diff --git a/channels/Makefile b/channels/Makefile
index 417046177ee41581faade1663acddbd5332cc303..8425ff61c3a6a66a25ababd36a3d6ef49ddabdd1 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 88fd428ee599f777bea92237b2d7e06afefecf03..0000000000000000000000000000000000000000
--- 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 c1613d3b66c37d26f6e9e2455ff391ca26316818..0000000000000000000000000000000000000000
--- 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;
-}