Ignore some files generated from 1.6.x/trunk build

[asterisk-bristuff.git] / channels / chan_oss.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
 * note-this code best seen with ts=8 (8-spaces tabs) in the editor
 *
 * 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.
 */

/*! \file
 *
 * \brief Channel driver for OSS sound cards
 *
 * \author Mark Spencer <markster@digium.com>
 * \author Luigi Rizzo
 *
 * \par See also
 * \arg \ref Config_oss
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
        <depend>ossaudio</depend>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <stdio.h>
#include <ctype.h>
#include <math.h>
#include <string.h>
#include <unistd.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <sys/time.h>
#include <stdlib.h>
#include <errno.h>

#ifdef __linux
#include <linux/soundcard.h>
#elif defined(__FreeBSD__)
#include <sys/soundcard.h>
#else
#include <soundcard.h>
#endif

#include "asterisk/lock.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/callerid.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 "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"

/* ringtones we use */
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"

/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
        .flags = 0,
        .max_size = -1,
        .resync_threshold = -1,
        .impl = "",
};
static struct ast_jb_conf global_jbconf;

/*
 * Basic mode of operation:
 *
 * we have one keyboard (which receives commands from the keyboard)
 * and multiple headset's connected to audio cards.
 * Cards/Headsets are named as the sections of oss.conf.
 * The section called [general] contains the default parameters.
 *
 * At any time, the keyboard is attached to one card, and you
 * can switch among them using the command 'console foo'
 * where 'foo' is the name of the card you want.
 *
 * oss.conf parameters are
START_CONFIG

[general]
    ; General config options, with default values shown.
    ; You should use one section per device, with [general] being used
    ; for the first device and also as a template for other devices.
    ;
    ; All but 'debug' can go also in the device-specific sections.
    ;
    ; debug = 0x0               ; misc debug flags, default is 0

    ; Set the device to use for I/O
    ; device = /dev/dsp

    ; Optional mixer command to run upon startup (e.g. to set
    ; volume levels, mutes, etc.
    ; mixer =

    ; Software mic volume booster (or attenuator), useful for sound
    ; cards or microphones with poor sensitivity. The volume level
    ; is in dB, ranging from -20.0 to +20.0
    ; boost = n                 ; mic volume boost in dB

    ; Set the callerid for outgoing calls
    ; callerid = John Doe <555-1234>

    ; autoanswer = no           ; no autoanswer on call
    ; autohangup = yes          ; hangup when other party closes
    ; extension = s             ; default extension to call
    ; context = default         ; default context for outgoing calls
    ; language = ""             ; default language

    ; Default Music on Hold class to use when this channel is placed on hold in
    ; the case that the music class is not set on the channel with
    ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
    ; putting this one on hold did not suggest a class to use.
    ;
    ; mohinterpret=default

    ; If you set overridecontext to 'yes', then the whole dial string
    ; will be interpreted as an extension, which is extremely useful
    ; to dial SIP, IAX and other extensions which use the '@' character.
    ; The default is 'no' just for backward compatibility, but the
    ; suggestion is to change it.
    ; overridecontext = no      ; if 'no', the last @ will start the context
                                ; if 'yes' the whole string is an extension.

    ; low level device parameters in case you have problems with the
    ; device driver on your operating system. You should not touch these
    ; unless you know what you are doing.
    ; queuesize = 10            ; frames in device driver
    ; frags = 8                 ; argument to SETFRAGMENT

    ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
    ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of an
                                  ; OSS channel. Defaults to "no". An enabled jitterbuffer will
                                  ; be used only if the sending side can create and the receiving
                                  ; side can not accept jitter. The OSS channel can't accept jitter,
                                  ; thus an enabled jitterbuffer on the receive OSS side will always
                                  ; be used if the sending side can create jitter.

    ; jbmaxsize = 200             ; Max length of the jitterbuffer in milliseconds.

    ; jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                                  ; resynchronized. Useful to improve the quality of the voice, with
                                  ; big jumps in/broken timestamps, usualy sent from exotic devices
                                  ; and programs. Defaults to 1000.

    ; jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of an OSS
                                  ; channel. Two implementations are currenlty available - "fixed"
                                  ; (with size always equals to jbmax-size) and "adaptive" (with
                                  ; variable size, actually the new jb of IAX2). Defaults to fixed.

    ; jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
    ;-----------------------------------------------------------------------------------

[card1]
    ; device = /dev/dsp1        ; alternate device

END_CONFIG

.. and so on for the other cards.

 */

/*
 * Helper macros to parse config arguments. They will go in a common
 * header file if their usage is globally accepted. In the meantime,
 * we define them here. Typical usage is as below.
 * Remember to open a block right before M_START (as it declares
 * some variables) and use the M_* macros WITHOUT A SEMICOLON:
 *
 *      {
 *              M_START(v->name, v->value) 
 *
 *              M_BOOL("dothis", x->flag1)
 *              M_STR("name", x->somestring)
 *              M_F("bar", some_c_code)
 *              M_END(some_final_statement)
 *              ... other code in the block
 *      }
 *
 * XXX NOTE these macros should NOT be replicated in other parts of asterisk. 
 * Likely we will come up with a better way of doing config file parsing.
 */
#define M_START(var, val) \
        char *__s = var; char *__val = val;
#define M_END(x)   x;
#define M_F(tag, f)                     if (!strcasecmp((__s), tag)) { f; } else
#define M_BOOL(tag, dst)        M_F(tag, (dst) = ast_true(__val) )
#define M_UINT(tag, dst)        M_F(tag, (dst) = strtoul(__val, NULL, 0) )
#define M_STR(tag, dst)         M_F(tag, ast_copy_string(dst, __val, sizeof(dst)))

/*
 * The following parameters are used in the driver:
 *
 *  FRAME_SIZE  the size of an audio frame, in samples.
 *              160 is used almost universally, so you should not change it.
 *
 *  FRAGS       the argument for the SETFRAGMENT ioctl.
 *              Overridden by the 'frags' parameter in oss.conf
 *
 *              Bits 0-7 are the base-2 log of the device's block size,
 *              bits 16-31 are the number of blocks in the driver's queue.
 *              There are a lot of differences in the way this parameter
 *              is supported by different drivers, so you may need to
 *              experiment a bit with the value.
 *              A good default for linux is 30 blocks of 64 bytes, which
 *              results in 6 frames of 320 bytes (160 samples).
 *              FreeBSD works decently with blocks of 256 or 512 bytes,
 *              leaving the number unspecified.
 *              Note that this only refers to the device buffer size,
 *              this module will then try to keep the lenght of audio
 *              buffered within small constraints.
 *
 *  QUEUE_SIZE  The max number of blocks actually allowed in the device
 *              driver's buffer, irrespective of the available number.
 *              Overridden by the 'queuesize' parameter in oss.conf
 *
 *              Should be >=2, and at most as large as the hw queue above
 *              (otherwise it will never be full).
 */

#define FRAME_SIZE      160
#define QUEUE_SIZE      10

#if defined(__FreeBSD__)
#define FRAGS   0x8
#else
#define FRAGS   ( ( (6 * 5) << 16 ) | 0x6 )
#endif

/*
 * XXX text message sizes are probably 256 chars, but i am
 * not sure if there is a suitable definition anywhere.
 */
#define TEXT_SIZE       256

#if 0
#define TRYOPEN 1                               /* try to open on startup */
#endif
#define O_CLOSE 0x444                   /* special 'close' mode for device */
/* Which device to use */
#if defined( __OpenBSD__ ) || defined( __NetBSD__ )
#define DEV_DSP "/dev/audio"
#else
#define DEV_DSP "/dev/dsp"
#endif

#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif

static char *config = "oss.conf";       /* default config file */

static int oss_debug;

/*
 * Each sound is made of 'datalen' samples of sound, repeated as needed to
 * generate 'samplen' samples of data, then followed by 'silencelen' samples
 * of silence. The loop is repeated if 'repeat' is set.
 */
struct sound {
        int ind;
        char *desc;
        short *data;
        int datalen;
        int samplen;
        int silencelen;
        int repeat;
};

static struct sound sounds[] = {
        { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
        { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 },
        { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 },
        { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 },
        { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 },
        { -1, NULL, 0, 0, 0, 0 },       /* end marker */
};


/*
 * descriptor for one of our channels.
 * There is one used for 'default' values (from the [general] entry in
 * the configuration file), and then one instance for each device
 * (the default is cloned from [general], others are only created
 * if the relevant section exists).
 */
struct chan_oss_pvt {
        struct chan_oss_pvt *next;

        char *name;
        /*
         * cursound indicates which in struct sound we play. -1 means nothing,
         * any other value is a valid sound, in which case sampsent indicates
         * the next sample to send in [0..samplen + silencelen]
         * nosound is set to disable the audio data from the channel
         * (so we can play the tones etc.).
         */
        int sndcmd[2];                          /* Sound command pipe */
        int cursound;                           /* index of sound to send */
        int sampsent;                           /* # of sound samples sent  */
        int nosound;                            /* set to block audio from the PBX */

        int total_blocks;                       /* total blocks in the output device */
        int sounddev;
        enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
        int autoanswer;
        int autohangup;
        int hookstate;
        char *mixer_cmd;                        /* initial command to issue to the mixer */
        unsigned int queuesize;         /* max fragments in queue */
        unsigned int frags;                     /* parameter for SETFRAGMENT */

        int warned;                                     /* various flags used for warnings */
#define WARN_used_blocks        1
#define WARN_speed              2
#define WARN_frag               4
        int w_errors;                           /* overfull in the write path */
        struct timeval lastopen;

        int overridecontext;
        int mute;

        /* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must
         * be representable in 16 bits to avoid overflows.
         */
#define BOOST_SCALE     (1<<9)
#define BOOST_MAX       40                      /* slightly less than 7 bits */
        int boost;                                      /* input boost, scaled by BOOST_SCALE */
        char device[64];                        /* device to open */

        pthread_t sthread;

        struct ast_channel *owner;
        char ext[AST_MAX_EXTENSION];
        char ctx[AST_MAX_CONTEXT];
        char language[MAX_LANGUAGE];
        char cid_name[256];                     /*XXX */
        char cid_num[256];                      /*XXX */
        char mohinterpret[MAX_MUSICCLASS];

        /* buffers used in oss_write */
        char oss_write_buf[FRAME_SIZE * 2];
        int oss_write_dst;
        /* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers
         * plus enough room for a full frame
         */
        char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
        int readpos;                            /* read position above */
        struct ast_frame read_f;        /* returned by oss_read */
};

static struct chan_oss_pvt oss_default = {
        .cursound = -1,
        .sounddev = -1,
        .duplex = M_UNSET,                      /* XXX check this */
        .autoanswer = 1,
        .autohangup = 1,
        .queuesize = QUEUE_SIZE,
        .frags = FRAGS,
        .ext = "s",
        .ctx = "default",
        .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */
        .lastopen = { 0, 0 },
        .boost = BOOST_SCALE,
};

static char *oss_active;         /* the active device */

static int setformat(struct chan_oss_pvt *o, int mode);

static struct ast_channel *oss_request(const char *type, int format, void *data
, int *cause);
static int oss_digit_begin(struct ast_channel *c, char digit);
static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration);
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, const void *data, size_t datalen);
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static char tdesc[] = "OSS Console Channel Driver";

static const struct ast_channel_tech oss_tech = {
        .type = "Console",
        .description = tdesc,
        .capabilities = AST_FORMAT_SLINEAR,
        .requester = oss_request,
        .send_digit_begin = oss_digit_begin,
        .send_digit_end = oss_digit_end,
        .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,
};

/*
 * returns a pointer to the descriptor with the given name
 */
static struct chan_oss_pvt *find_desc(char *dev)
{
        struct chan_oss_pvt *o = NULL;

        if (!dev)
                ast_log(LOG_WARNING, "null dev\n");

        for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next);

        if (!o)
                ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--");

        return o;
}

/*
 * split a string in extension-context, returns pointers to malloc'ed
 * strings.
 * If we do not have 'overridecontext' then the last @ is considered as
 * a context separator, and the context is overridden.
 * This is usually not very necessary as you can play with the dialplan,
 * and it is nice not to need it because you have '@' in SIP addresses.
 * Return value is the buffer address.
 */
static char *ast_ext_ctx(const char *src, char **ext, char **ctx)
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (ext == NULL || ctx == NULL)
                return NULL;                    /* error */

        *ext = *ctx = NULL;

        if (src && *src != '\0')
                *ext = ast_strdup(src);

        if (*ext == NULL)
                return NULL;

        if (!o->overridecontext) {
                /* parse from the right */
                *ctx = strrchr(*ext, '@');
                if (*ctx)
                        *(*ctx)++ = '\0';
        }

        return *ext;
}

/*
 * Returns the number of blocks used in the audio output channel
 */
static int used_blocks(struct chan_oss_pvt *o)
{
        struct audio_buf_info info;

        if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) {
                if (!(o->warned & WARN_used_blocks)) {
                        ast_log(LOG_WARNING, "Error reading output space\n");
                        o->warned |= WARN_used_blocks;
                }
                return 1;
        }

        if (o->total_blocks == 0) {
                if (0)                                  /* debugging */
                        ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments);
                o->total_blocks = info.fragments;
        }

        return o->total_blocks - info.fragments;
}

/* Write an exactly FRAME_SIZE sized frame */
static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
{
        int res;

        if (o->sounddev < 0)
                setformat(o, O_RDWR);
        if (o->sounddev < 0)
                return 0;                               /* not fatal */
        /*
         * Nothing complex to manage the audio device queue.
         * If the buffer is full just drop the extra, otherwise write.
         * XXX in some cases it might be useful to write anyways after
         * a number of failures, to restart the output chain.
         */
        res = used_blocks(o);
        if (res > o->queuesize) {       /* no room to write a block */
                if (o->w_errors++ == 0 && (oss_debug & 0x4))
                        ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors);
                return 0;
        }
        o->w_errors = 0;
        return write(o->sounddev, ((void *) data), FRAME_SIZE * 2);
}

/*
 * Handler for 'sound writable' events from the sound thread.
 * Builds a frame from the high level description of the sounds,
 * and passes it to the audio device.
 * The actual sound is made of 1 or more sequences of sound samples
 * (s->datalen, repeated to make s->samplen samples) followed by
 * s->silencelen samples of silence. The position in the sequence is stored
 * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen.
 * In case we fail to write a frame, don't update o->sampsent.
 */
static void send_sound(struct chan_oss_pvt *o)
{
        short myframe[FRAME_SIZE];
        int ofs, l, start;
        int l_sampsent = o->sampsent;
        struct sound *s;

        if (o->cursound < 0)            /* no sound to send */
                return;

        s = &sounds[o->cursound];

        for (ofs = 0; ofs < FRAME_SIZE; ofs += l) {
                l = s->samplen - l_sampsent;    /* # of available samples */
                if (l > 0) {
                        start = l_sampsent % s->datalen;        /* source offset */
                        if (l > FRAME_SIZE - ofs)       /* don't overflow the frame */
                                l = FRAME_SIZE - ofs;
                        if (l > s->datalen - start)     /* don't overflow the source */
                                l = s->datalen - start;
                        bcopy(s->data + start, myframe + ofs, l * 2);
                        if (0)
                                ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", l_sampsent, l, s->samplen, ofs);
                        l_sampsent += l;
                } else {                                /* end of samples, maybe some silence */
                        static const short silence[FRAME_SIZE] = { 0, };

                        l += s->silencelen;
                        if (l > 0) {
                                if (l > FRAME_SIZE - ofs)
                                        l = FRAME_SIZE - ofs;
                                bcopy(silence, myframe + ofs, l * 2);
                                l_sampsent += l;
                        } else {                        /* silence is over, restart sound if loop */
                                if (s->repeat == 0) {   /* last block */
                                        o->cursound = -1;
                                        o->nosound = 0; /* allow audio data */
                                        if (ofs < FRAME_SIZE)   /* pad with silence */
                                                bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs) * 2);
                                }
                                l_sampsent = 0;
                        }
                }
        }
        l = soundcard_writeframe(o, myframe);
        if (l > 0)
                o->sampsent = l_sampsent;       /* update status */
}

static void *sound_thread(void *arg)
{
        char ign[4096];
        struct chan_oss_pvt *o = (struct chan_oss_pvt *) arg;

        /*
         * Just in case, kick the driver by trying to read from it.
         * Ignore errors - this read is almost guaranteed to fail.
         */
        read(o->sounddev, ign, sizeof(ign));
        for (;;) {
                fd_set rfds, wfds;
                int maxfd, res;

                FD_ZERO(&rfds);
                FD_ZERO(&wfds);
                FD_SET(o->sndcmd[0], &rfds);
                maxfd = o->sndcmd[0];   /* pipe from the main process */
                if (o->cursound > -1 && o->sounddev < 0)
                        setformat(o, O_RDWR);   /* need the channel, try to reopen */
                else if (o->cursound == -1 && o->owner == NULL)
                        setformat(o, O_CLOSE);  /* can close */
                if (o->sounddev > -1) {
                        if (!o->owner) {        /* no one owns the audio, so we must drain it */
                                FD_SET(o->sounddev, &rfds);
                                maxfd = MAX(o->sounddev, maxfd);
                        }
                        if (o->cursound > -1) {
                                FD_SET(o->sounddev, &wfds);
                                maxfd = MAX(o->sounddev, maxfd);
                        }
                }
                /* ast_select emulates linux behaviour in terms of timeout handling */
                res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL);
                if (res < 1) {
                        ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
                        sleep(1);
                        continue;
                }
                if (FD_ISSET(o->sndcmd[0], &rfds)) {
                        /* read which sound to play from the pipe */
                        int i, what = -1;

                        read(o->sndcmd[0], &what, sizeof(what));
                        for (i = 0; sounds[i].ind != -1; i++) {
                                if (sounds[i].ind == what) {
                                        o->cursound = i;
                                        o->sampsent = 0;
                                        o->nosound = 1; /* block audio from pbx */
                                        break;
                                }
                        }
                        if (sounds[i].ind == -1)
                                ast_log(LOG_WARNING, "invalid sound index: %d\n", what);
                }
                if (o->sounddev > -1) {
                        if (FD_ISSET(o->sounddev, &rfds))       /* read and ignore errors */
                                read(o->sounddev, ign, sizeof(ign));
                        if (FD_ISSET(o->sounddev, &wfds))
                                send_sound(o);
                }
        }
        return NULL;                            /* Never reached */
}

/*
 * reset and close the device if opened,
 * then open and initialize it in the desired mode,
 * trigger reads and writes so we can start using it.
 */
static int setformat(struct chan_oss_pvt *o, int mode)
{
        int fmt, desired, res, fd;

        if (o->sounddev >= 0) {
                ioctl(o->sounddev, SNDCTL_DSP_RESET, 0);
                close(o->sounddev);
                o->duplex = M_UNSET;
                o->sounddev = -1;
        }
        if (mode == O_CLOSE)            /* we are done */
                return 0;
        if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000)
                return -1;                              /* don't open too often */
        o->lastopen = ast_tvnow();
        fd = o->sounddev = open(o->device, mode | O_NONBLOCK);
        if (fd < 0) {
                ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno));
                return -1;
        }
        if (o->owner)
                o->owner->fds[0] = fd;

#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;
        }
        switch (mode) {
                case O_RDWR:
                        res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
                        /* Check to see if duplex set (FreeBSD Bug) */
                        res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
                        if (res == 0 && (fmt & DSP_CAP_DUPLEX)) {
                                if (option_verbose > 1)
                                        ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
                                o->duplex = M_FULL;
                        };
                        break;
                case O_WRONLY:
                        o->duplex = M_WRITE;
                        break;
                case O_RDONLY:
                        o->duplex = M_READ;
                        break;
        }

        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;
        }
        fmt = desired = DEFAULT_SAMPLE_RATE;    /* 8000 Hz 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 (!(o->warned & WARN_speed)) {
                        ast_log(LOG_WARNING,
                            "Requested %d Hz, got %d Hz -- sound may be choppy\n",
                            desired, fmt);
                        o->warned |= WARN_speed;
                }
        }
        /*
         * on Freebsd, SETFRAGMENT does not work very well on some cards.
         * Default to use 256 bytes, let the user override
         */
        if (o->frags) {
                fmt = o->frags;
                res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
                if (res < 0) {
                        if (!(o->warned & WARN_frag)) {
                                ast_log(LOG_WARNING,
                                        "Unable to set fragment size -- sound may be choppy\n");
                                o->warned |= WARN_frag;
                        }
                }
        }
        /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */
        res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT;
        res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res);
        /* it may fail if we are in half duplex, never mind */
        return 0;
}

/*
 * some of the standard methods supported by channels.
 */
static int oss_digit_begin(struct ast_channel *c, char digit)
{
        return 0;
}

static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration)
{
        /* no better use for received digits than print them */
        ast_verbose(" << Console Received digit %c of duration %u ms >> \n", 
                digit, duration);
        return 0;
}

static int oss_text(struct ast_channel *c, const char *text)
{
        /* print received messages */
        ast_verbose(" << Console Received text %s >> \n", text);
        return 0;
}

/* Play ringtone 'x' on device 'o' */
static void ring(struct chan_oss_pvt *o, int x)
{
        write(o->sndcmd[1], &x, sizeof(x));
}


/*
 * handler for incoming calls. Either autoanswer, or start ringing
 */
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
        struct chan_oss_pvt *o = c->tech_pvt;
        struct ast_frame f = { 0, };

        ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n", dest, c->cid.cid_dnid, c->cid.cid_rdnis, c->cid.cid_name, c->cid.cid_num);
        if (o->autoanswer) {
                ast_verbose(" << Auto-answered >> \n");
                f.frametype = AST_FRAME_CONTROL;
                f.subclass = AST_CONTROL_ANSWER;
                ast_queue_frame(c, &f);
        } else {
                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);
                ring(o, AST_CONTROL_RING);
        }
        return 0;
}

/*
 * remote side answered the phone
 */
static int oss_answer(struct ast_channel *c)
{
        struct chan_oss_pvt *o = c->tech_pvt;

        ast_verbose(" << Console call has been answered >> \n");
#if 0
        /* play an answer tone (XXX do we really need it ?) */
        ring(o, AST_CONTROL_ANSWER);
#endif
        ast_setstate(c, AST_STATE_UP);
        o->cursound = -1;
        o->nosound = 0;
        return 0;
}

static int oss_hangup(struct ast_channel *c)
{
        struct chan_oss_pvt *o = c->tech_pvt;

        o->cursound = -1;
        o->nosound = 0;
        c->tech_pvt = NULL;
        o->owner = NULL;
        ast_verbose(" << Hangup on console >> \n");
        ast_module_unref(ast_module_info->self);
        if (o->hookstate) {
                if (o->autoanswer || o->autohangup) {
                        /* Assume auto-hangup too */
                        o->hookstate = 0;
                        setformat(o, O_CLOSE);
                } else {
                        /* Make congestion noise */
                        ring(o, AST_CONTROL_CONGESTION);
                }
        }
        return 0;
}

/* used for data coming from the network */
static int oss_write(struct ast_channel *c, struct ast_frame *f)
{
        int src;
        struct chan_oss_pvt *o = c->tech_pvt;

        /* Immediately return if no sound is enabled */
        if (o->nosound)
                return 0;
        /* Stop any currently playing sound */
        o->cursound = -1;
        /*
         * we could receive a block which is not a multiple of our
         * FRAME_SIZE, so buffer it locally and write to the device
         * in FRAME_SIZE chunks.
         * Keep the residue stored for future use.
         */
        src = 0;                                        /* read position into f->data */
        while (src < f->datalen) {
                /* Compute spare room in the buffer */
                int l = sizeof(o->oss_write_buf) - o->oss_write_dst;

                if (f->datalen - src >= l) {    /* enough to fill a frame */
                        memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
                        soundcard_writeframe(o, (short *) o->oss_write_buf);
                        src += l;
                        o->oss_write_dst = 0;
                } else {                                /* copy residue */
                        l = f->datalen - src;
                        memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l);
                        src += l;                       /* but really, we are done */
                        o->oss_write_dst += l;
                }
        }
        return 0;
}

static struct ast_frame *oss_read(struct ast_channel *c)
{
        int res;
        struct chan_oss_pvt *o = c->tech_pvt;
        struct ast_frame *f = &o->read_f;

        /* XXX can be simplified returning &ast_null_frame */
        /* prepare a NULL frame in case we don't have enough data to return */
        bzero(f, sizeof(struct ast_frame));
        f->frametype = AST_FRAME_NULL;
        f->src = oss_tech.type;

        res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos);
        if (res < 0)                            /* audio data not ready, return a NULL frame */
                return f;

        o->readpos += res;
        if (o->readpos < sizeof(o->oss_read_buf))       /* not enough samples */
                return f;

        if (o->mute)
                return f;

        o->readpos = AST_FRIENDLY_OFFSET;       /* reset read pointer for next frame */
        if (c->_state != AST_STATE_UP)  /* drop data if frame is not up */
                return f;
        /* ok we can build and deliver the frame to the caller */
        f->frametype = AST_FRAME_VOICE;
        f->subclass = AST_FORMAT_SLINEAR;
        f->samples = FRAME_SIZE;
        f->datalen = FRAME_SIZE * 2;
        f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET;
        if (o->boost != BOOST_SCALE) {  /* scale and clip values */
                int i, x;
                int16_t *p = (int16_t *) f->data;
                for (i = 0; i < f->samples; i++) {
                        x = (p[i] * o->boost) / BOOST_SCALE;
                        if (x > 32767)
                                x = 32767;
                        else if (x < -32768)
                                x = -32768;
                        p[i] = x;
                }
        }

        f->offset = AST_FRIENDLY_OFFSET;
        return f;
}

static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
        struct chan_oss_pvt *o = newchan->tech_pvt;
        o->owner = newchan;
        return 0;
}

static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen)
{
        struct chan_oss_pvt *o = c->tech_pvt;
        int res = -1;

        switch (cond) {
        case AST_CONTROL_BUSY:
        case AST_CONTROL_CONGESTION:
        case AST_CONTROL_RINGING:
                        res = cond;
                        break;
                        
        case -1:
                o->cursound = -1;
                o->nosound = 0;         /* when cursound is -1 nosound must be 0 */
                return 0;
                
        case AST_CONTROL_VIDUPDATE:
                res = -1;
                break;
        case AST_CONTROL_HOLD:
                ast_verbose(" << Console Has Been Placed on Hold >> \n");
                ast_moh_start(c, data, o->mohinterpret);
                        break;
        case AST_CONTROL_UNHOLD:
                ast_verbose(" << Console Has Been Retrieved from Hold >> \n");
                ast_moh_stop(c);
                break;
        case AST_CONTROL_SRCUPDATE:
                break;
        default:
                ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name);
                return -1;
        }

        if (res > -1)
                ring(o, res);

        return 0;
}

/*
 * allocate a new channel.
 */
static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state)
{
        struct ast_channel *c;

        c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, 0, "Console/%s", o->device + 5);
        if (c == NULL)
                return NULL;
        c->tech = &oss_tech;
        if (o->sounddev < 0)
                setformat(o, O_RDWR);
        c->fds[0] = o->sounddev;        /* -1 if device closed, override later */
        c->nativeformats = AST_FORMAT_SLINEAR;
        c->readformat = AST_FORMAT_SLINEAR;
        c->writeformat = AST_FORMAT_SLINEAR;
        c->tech_pvt = o;

        if (!ast_strlen_zero(o->language))
                ast_string_field_set(c, language, o->language);
        /* Don't use ast_set_callerid() here because it will
         * generate a needless NewCallerID event */
        c->cid.cid_ani = ast_strdup(o->cid_num);
        if (!ast_strlen_zero(ext))
                c->cid.cid_dnid = ast_strdup(ext);

        o->owner = c;
        ast_module_ref(ast_module_info->self);
        ast_jb_configure(c, &global_jbconf);
        if (state != AST_STATE_DOWN) {
                if (ast_pbx_start(c)) {
                        ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name);
                        ast_hangup(c);
                        o->owner = c = NULL;
                        /* XXX what about the channel itself ? */
                        /* XXX what about usecnt ? */
                }
        }

        return c;
}

static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
{
        struct ast_channel *c;
        struct chan_oss_pvt *o = find_desc(data);

        ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, (char *) data);
        if (o == NULL) {
                ast_log(LOG_NOTICE, "Device %s not found\n", (char *) data);
                /* XXX we could default to 'dsp' perhaps ? */
                return NULL;
        }
        if ((format & AST_FORMAT_SLINEAR) == 0) {
                ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format);
                return NULL;
        }
        if (o->owner) {
                ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner);
                *cause = AST_CAUSE_BUSY;
                return NULL;
        }
        c = oss_new(o, NULL, NULL, AST_STATE_DOWN);
        if (c == NULL) {
                ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
                return NULL;
        }
        return c;
}

static int console_autoanswer_deprecated(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc == 1) {
                ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
                return RESULT_SUCCESS;
        }
        if (argc != 2)
                return RESULT_SHOWUSAGE;
        if (o == NULL) {
                ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n", oss_active);
                return RESULT_FAILURE;
        }
        if (!strcasecmp(argv[1], "on"))
                o->autoanswer = -1;
        else if (!strcasecmp(argv[1], "off"))
                o->autoanswer = 0;
        else
                return RESULT_SHOWUSAGE;
        return RESULT_SUCCESS;
}

static int console_autoanswer(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc == 2) {
                ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
                return RESULT_SUCCESS;
        }
        if (argc != 3)
                return RESULT_SHOWUSAGE;
        if (o == NULL) {
                ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
                    oss_active);
                return RESULT_FAILURE;
        }
        if (!strcasecmp(argv[2], "on"))
                o->autoanswer = -1;
        else if (!strcasecmp(argv[2], "off"))
                o->autoanswer = 0;
        else
                return RESULT_SHOWUSAGE;
        return RESULT_SUCCESS;
}

static char *autoanswer_complete_deprecated(const char *line, const char *word, int pos, int state)
{
        static char *choices[] = { "on", "off", NULL };

        return (pos != 2) ? NULL : ast_cli_complete(word, choices, state);
}

static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
{
        static char *choices[] = { "on", "off", NULL };

        return (pos != 3) ? NULL : ast_cli_complete(word, choices, state);
}

static char autoanswer_usage[] =
        "Usage: console 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";

/*
 * answer command from the console
 */
static int console_answer_deprecated(int fd, int argc, char *argv[])
{
        struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 1)
                return RESULT_SHOWUSAGE;
        if (!o->owner) {
                ast_cli(fd, "No one is calling us\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 1;
        o->cursound = -1;
        o->nosound = 0;
        ast_queue_frame(o->owner, &f);
#if 0
        /* XXX do we really need it ? considering we shut down immediately... */
        ring(o, AST_CONTROL_ANSWER);
#endif
        return RESULT_SUCCESS;
}

static int console_answer(int fd, int argc, char *argv[])
{
        struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 2)
                return RESULT_SHOWUSAGE;
        if (!o->owner) {
                ast_cli(fd, "No one is calling us\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 1;
        o->cursound = -1;
        o->nosound = 0;
        ast_queue_frame(o->owner, &f);
#if 0
        /* XXX do we really need it ? considering we shut down immediately... */
        ring(o, AST_CONTROL_ANSWER);
#endif
        return RESULT_SUCCESS;
}

static char answer_usage[] =
        "Usage: console answer\n"
        "       Answers an incoming call on the console (OSS) channel.\n";

/*
 * concatenate all arguments into a single string. argv is NULL-terminated
 * so we can use it right away
 */
static int console_sendtext_deprecated(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);
        char buf[TEXT_SIZE];

        if (argc < 2)
                return RESULT_SHOWUSAGE;
        if (!o->owner) {
                ast_cli(fd, "Not in a call\n");
                return RESULT_FAILURE;
        }
        ast_join(buf, sizeof(buf) - 1, argv + 2);
        if (!ast_strlen_zero(buf)) {
                struct ast_frame f = { 0, };
                int i = strlen(buf);
                buf[i] = '\n';
                f.frametype = AST_FRAME_TEXT;
                f.subclass = 0;
                f.data = buf;
                f.datalen = i + 1;
                ast_queue_frame(o->owner, &f);
        }
        return RESULT_SUCCESS;
}

static int console_sendtext(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);
        char buf[TEXT_SIZE];

        if (argc < 3)
                return RESULT_SHOWUSAGE;
        if (!o->owner) {
                ast_cli(fd, "Not in a call\n");
                return RESULT_FAILURE;
        }
        ast_join(buf, sizeof(buf) - 1, argv + 3);
        if (!ast_strlen_zero(buf)) {
                struct ast_frame f = { 0, };
                int i = strlen(buf);
                buf[i] = '\n';
                f.frametype = AST_FRAME_TEXT;
                f.subclass = 0;
                f.data = buf;
                f.datalen = i + 1;
                ast_queue_frame(o->owner, &f);
        }
        return RESULT_SUCCESS;
}

static char sendtext_usage[] =
        "Usage: console send text <message>\n"
        "       Sends a text message for display on the remote terminal.\n";

static int console_hangup_deprecated(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 1)
                return RESULT_SHOWUSAGE;
        o->cursound = -1;
        o->nosound = 0;
        if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
                ast_cli(fd, "No call to hang up\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 0;
        if (o->owner)
                ast_queue_hangup(o->owner);
        setformat(o, O_CLOSE);
        return RESULT_SUCCESS;
}

static int console_hangup(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 2)
                return RESULT_SHOWUSAGE;
        o->cursound = -1;
        o->nosound = 0;
        if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
                ast_cli(fd, "No call to hang up\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 0;
        if (o->owner)
                ast_queue_hangup(o->owner);
        setformat(o, O_CLOSE);
        return RESULT_SUCCESS;
}

static char hangup_usage[] =
        "Usage: console hangup\n"
        "       Hangs up any call currently placed on the console.\n";

static int console_flash_deprecated(int fd, int argc, char *argv[])
{
        struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 1)
                return RESULT_SHOWUSAGE;
        o->cursound = -1;
        o->nosound = 0; /* when cursound is -1 nosound must be 0 */
        if (!o->owner) { /* XXX maybe !o->hookstate too ? */
                ast_cli(fd, "No call to flash\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 0;
        if (o->owner) /* XXX must be true, right ? */
                ast_queue_frame(o->owner, &f);
        return RESULT_SUCCESS;
}

static int console_flash(int fd, int argc, char *argv[])
{
        struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 2)
                return RESULT_SHOWUSAGE;
        o->cursound = -1;
        o->nosound = 0;                         /* when cursound is -1 nosound must be 0 */
        if (!o->owner) {                        /* XXX maybe !o->hookstate too ? */
                ast_cli(fd, "No call to flash\n");
                return RESULT_FAILURE;
        }
        o->hookstate = 0;
        if (o->owner)                           /* XXX must be true, right ? */
                ast_queue_frame(o->owner, &f);
        return RESULT_SUCCESS;
}

static char flash_usage[] =
        "Usage: console flash\n"
        "       Flashes the call currently placed on the console.\n";

static int console_dial_deprecated(int fd, int argc, char *argv[])
{
        char *s = NULL, *mye = NULL, *myc = NULL;
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 1 && argc != 2)
                return RESULT_SHOWUSAGE;
        if (o->owner) { /* already in a call */
                int i;
                struct ast_frame f = { AST_FRAME_DTMF, 0 };

                if (argc == 1) { /* argument is mandatory here */
                        ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
                        return RESULT_FAILURE;
                }
                s = argv[1];
                /* send the string one char at a time */
                for (i = 0; i < strlen(s); i++) {
                        f.subclass = s[i];
                        ast_queue_frame(o->owner, &f);
                }
                return RESULT_SUCCESS;
        }
        /* if we have an argument split it into extension and context */
        if (argc == 2)
                s = ast_ext_ctx(argv[1], &mye, &myc);
        /* supply default values if needed */
        if (mye == NULL)
                mye = o->ext;
        if (myc == NULL)
                myc = o->ctx;
        if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
                o->hookstate = 1;
                oss_new(o, mye, myc, AST_STATE_RINGING);
        } else
                ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
        if (s)
                free(s);
        return RESULT_SUCCESS;
}

static int console_dial(int fd, int argc, char *argv[])
{
        char *s = NULL, *mye = NULL, *myc = NULL;
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc != 2 && argc != 3)
                return RESULT_SHOWUSAGE;
        if (o->owner) { /* already in a call */
                int i;
                struct ast_frame f = { AST_FRAME_DTMF, 0 };

                if (argc == 2) {        /* argument is mandatory here */
                        ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n");
                        return RESULT_FAILURE;
                }
                s = argv[2];
                /* send the string one char at a time */
                for (i = 0; i < strlen(s); i++) {
                        f.subclass = s[i];
                        ast_queue_frame(o->owner, &f);
                }
                return RESULT_SUCCESS;
        }
        /* if we have an argument split it into extension and context */
        if (argc == 3)
                s = ast_ext_ctx(argv[2], &mye, &myc);
        /* supply default values if needed */
        if (mye == NULL)
                mye = o->ext;
        if (myc == NULL)
                myc = o->ctx;
        if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
                o->hookstate = 1;
                oss_new(o, mye, myc, AST_STATE_RINGING);
        } else
                ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
        if (s)
                free(s);
        return RESULT_SUCCESS;
}

static char dial_usage[] =
        "Usage: console dial [extension[@context]]\n"
        "       Dials a given extension (and context if specified)\n";

static int __console_mute_unmute(int mute)
{
        struct chan_oss_pvt *o = find_desc(oss_active);
        
        o->mute = mute;
        return RESULT_SUCCESS;
}

static int console_mute_deprecated(int fd, int argc, char *argv[])
{
        if (argc != 1)
                return RESULT_SHOWUSAGE;

        return __console_mute_unmute(1);
}

static int console_mute(int fd, int argc, char *argv[])
{
        if (argc != 2)
                return RESULT_SHOWUSAGE;

        return __console_mute_unmute(1);
}

static char mute_usage[] =
        "Usage: console mute\nMutes the microphone\n";

static int console_unmute_deprecated(int fd, int argc, char *argv[])
{
        if (argc != 1)
                return RESULT_SHOWUSAGE;

        return __console_mute_unmute(0);
}

static int console_unmute(int fd, int argc, char *argv[])
{
        if (argc != 2)
                return RESULT_SHOWUSAGE;

        return __console_mute_unmute(0);
}

static char unmute_usage[] =
        "Usage: console unmute\nUnmutes the microphone\n";

static int console_transfer_deprecated(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);
        struct ast_channel *b = NULL;
        char *tmp, *ext, *ctx;

        if (argc != 2)
                return RESULT_SHOWUSAGE;
        if (o == NULL)
                return RESULT_FAILURE;
        if (o->owner ==NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
                ast_cli(fd, "There is no call to transfer\n");
                return RESULT_SUCCESS;
        }

        tmp = ast_ext_ctx(argv[1], &ext, &ctx);
        if (ctx == NULL)                /* supply default context if needed */
                ctx = o->owner->context;
        if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
                ast_cli(fd, "No such extension exists\n");
        else {
                ast_cli(fd, "Whee, transferring %s to %s@%s.\n",
                        b->name, ext, ctx);
                if (ast_async_goto(b, ctx, ext, 1))
                        ast_cli(fd, "Failed to transfer :(\n");
        }
        if (tmp)
                free(tmp);
        return RESULT_SUCCESS;
}

static int console_transfer(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);
        struct ast_channel *b = NULL;
        char *tmp, *ext, *ctx;

        if (argc != 3)
                return RESULT_SHOWUSAGE;
        if (o == NULL)
                return RESULT_FAILURE;
        if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
                ast_cli(fd, "There is no call to transfer\n");
                return RESULT_SUCCESS;
        }

        tmp = ast_ext_ctx(argv[2], &ext, &ctx);
        if (ctx == NULL)                        /* supply default context if needed */
                ctx = o->owner->context;
        if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num))
                ast_cli(fd, "No such extension exists\n");
        else {
                ast_cli(fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx);
                if (ast_async_goto(b, ctx, ext, 1))
                        ast_cli(fd, "Failed to transfer :(\n");
        }
        if (tmp)
                free(tmp);
        return RESULT_SUCCESS;
}

static char transfer_usage[] =
        "Usage: console transfer <extension>[@context]\n"
        "       Transfers the currently connected call to the given extension (and\n"
        "context if specified)\n";

static int console_active_deprecated(int fd, int argc, char *argv[])
{
        if (argc == 1)
                ast_cli(fd, "active console is [%s]\n", oss_active);
        else if (argc != 2)
                return RESULT_SHOWUSAGE;
        else {
                struct chan_oss_pvt *o;
                if (strcmp(argv[1], "show") == 0) {
                        for (o = oss_default.next; o; o = o->next)
                                ast_cli(fd, "device [%s] exists\n", o->name);
                        return RESULT_SUCCESS;
                }
                o = find_desc(argv[1]);
                if (o == NULL)
                        ast_cli(fd, "No device [%s] exists\n", argv[1]);
                else
                        oss_active = o->name;
        }
        return RESULT_SUCCESS;
}

static int console_active(int fd, int argc, char *argv[])
{
        if (argc == 2)
                ast_cli(fd, "active console is [%s]\n", oss_active);
        else if (argc != 3)
                return RESULT_SHOWUSAGE;
        else {
                struct chan_oss_pvt *o;
                if (strcmp(argv[2], "show") == 0) {
                        for (o = oss_default.next; o; o = o->next)
                                ast_cli(fd, "device [%s] exists\n", o->name);
                        return RESULT_SUCCESS;
                }
                o = find_desc(argv[2]);
                if (o == NULL)
                        ast_cli(fd, "No device [%s] exists\n", argv[2]);
                else
                        oss_active = o->name;
        }
        return RESULT_SUCCESS;
}

static char active_usage[] =
        "Usage: console active [device]\n"
        "       If used without a parameter, displays which device is the current\n"
        "console.  If a device is specified, the console sound device is changed to\n"
        "the device specified.\n";

/*
 * store the boost factor
 */
static void store_boost(struct chan_oss_pvt *o, char *s)
{
        double boost = 0;
        if (sscanf(s, "%lf", &boost) != 1) {
                ast_log(LOG_WARNING, "invalid boost <%s>\n", s);
                return;
        }
        if (boost < -BOOST_MAX) {
                ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX);
                boost = -BOOST_MAX;
        } else if (boost > BOOST_MAX) {
                ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX);
                boost = BOOST_MAX;
        }
        boost = exp(log(10) * boost / 20) * BOOST_SCALE;
        o->boost = boost;
        ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost);
}

static int do_boost(int fd, int argc, char *argv[])
{
        struct chan_oss_pvt *o = find_desc(oss_active);

        if (argc == 2)
                ast_cli(fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
        else if (argc == 3)
                store_boost(o, argv[2]);
        return RESULT_SUCCESS;
}

static struct ast_cli_entry cli_oss_answer_deprecated = {
        { "answer", NULL },
        console_answer_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_hangup_deprecated = {
        { "hangup", NULL },
        console_hangup_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_flash_deprecated = {
        { "flash", NULL },
        console_flash_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_dial_deprecated = {
        { "dial", NULL },
        console_dial_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_mute_deprecated = {
        { "mute", NULL },
        console_mute_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_unmute_deprecated = {
        { "unmute", NULL },
        console_unmute_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_transfer_deprecated = {
        { "transfer", NULL },
        console_transfer_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_send_text_deprecated = {
        { "send", "text", NULL },
        console_sendtext_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss_autoanswer_deprecated = {
        { "autoanswer", NULL },
        console_autoanswer_deprecated, NULL,
        NULL, autoanswer_complete_deprecated };

static struct ast_cli_entry cli_oss_boost_deprecated = {
        { "oss", "boost", NULL },
        do_boost, NULL,
        NULL };

static struct ast_cli_entry cli_oss_active_deprecated = {
        { "console", NULL },
        console_active_deprecated, NULL,
        NULL };

static struct ast_cli_entry cli_oss[] = {
        { { "console", "answer", NULL },
        console_answer, "Answer an incoming console call",
        answer_usage, NULL, &cli_oss_answer_deprecated },

        { { "console", "hangup", NULL },
        console_hangup, "Hangup a call on the console",
        hangup_usage, NULL, &cli_oss_hangup_deprecated },

        { { "console", "flash", NULL },
        console_flash, "Flash a call on the console",
        flash_usage, NULL, &cli_oss_flash_deprecated },

        { { "console", "dial", NULL },
        console_dial, "Dial an extension on the console",
        dial_usage, NULL, &cli_oss_dial_deprecated },

        { { "console", "mute", NULL },
        console_mute, "Disable mic input",
        mute_usage, NULL, &cli_oss_mute_deprecated },

        { { "console", "unmute", NULL },
        console_unmute, "Enable mic input",
        unmute_usage, NULL, &cli_oss_unmute_deprecated },

        { { "console", "transfer", NULL },
        console_transfer, "Transfer a call to a different extension",
        transfer_usage, NULL, &cli_oss_transfer_deprecated },

        { { "console", "send", "text", NULL },
        console_sendtext, "Send text to the remote device",
        sendtext_usage, NULL, &cli_oss_send_text_deprecated },

        { { "console", "autoanswer", NULL },
        console_autoanswer, "Sets/displays autoanswer",
        autoanswer_usage, autoanswer_complete, &cli_oss_autoanswer_deprecated },

        { { "console", "boost", NULL },
        do_boost, "Sets/displays mic boost in dB",
        NULL, NULL, &cli_oss_boost_deprecated },

        { { "console", "active", NULL },
        console_active, "Sets/displays active console",
        active_usage, NULL, &cli_oss_active_deprecated },
};

/*
 * store the mixer argument from the config file, filtering possibly
 * invalid or dangerous values (the string is used as argument for
 * system("mixer %s")
 */
static void store_mixer(struct chan_oss_pvt *o, char *s)
{
        int i;

        for (i = 0; i < strlen(s); i++) {
                if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) {
                        ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s);
                        return;
                }
        }
        if (o->mixer_cmd)
                free(o->mixer_cmd);
        o->mixer_cmd = ast_strdup(s);
        ast_log(LOG_WARNING, "setting mixer %s\n", s);
}

/*
 * store the callerid components
 */
static void store_callerid(struct chan_oss_pvt *o, char *s)
{
        ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
}

/*
 * grab fields from the config file, init the descriptor and open the device.
 */
static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg)
{
        struct ast_variable *v;
        struct chan_oss_pvt *o;

        if (ctg == NULL) {
                o = &oss_default;
                ctg = "general";
        } else {
                if (!(o = ast_calloc(1, sizeof(*o))))
                        return NULL;
                *o = oss_default;
                /* "general" is also the default thing */
                if (strcmp(ctg, "general") == 0) {
                        o->name = ast_strdup("dsp");
                        oss_active = o->name;
                        goto openit;
                }
                o->name = ast_strdup(ctg);
        }

        strcpy(o->mohinterpret, "default");

        o->lastopen = ast_tvnow();      /* don't leave it 0 or tvdiff may wrap */
        /* fill other fields from configuration */
        for (v = ast_variable_browse(cfg, ctg); v; v = v->next) {
                M_START(v->name, v->value);

                /* handle jb conf */
                if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
                        continue;

                M_BOOL("autoanswer", o->autoanswer)
                        M_BOOL("autohangup", o->autohangup)
                        M_BOOL("overridecontext", o->overridecontext)
                        M_STR("device", o->device)
                        M_UINT("frags", o->frags)
                        M_UINT("debug", oss_debug)
                        M_UINT("queuesize", o->queuesize)
                        M_STR("context", o->ctx)
                        M_STR("language", o->language)
                        M_STR("mohinterpret", o->mohinterpret)
                        M_STR("extension", o->ext)
                        M_F("mixer", store_mixer(o, v->value))
                        M_F("callerid", store_callerid(o, v->value))
                        M_F("boost", store_boost(o, v->value))
                        M_END(;
                        );
        }
        if (ast_strlen_zero(o->device))
                ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
        if (o->mixer_cmd) {
                char *cmd;

                asprintf(&cmd, "mixer %s", o->mixer_cmd);
                ast_log(LOG_WARNING, "running [%s]\n", cmd);
                system(cmd);
                free(cmd);
        }
        if (o == &oss_default)          /* we are done with the default */
                return NULL;

  openit:
#if TRYOPEN
        if (setformat(o, O_RDWR) < 0) { /* open device */
                if (option_verbose > 0) {
                        ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg);
                        ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
                }
                goto error;
        }
        if (o->duplex != M_FULL)
                ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n");
#endif /* TRYOPEN */
        if (pipe(o->sndcmd) != 0) {
                ast_log(LOG_ERROR, "Unable to create pipe\n");
                goto error;
        }
        ast_pthread_create_background(&o->sthread, NULL, sound_thread, o);
        /* link into list of devices */
        if (o != &oss_default) {
                o->next = oss_default.next;
                oss_default.next = o;
        }
        return o;

  error:
        if (o != &oss_default)
                free(o);
        return NULL;
}

static int load_module(void)
{
        struct ast_config *cfg = NULL;
        char *ctg = NULL;

        /* Copy the default jb config over global_jbconf */
        memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));

        /* load config file */
        if (!(cfg = ast_config_load(config))) {
                ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
                return AST_MODULE_LOAD_DECLINE;
        }

        do {
                store_config(cfg, ctg);
        } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);

        ast_config_destroy(cfg);

        if (find_desc(oss_active) == NULL) {
                ast_log(LOG_NOTICE, "Device %s not found\n", oss_active);
                /* XXX we could default to 'dsp' perhaps ? */
                /* XXX should cleanup allocated memory etc. */
                return AST_MODULE_LOAD_FAILURE;
        }

        if (ast_channel_register(&oss_tech)) {
                ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
                return AST_MODULE_LOAD_FAILURE;
        }

        ast_cli_register_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));

        return AST_MODULE_LOAD_SUCCESS;
}


static int unload_module(void)
{
        struct chan_oss_pvt *o;

        ast_channel_unregister(&oss_tech);
        ast_cli_unregister_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry));

        for (o = oss_default.next; o; o = o->next) {
                close(o->sounddev);
                if (o->sndcmd[0] > 0) {
                        close(o->sndcmd[0]);
                        close(o->sndcmd[1]);
                }
                if (o->owner)
                        ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
                if (o->owner)                   /* XXX how ??? */
                        return -1;
                /* XXX what about the thread ? */
                /* XXX what about the memory allocated ? */
        }
        return 0;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");