	.indicate = oss_indicate,
-	.fixup = oss_fixup,
-};
-
-static int time_has_passed(void)
-{
-	struct timeval tv;
-	int ms;
-	gettimeofday(&tv, NULL);
-	ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
-			(tv.tv_usec - lasttime.tv_usec) / 1000;
-	if (ms > MIN_SWITCH_TIME)
-		return -1;
-	return 0;
-}
-
-/* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
-   with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
-   usually plenty. */
-
-static pthread_t sthread;
-
-#define MAX_BUFFER_SIZE 100
-static int buffersize = 3;
-
-static int full_duplex = 0;
-
-/* Are we reading or writing (simulated full duplex) */
-static int readmode = 1;
-
-/* File descriptor for sound device */
-static int sounddev = -1;
-
-static int autoanswer = 1;
- 
-#if 0
-static int calc_loudness(short *frame)
-{
-	int sum = 0;
-	int x;
-	for (x=0;x<FRAME_SIZE;x++) {
-		if (frame[x] < 0)
-			sum -= frame[x];
-		else
-			sum += frame[x];
-	}
-	sum = sum/FRAME_SIZE;
-	return sum;
-}
-#endif
-
-static int cursound = -1;
-static int sampsent = 0;
-static int silencelen=0;
-static int offset=0;
-static int nosound=0;
-
-static int send_sound(void)
-{
-	short myframe[FRAME_SIZE];
-	int total = FRAME_SIZE;
-	short *frame = NULL;
-	int amt=0;
-	int res;
-	int myoff;
-	audio_buf_info abi;
-	if (cursound > -1) {
-		res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
-		if (res) {
-			ast_log(LOG_WARNING, "Unable to read output space\n");
-			return -1;
-		}
-		/* Calculate how many samples we can send, max */
-		if (total > (abi.fragments * abi.fragsize / 2)) 
-			total = abi.fragments * abi.fragsize / 2;
-		res = total;
-		if (sampsent < sounds[cursound].samplen) {
-			myoff=0;
-			while(total) {
-				amt = total;
-				if (amt > (sounds[cursound].datalen - offset)) 
-					amt = sounds[cursound].datalen - offset;
-				memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
-				total -= amt;
-				offset += amt;
-				sampsent += amt;
-				myoff += amt;
-				if (offset >= sounds[cursound].datalen)
-					offset = 0;
-			}
-			/* Set it up for silence */
-			if (sampsent >= sounds[cursound].samplen) 
-				silencelen = sounds[cursound].silencelen;
-			frame = myframe;
-		} else {
-			if (silencelen > 0) {
-				frame = silence;
-				silencelen -= res;
-			} else {
-				if (sounds[cursound].repeat) {
-					/* Start over */
-					sampsent = 0;
-					offset = 0;
-				} else {
-					cursound = -1;
-					nosound = 0;
-				}
-			}
-		}
-		if (frame)
-			res = write(sounddev, frame, res * 2);
-		if (res > 0)
-			return 0;
-		return res;
-	}
-	return 0;
-}
-
-static void *sound_thread(void *unused)
-{
-	fd_set rfds;
-	fd_set wfds;
-	int max;
-	int res;
-	char ign[4096];
-	if (read(sounddev, ign, sizeof(sounddev)) < 0)
-		ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
-	for(;;) {
-		FD_ZERO(&rfds);
-		FD_ZERO(&wfds);
-		max = sndcmd[0];
-		FD_SET(sndcmd[0], &rfds);
-		if (!oss.owner) {
-			FD_SET(sounddev, &rfds);
-			if (sounddev > max)
-				max = sounddev;
-		}
-		if (cursound > -1) {
-			FD_SET(sounddev, &wfds);
-			if (sounddev > max)
-				max = sounddev;
-		}
-		res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
-		if (res < 1) {
-			ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
-			continue;
-		}
-		if (FD_ISSET(sndcmd[0], &rfds)) {
-			read(sndcmd[0], &cursound, sizeof(cursound));
-			silencelen = 0;
-			offset = 0;
-			sampsent = 0;
-		}
-		if (FD_ISSET(sounddev, &rfds)) {
-			/* Ignore read */
-			if (read(sounddev, ign, sizeof(ign)) < 0)
-				ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
-		}
-		if (FD_ISSET(sounddev, &wfds))
-			if (send_sound())
-				ast_log(LOG_WARNING, "Failed to write sound\n");
-	}
-	/* Never reached */
-	return NULL;
-}
-
-#if 0
-static int silence_suppress(short *buf)
-{
-#define SILBUF 3
-	int loudness;
-	static int silentframes = 0;
-	static char silbuf[FRAME_SIZE * 2 * SILBUF];
-	static int silbufcnt=0;
-	if (!silencesuppression)
-		return 0;
-	loudness = calc_loudness((short *)(buf));
-	if (option_debug)
-		ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
-	if (loudness < silencethreshold) {
-		silentframes++;
-		silbufcnt++;
-		/* Keep track of the last few bits of silence so we can play
-		   them as lead-in when the time is right */
-		if (silbufcnt >= SILBUF) {
-			/* Make way for more buffer */
-			memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
-			silbufcnt--;
-		}
-		memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
-		if (silentframes > 10) {
-			/* We've had plenty of silence, so compress it now */
-			return 1;
-		}
-	} else {
-		silentframes=0;
-		/* Write any buffered silence we have, it may have something
-		   important */
-		if (silbufcnt) {
-			write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
-			silbufcnt = 0;
-		}
-	}
-	return 0;
-}
-#endif
-
-static int setformat(void)
-{
-	int fmt, desired, res, fd = sounddev;
-	static int warnedalready = 0;
-	static int warnedalready2 = 0;
-
-#if __BYTE_ORDER == __LITTLE_ENDIAN
-	fmt = AFMT_S16_LE;
-#else
-	fmt = AFMT_S16_BE;
-#endif
-
-	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
-		return -1;
-	}
-	res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
-	
-	/* Check to see if duplex set (FreeBSD Bug)*/
-	res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
-	
-	if ((fmt & DSP_CAP_DUPLEX) && !res) {
-		if (option_verbose > 1) 
-			ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
-		full_duplex = -1;
-	}
-	fmt = 0;
-	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
-		return -1;
-	}
-	/* 8000 Hz desired */
-	desired = 8000;
-	fmt = desired;
-	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
-		return -1;
-	}
-	if (fmt != desired) {
-		if (!warnedalready++)
-			ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
-	}
-#if 1
-	fmt = BUFFER_FMT;
-	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
-	if (res < 0) {
-		if (!warnedalready2++)
-			ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
-	}
-#endif
-	return 0;
-}
-
-static int soundcard_setoutput(int force)
-{
-	/* Make sure the soundcard is in output mode.  */
-	int fd = sounddev;
-	if (full_duplex || (!readmode && !force))
-		return 0;
-	readmode = 0;
-	if (force || time_has_passed()) {
-		ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-		/* Keep the same fd reserved by closing the sound device and copying stdin at the same
-		   time. */
-		/* dup2(0, sound); */ 
-		close(sounddev);
-		fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
-		if (fd < 0) {
-			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-			return -1;
-		}
-		/* dup2 will close the original and make fd be sound */
-		if (dup2(fd, sounddev) < 0) {
-			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-			return -1;
-		}
-		if (setformat()) {
-			return -1;
-		}
-		return 0;
-	}
-	return 1;
-}
-
-static int soundcard_setinput(int force)
-{
-	int fd = sounddev;
-	if (full_duplex || (readmode && !force))
-		return 0;
-	readmode = -1;
-	if (force || time_has_passed()) {
-		ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-		close(sounddev);
-		/* dup2(0, sound); */
-		fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
-		if (fd < 0) {
-			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-			return -1;
-		}
-		/* dup2 will close the original and make fd be sound */
-		if (dup2(fd, sounddev) < 0) {
-			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
-			return -1;
-		}
-		if (setformat()) {
-			return -1;
-		}
-		return 0;
-	}
-	return 1;
-}
-
-static int soundcard_init(void)
-{
-	/* Assume it's full duplex for starters */
-	int fd = open(DEV_DSP, 	O_RDWR | O_NONBLOCK);
-	if (fd < 0) {
-		ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
-		return fd;
-	}
-	gettimeofday(&lasttime, NULL);
-	sounddev = fd;
-	setformat();
-	if (!full_duplex) 
-		soundcard_setinput(1);
-	return sounddev;
-}
-
-static int oss_digit(struct ast_channel *c, char digit)
-{
-	ast_verbose( " << Console Received digit %c >> \n", digit);
-	return 0;
-}
-
-static int oss_text(struct ast_channel *c, const char *text)
-{
-	ast_verbose( " << Console Received text %s >> \n", text);
-	return 0;
-}
-
-static int oss_call(struct ast_channel *c, char *dest, int timeout)
-{
-	int res = 3;
-	struct ast_frame f = { 0, };
-	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
-	if (autoanswer) {
-		ast_verbose( " << Auto-answered >> \n" );
-		f.frametype = AST_FRAME_CONTROL;
-		f.subclass = AST_CONTROL_ANSWER;
-		ast_queue_frame(c, &f);
-	} else {
-		nosound = 1;
-		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
-		f.frametype = AST_FRAME_CONTROL;
-		f.subclass = AST_CONTROL_RINGING;
-		ast_queue_frame(c, &f);
-		write(sndcmd[1], &res, sizeof(res));
-	}
-	return 0;
-}
-
-static void answer_sound(void)
-{
-	int res;
-	nosound = 1;
-	res = 4;
-	write(sndcmd[1], &res, sizeof(res));
-	
-}
-
-static int oss_answer(struct ast_channel *c)
-{
-	ast_verbose( " << Console call has been answered >> \n");
-	answer_sound();
-	ast_setstate(c, AST_STATE_UP);
-	cursound = -1;
-	nosound=0;
-	return 0;
-}
-
-static int oss_hangup(struct ast_channel *c)
-{
-	int res = 0;
-	cursound = -1;
-	c->tech_pvt = NULL;
-	oss.owner = NULL;
-	ast_verbose( " << Hangup on console >> \n");
-	ast_mutex_lock(&usecnt_lock);
-	usecnt--;
-	ast_mutex_unlock(&usecnt_lock);
-	if (hookstate) {
-		if (autoanswer) {
-			/* Assume auto-hangup too */
-			hookstate = 0;
-		} else {
-			/* Make congestion noise */
-			res = 2;
-			write(sndcmd[1], &res, sizeof(res));
-			hookstate = 0;
-		}
-	}
-	return 0;
-}
-
-static int soundcard_writeframe(short *data)
-{	
-	/* Write an exactly FRAME_SIZE sized of frame */
-	static int bufcnt = 0;
-	static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5];
-	struct audio_buf_info info;
-	int res;
-	int fd = sounddev;
-	static int warned=0;
-	if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
-		if (!warned)
-			ast_log(LOG_WARNING, "Error reading output space\n");
-		bufcnt = buffersize;
-		warned++;
-	}
-	if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
-		/* We've run out of stuff, buffer again */
-		bufcnt = 0;
-	}
-	if (bufcnt == buffersize) {
-		/* Write sample immediately */
-		res = write(fd, ((void *)data), FRAME_SIZE * 2);
-	} else {
-		/* Copy the data into our buffer */
-		res = FRAME_SIZE * 2;
-		memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2);
-		bufcnt++;
-		if (bufcnt == buffersize) {
-			res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
-		}
-	}
-	return res;
-}
-
-
-static int oss_write(struct ast_channel *chan, struct ast_frame *f)
-{
-	int res;
-	static char sizbuf[8000];
-	static int sizpos = 0;
-	int len = sizpos;
-	int pos;
-	/* Immediately return if no sound is enabled */
-	if (nosound)
-		return 0;
-	/* Stop any currently playing sound */
-	cursound = -1;
-	if (!full_duplex && !playbackonly) {
-		/* If we're half duplex, we have to switch to read mode
-		   to honor immediate needs if necessary.  But if we are in play
-		   back only mode, then we don't switch because the console
-		   is only being used one way -- just to playback something. */
-		res = soundcard_setinput(1);
-		if (res < 0) {
-			ast_log(LOG_WARNING, "Unable to set device to input mode\n");
-			return -1;
-		}
-		return 0;
-	}
-	res = soundcard_setoutput(0);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Unable to set output device\n");
-		return -1;
-	} else if (res > 0) {
-		/* The device is still in read mode, and it's too soon to change it,
-		   so just pretend we wrote it */
-		return 0;
-	}
-	/* We have to digest the frame in 160-byte portions */
-	if (f->datalen > sizeof(sizbuf) - sizpos) {
-		ast_log(LOG_WARNING, "Frame too large\n");
-		return -1;
-	}
-	memcpy(sizbuf + sizpos, f->data, f->datalen);
-	len += f->datalen;
-	pos = 0;
-	while(len - pos > FRAME_SIZE * 2) {
-		soundcard_writeframe((short *)(sizbuf + pos));
-		pos += FRAME_SIZE * 2;
-	}
-	if (len - pos) 
-		memmove(sizbuf, sizbuf + pos, len - pos);
-	sizpos = len - pos;
-	return 0;
-}
-
-static struct ast_frame *oss_read(struct ast_channel *chan)
-{
-	static struct ast_frame f;
-	static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
-	static int readpos = 0;
-	int res;
-	
-#if 0
-	ast_log(LOG_DEBUG, "oss_read()\n");
-#endif
-		
-	f.frametype = AST_FRAME_NULL;
-	f.subclass = 0;
-	f.samples = 0;
-	f.datalen = 0;
-	f.data = NULL;
-	f.offset = 0;
-	f.src = type;
-	f.mallocd = 0;
-	f.delivery.tv_sec = 0;
-	f.delivery.tv_usec = 0;
-	
-	res = soundcard_setinput(0);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Unable to set input mode\n");
-		return NULL;
-	}
-	if (res > 0) {
-		/* Theoretically shouldn't happen, but anyway, return a NULL frame */
-		return &f;
-	}
-	res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-	if (res < 0) {
-		ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
-#if 0
-		CRASH;
-#endif		
-		return NULL;
-	}
-	readpos += res;
-	
-	if (readpos >= FRAME_SIZE * 2) {
-		/* A real frame */
-		readpos = 0;
-		if (chan->_state != AST_STATE_UP) {
-			/* Don't transmit unless it's up */
-			return &f;
-		}
-		f.frametype = AST_FRAME_VOICE;
-		f.subclass = AST_FORMAT_SLINEAR;
-		f.samples = FRAME_SIZE;
-		f.datalen = FRAME_SIZE * 2;
-		f.data = buf + AST_FRIENDLY_OFFSET;
-		f.offset = AST_FRIENDLY_OFFSET;
-		f.src = type;
-		f.mallocd = 0;
-		f.delivery.tv_sec = 0;
-		f.delivery.tv_usec = 0;
-#if 0
-		{ static int fd = -1;
-		  if (fd < 0)
-		  	fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT);
-		  write(fd, f.data, f.datalen);
-		}
-#endif		
-	}
-	return &f;
-}
-
-static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
-{
-	struct chan_oss_pvt *p = newchan->tech_pvt;
-	p->owner = newchan;
-	return 0;
-}
-
-static int oss_indicate(struct ast_channel *chan, int cond)
-{
-	int res;
-	switch(cond) {
-	case AST_CONTROL_BUSY:
-		res = 1;
-		break;
-	case AST_CONTROL_CONGESTION:
-		res = 2;
-		break;
-	case AST_CONTROL_RINGING:
-		res = 0;
-		break;
-	case -1:
-		cursound = -1;
-		return 0;
-	default:
-		ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
-		return -1;
-	}
-	if (res > -1) {
-		write(sndcmd[1], &res, sizeof(res));
-	}
-	return 0;	
-}
-
-static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
-{
-	struct ast_channel *tmp;
-	tmp = ast_channel_alloc(1);
-	if (tmp) {
-		tmp->tech = &oss_tech;
-		snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
-		tmp->type = type;
-		tmp->fds[0] = sounddev;
-		tmp->nativeformats = AST_FORMAT_SLINEAR;
-		tmp->readformat = AST_FORMAT_SLINEAR;
-		tmp->writeformat = AST_FORMAT_SLINEAR;
-		tmp->tech_pvt = p;
-		if (!ast_strlen_zero(p->context))
-			strncpy(tmp->context, p->context, sizeof(tmp->context)-1);
-		if (!ast_strlen_zero(p->exten))
-			strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1);
-		if (!ast_strlen_zero(language))
-			strncpy(tmp->language, language, sizeof(tmp->language)-1);
-		p->owner = tmp;
-		ast_setstate(tmp, state);
-		ast_mutex_lock(&usecnt_lock);
-		usecnt++;
-		ast_mutex_unlock(&usecnt_lock);
-		ast_update_use_count();
-		if (state != AST_STATE_DOWN) {
-			if (ast_pbx_start(tmp)) {
-				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
-				ast_hangup(tmp);
-				tmp = NULL;
-			}
-		}
-	}
-	return tmp;
-}
-
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
-{
-	int oldformat = format;
-	struct ast_channel *tmp;
-	format &= AST_FORMAT_SLINEAR;
-	if (!format) {
-		ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
-		return NULL;
-	}
-	if (oss.owner) {
-		ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
-		*cause = AST_CAUSE_BUSY;
-		return NULL;
-	}
-	tmp= oss_new(&oss, AST_STATE_DOWN);
-	if (!tmp) {
-		ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
-	}
-	return tmp;
-}
-
-static int console_autoanswer(int fd, int argc, char *argv[])
-{
-	if ((argc != 1) && (argc != 2))
-		return RESULT_SHOWUSAGE;
-	if (argc == 1) {
-		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
-		return RESULT_SUCCESS;
-	} else {
-		if (!strcasecmp(argv[1], "on"))
-			autoanswer = -1;
-		else if (!strcasecmp(argv[1], "off"))
-			autoanswer = 0;
-		else
-			return RESULT_SHOWUSAGE;
-	}
-	return RESULT_SUCCESS;
-}
-
-static char *autoanswer_complete(char *line, char *word, int pos, int state)
-{
-#ifndef MIN
-#define MIN(a,b) ((a) < (b) ? (a) : (b))
-#endif
-	switch(state) {
-	case 0:
-		if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
-			return strdup("on");
-	case 1:
-		if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
-			return strdup("off");
-	default:
-		return NULL;
-	}
-	return NULL;
-}
-
-static char autoanswer_usage[] =
-"Usage: autoanswer [on|off]\n"
-"       Enables or disables autoanswer feature.  If used without\n"
-"       argument, displays the current on/off status of autoanswer.\n"
-"       The default value of autoanswer is in 'oss.conf'.\n";
-
-static int console_answer(int fd, int argc, char *argv[])
-{
-	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
-	if (argc != 1)
-		return RESULT_SHOWUSAGE;
-	if (!oss.owner) {
-		ast_cli(fd, "No one is calling us\n");
-		return RESULT_FAILURE;
-	}
-	hookstate = 1;
-	cursound = -1;
-	ast_queue_frame(oss.owner, &f);
-	answer_sound();
-	return RESULT_SUCCESS;
-}
-
-static char sendtext_usage[] =
-"Usage: send text <message>\n"
-"       Sends a text message for display on the remote terminal.\n";
-
-static int console_sendtext(int fd, int argc, char *argv[])
-{
-	int tmparg = 2;
-	char text2send[256] = "";
-	struct ast_frame f = { 0, };
-	if (argc < 2)
-		return RESULT_SHOWUSAGE;
-	if (!oss.owner) {
-		ast_cli(fd, "No one is calling us\n");
-		return RESULT_FAILURE;
-	}
-	if (!ast_strlen_zero(text2send))
-		ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
-	text2send[0] = '\0';
-	while(tmparg < argc) {
-		strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
-		strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
-	}
-	if (!ast_strlen_zero(text2send)) {
-		f.frametype = AST_FRAME_TEXT;
-		f.subclass = 0;
-		f.data = text2send;
-		f.datalen = strlen(text2send);
-		ast_queue_frame(oss.owner, &f);
-	}
-	return RESULT_SUCCESS;
-}
-
-static char answer_usage[] =
-"Usage: answer\n"
-"       Answers an incoming call on the console (OSS) channel.\n";
-
-static int console_hangup(int fd, int argc, char *argv[])
-{
-	if (argc != 1)
-		return RESULT_SHOWUSAGE;
-	cursound = -1;
-	if (!oss.owner && !hookstate) {
-		ast_cli(fd, "No call to hangup up\n");
-		return RESULT_FAILURE;
-	}
-	hookstate = 0;
-	if (oss.owner) {
-		ast_queue_hangup(oss.owner);
-	}
-	return RESULT_SUCCESS;
-}
-
-static int console_flash(int fd, int argc, char *argv[])
-{
-	struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
-	if (argc != 1)
-		return RESULT_SHOWUSAGE;
-	cursound = -1;
-	if (!oss.owner) {
-		ast_cli(fd, "No call to flash\n");
-		return RESULT_FAILURE;
-	}
-	hookstate = 0;
-	if (oss.owner) {
-		ast_queue_frame(oss.owner, &f);
-	}
-	return RESULT_SUCCESS;
-}
-
-static char hangup_usage[] =
-"Usage: hangup\n"
-"       Hangs up any call currently placed on the console.\n";
-
-
-static char flash_usage[] =
-"Usage: flash\n"
-"       Flashes the call currently placed on the console.\n";
-
-static int console_dial(int fd, int argc, char *argv[])
-{
-	char tmp[256], *tmp2;
-	char *mye, *myc;
-	int x;
-	struct ast_frame f = { AST_FRAME_DTMF, 0 };
-	if ((argc != 1) && (argc != 2))
-		return RESULT_SHOWUSAGE;
-	if (oss.owner) {
-		if (argc == 2) {
-			for (x=0;x<strlen(argv[1]);x++) {
-				f.subclass = argv[1][x];
-				ast_queue_frame(oss.owner, &f);
-			}
-		} else {
-			ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
-			return RESULT_FAILURE;
-		}
-		return RESULT_SUCCESS;
-	}
-	mye = exten;
-	myc = context;
-	if (argc == 2) {
-		char *stringp=NULL;
-		strncpy(tmp, argv[1], sizeof(tmp)-1);
-		stringp=tmp;
-		strsep(&stringp, "@");
-		tmp2 = strsep(&stringp, "@");
-		if (!ast_strlen_zero(tmp))
-			mye = tmp;
-		if (!ast_strlen_zero(tmp2))
-			myc = tmp2;
-	}
-	if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
-		strncpy(oss.exten, mye, sizeof(oss.exten)-1);
-		strncpy(oss.context, myc, sizeof(oss.context)-1);
-		hookstate = 1;
-		oss_new(&oss, AST_STATE_RINGING);
-	} else
-		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
-	return RESULT_SUCCESS;
-}
-
-static char dial_usage[] =
-"Usage: dial [extension[@context]]\n"
-"       Dials a given extensison (and context if specified)\n";
-
-static int console_transfer(int fd, int argc, char *argv[])
-{
-	char tmp[256];
-	char *context;
-	if (argc != 2)
-		return RESULT_SHOWUSAGE;
-	if (oss.owner && ast_bridged_channel(oss.owner)) {
-		strncpy(tmp, argv[1], sizeof(tmp) - 1);
-		context = strchr(tmp, '@');
-		if (context) {
-			*context = '\0';
-			context++;
-		} else
-			context = oss.owner->context;
-		if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
-			ast_cli(fd, "Whee, transferring %s to %s@%s.\n", 
-					ast_bridged_channel(oss.owner)->name, tmp, context);
-			if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
-				ast_cli(fd, "Failed to transfer :(\n");
-		} else {
-			ast_cli(fd, "No such extension exists\n");
-		}
-	} else {
-		ast_cli(fd, "There is no call to transfer\n");
-	}
-	return RESULT_SUCCESS;
-}
-
-static char transfer_usage[] =
-"Usage: transfer <extension>[@context]\n"
-"       Transfers the currently connected call to the given extension (and\n"
-"context if specified)\n";
-
-static struct ast_cli_entry myclis[] = {
-	{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
-	{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
-	{ { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage },
-	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
-	{ { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_usage },
-	{ { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
-	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
-};
-
-int load_module()
-{
-	int res;
-	int x;
-	struct ast_config *cfg;
-	struct ast_variable *v;
-	res = pipe(sndcmd);
-	if (res) {
-		ast_log(LOG_ERROR, "Unable to create pipe\n");
-		return -1;
-	}
-	res = soundcard_init();
-	if (res < 0) {
-		if (option_verbose > 1) {
-			ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
-			ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
-		}
-		return 0;
-	}
-	if (!full_duplex)
-		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
-	res = ast_channel_register(&oss_tech);
-	if (res < 0) {
-		ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
-		return -1;
-	}
-	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
-		ast_cli_register(myclis + x);
-	if ((cfg = ast_config_load(config))) {
-		v = ast_variable_browse(cfg, "general");
-		while(v) {
-			if (!strcasecmp(v->name, "autoanswer"))
-				autoanswer = ast_true(v->value);
-			else if (!strcasecmp(v->name, "silencesuppression"))
-				silencesuppression = ast_true(v->value);
-			else if (!strcasecmp(v->name, "silencethreshold"))
-				silencethreshold = atoi(v->value);
-			else if (!strcasecmp(v->name, "context"))
-				strncpy(context, v->value, sizeof(context)-1);
-			else if (!strcasecmp(v->name, "language"))
-				strncpy(language, v->value, sizeof(language)-1);
-			else if (!strcasecmp(v->name, "extension"))
-				strncpy(exten, v->value, sizeof(exten)-1);
-			else if (!strcasecmp(v->name, "playbackonly"))
-				playbackonly = ast_true(v->value);
-			v=v->next;
-		}
-		ast_config_destroy(cfg);
-	}
-	ast_pthread_create(&sthread, NULL, sound_thread, NULL);
-	return 0;
-}
-
-
-
-int unload_module()
-{
-	int x;
-
-	ast_channel_unregister(&oss_tech);
-	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
-		ast_cli_unregister(myclis + x);
-	close(sounddev);
-	if (sndcmd[0] > 0) {
-		close(sndcmd[0]);
-		close(sndcmd[1]);
-	}
-	if (oss.owner)
-		ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
-	if (oss.owner)
-		return -1;
-	return 0;
-}
-
-char *description()
-{
-	return (char *) desc;
-}
-
-int usecount()
-{
-	return usecnt;
-}
-
-char *key()
-{
-	return ASTERISK_GPL_KEY;
-}