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asterisk/channels/chan_oss.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2007, 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.
*/
// #define HAVE_VIDEO_CONSOLE // uncomment to enable video
/*! \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>oss</depend>
<support_level>extended</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <ctype.h> /* isalnum() used here */
#include <math.h>
#include <sys/ioctl.h>
#ifdef __linux
#include <linux/soundcard.h>
#elif defined(__FreeBSD__) || defined(__CYGWIN__) || defined(__GLIBC__)
#include <sys/soundcard.h>
#else
#include <soundcard.h>
#endif
#include "asterisk/channel.h"
#include "asterisk/file.h"
#include "asterisk/callerid.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/cli.h"
#include "asterisk/causes.h"
#include "asterisk/musiconhold.h"
#include "asterisk/app.h"
#include "console_video.h"
/*! Global jitterbuffer configuration - by default, jb is disabled
* \note Values shown here match the defaults shown in oss.conf.sample */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = 200,
.resync_threshold = 1000,
.impl = "fixed",
.target_extra = 40,
};
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.
*/
/*
* 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
static char *config = "oss.conf"; /* default config file */
static int oss_debug;
/*!
* \brief 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;
int total_blocks; /*!< total blocks in the output device */
int sounddev;
enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex;
int autoanswer; /*!< Boolean: whether to answer the immediately upon calling */
int autohangup; /*!< Boolean: whether to hangup the call when the remote end hangs up */
int hookstate; /*!< Boolean: 1 if offhook; 0 if onhook */
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;
struct video_desc *env; /*!< parameters for video support */
char ext[AST_MAX_EXTENSION];
char ctx[AST_MAX_CONTEXT];
char language[MAX_LANGUAGE];
char cid_name[256]; /*!< Initial CallerID name */
char cid_num[256]; /*!< Initial CallerID number */
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 */
};
/*! forward declaration */
static struct chan_oss_pvt *find_desc(const char *dev);
static char *oss_active; /*!< the active device */
/*! \brief return the pointer to the video descriptor */
struct video_desc *get_video_desc(struct ast_channel *c)
{
struct chan_oss_pvt *o = c ? c->tech_pvt : find_desc(oss_active);
return o ? o->env : NULL;
}
static struct chan_oss_pvt oss_default = {
.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 int setformat(struct chan_oss_pvt *o, int mode);
static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor,
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";
/* cannot do const because need to update some fields at runtime */
static struct ast_channel_tech oss_tech = {
.type = "Console",
.description = tdesc,
.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,
.write_video = console_write_video,
.indicate = oss_indicate,
.fixup = oss_fixup,
};
/*!
* \brief returns a pointer to the descriptor with the given name
*/
static struct chan_oss_pvt *find_desc(const 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;
}
/* !
* \brief 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 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;
}
/*!
* \brief 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);
}
/*!
* 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)
ast_channel_set_fd(o->owner, 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)) {
ast_verb(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;
}
/*!
* \brief 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 = { AST_FRAME_CONTROL, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(name);
AST_APP_ARG(flags);
);
char *parse = ast_strdupa(dest);
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n",
dest,
S_OR(c->dialed.number.str, ""),
S_COR(c->redirecting.from.number.valid, c->redirecting.from.number.str, ""),
S_COR(c->caller.id.name.valid, c->caller.id.name.str, ""),
S_COR(c->caller.id.number.valid, c->caller.id.number.str, ""));
if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "answer") == 0) {
f.subclass.integer = AST_CONTROL_ANSWER;
ast_queue_frame(c, &f);
} else if (!ast_strlen_zero(args.flags) && strcasecmp(args.flags, "noanswer") == 0) {
f.subclass.integer = AST_CONTROL_RINGING;
ast_queue_frame(c, &f);
ast_indicate(c, AST_CONTROL_RINGING);
} else if (o->autoanswer) {
ast_verbose(" << Auto-answered >> \n");
f.subclass.integer = AST_CONTROL_ANSWER;
ast_queue_frame(c, &f);
o->hookstate = 1;
} else {
ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
f.subclass.integer = AST_CONTROL_RINGING;
ast_queue_frame(c, &f);
ast_indicate(c, AST_CONTROL_RINGING);
}
return 0;
}
/*!
* \brief 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");
ast_setstate(c, AST_STATE_UP);
o->hookstate = 1;
return 0;
}
static int oss_hangup(struct ast_channel *c)
{
struct chan_oss_pvt *o = c->tech_pvt;
c->tech_pvt = NULL;
o->owner = NULL;
ast_verbose(" << Hangup on console >> \n");
console_video_uninit(o->env);
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);
}
}
return 0;
}
/*! \brief 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;
/*
* 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.ptr + 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.ptr + 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 */
memset(f, '\0', 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;
ast_format_set(&f->subclass.format, AST_FORMAT_SLINEAR, 0);
f->samples = FRAME_SIZE;
f->datalen = FRAME_SIZE * 2;
f->data.ptr = 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.ptr;
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 = 0;
switch (cond) {
case AST_CONTROL_BUSY:
case AST_CONTROL_CONGESTION:
case AST_CONTROL_RINGING:
case -1:
res = -1;
break;
case AST_CONTROL_PROGRESS:
case AST_CONTROL_PROCEEDING:
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
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;
default:
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name);
return -1;
}
return res;
}
/*!
* \brief allocate a new channel.
*/
static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state, const char *linkedid)
{
struct ast_channel *c;
c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, linkedid, 0, "Console/%s", o->device + 5);
if (c == NULL)
return NULL;
c->tech = &oss_tech;
if (o->sounddev < 0)
setformat(o, O_RDWR);
ast_channel_set_fd(c, 0, o->sounddev); /* -1 if device closed, override later */
ast_format_set(&c->readformat, AST_FORMAT_SLINEAR, 0);
ast_format_set(&c->writeformat, AST_FORMAT_SLINEAR, 0);
ast_format_cap_add(c->nativeformats, &c->readformat);
/* if the console makes the call, add video to the offer */
/* if (state == AST_STATE_RINGING) TODO XXX CONSOLE VIDEO IS DISABLED UNTIL IT GETS A MAINTAINER
c->nativeformats |= console_video_formats; */
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 */
if (!ast_strlen_zero(o->cid_num)) {
c->caller.ani.number.valid = 1;
c->caller.ani.number.str = ast_strdup(o->cid_num);
}
if (!ast_strlen_zero(ext)) {
c->dialed.number.str = 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;
}
}
console_video_start(get_video_desc(c), c); /* XXX cleanup */
return c;
}
static struct ast_channel *oss_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, void *data, int *cause)
{
struct ast_channel *c;
struct chan_oss_pvt *o;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(name);
AST_APP_ARG(flags);
);
char *parse = ast_strdupa(data);
char buf[256];
struct ast_format tmpfmt;
AST_NONSTANDARD_APP_ARGS(args, parse, '/');
o = find_desc(args.name);
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", args.name);
/* XXX we could default to 'dsp' perhaps ? */
return NULL;
}
if (!(ast_format_cap_iscompatible(cap, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0)))) {
ast_log(LOG_NOTICE, "Format %s unsupported\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
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, requestor ? requestor->linkedid : NULL);
if (c == NULL) {
ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
return NULL;
}
return c;
}
static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value);
/*! Generic console command handler. Basically a wrapper for a subset
* of config file options which are also available from the CLI
*/
static char *console_cmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
const char *var, *value;
switch (cmd) {
case CLI_INIT:
e->command = CONSOLE_VIDEO_CMDS;
e->usage =
"Usage: " CONSOLE_VIDEO_CMDS "...\n"
" Generic handler for console commands.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc < e->args)
return CLI_SHOWUSAGE;
if (o == NULL) {
ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
oss_active);
return CLI_FAILURE;
}
var = a->argv[e->args-1];
value = a->argc > e->args ? a->argv[e->args] : NULL;
if (value) /* handle setting */
store_config_core(o, var, value);
if (!console_video_cli(o->env, var, a->fd)) /* print video-related values */
return CLI_SUCCESS;
/* handle other values */
if (!strcasecmp(var, "device")) {
ast_cli(a->fd, "device is [%s]\n", o->device);
}
return CLI_SUCCESS;
}
static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
switch (cmd) {
case CLI_INIT:
e->command = "console {set|show} autoanswer [on|off]";
e->usage =
"Usage: console {set|show} 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";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == e->args - 1) {
ast_cli(a->fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off");
return CLI_SUCCESS;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (o == NULL) {
ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
oss_active);
return CLI_FAILURE;
}
if (!strcasecmp(a->argv[e->args-1], "on"))
o->autoanswer = 1;
else if (!strcasecmp(a->argv[e->args - 1], "off"))
o->autoanswer = 0;
else
return CLI_SHOWUSAGE;
return CLI_SUCCESS;
}
/*! \brief helper function for the answer key/cli command */
static char *console_do_answer(int fd)
{
struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_ANSWER } };
struct chan_oss_pvt *o = find_desc(oss_active);
if (!o->owner) {
if (fd > -1)
ast_cli(fd, "No one is calling us\n");
return CLI_FAILURE;
}
o->hookstate = 1;
ast_queue_frame(o->owner, &f);
return CLI_SUCCESS;
}
/*!
* \brief answer command from the console
*/
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "console answer";
e->usage =
"Usage: console answer\n"
" Answers an incoming call on the console (OSS) channel.\n";
return NULL;
case CLI_GENERATE:
return NULL; /* no completion */
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
return console_do_answer(a->fd);
}
/*!
* \brief Console send text CLI command
*
* \note concatenate all arguments into a single string. argv is NULL-terminated
* so we can use it right away
*/
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
char buf[TEXT_SIZE];
if (cmd == CLI_INIT) {
e->command = "console send text";
e->usage =
"Usage: console send text <message>\n"
" Sends a text message for display on the remote terminal.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc < e->args + 1)
return CLI_SHOWUSAGE;
if (!o->owner) {
ast_cli(a->fd, "Not in a call\n");
return CLI_FAILURE;
}
ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
if (!ast_strlen_zero(buf)) {
struct ast_frame f = { 0, };
int i = strlen(buf);
buf[i] = '\n';
f.frametype = AST_FRAME_TEXT;
f.subclass.integer = 0;
f.data.ptr = buf;
f.datalen = i + 1;
ast_queue_frame(o->owner, &f);
}
return CLI_SUCCESS;
}
static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console hangup";
e->usage =
"Usage: console hangup\n"
" Hangs up any call currently placed on the console.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
ast_cli(a->fd, "No call to hang up\n");
return CLI_FAILURE;
}
o->hookstate = 0;
if (o->owner)
ast_queue_hangup_with_cause(o->owner, AST_CAUSE_NORMAL_CLEARING);
setformat(o, O_CLOSE);
return CLI_SUCCESS;
}
static char *console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH } };
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console flash";
e->usage =
"Usage: console flash\n"
" Flashes the call currently placed on the console.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!o->owner) { /* XXX maybe !o->hookstate too ? */
ast_cli(a->fd, "No call to flash\n");
return CLI_FAILURE;
}
o->hookstate = 0;
if (o->owner)
ast_queue_frame(o->owner, &f);
return CLI_SUCCESS;
}
static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char *s = NULL;
char *mye = NULL, *myc = NULL;
struct chan_oss_pvt *o = find_desc(oss_active);
if (cmd == CLI_INIT) {
e->command = "console dial";
e->usage =
"Usage: console dial [extension[@context]]\n"
" Dials a given extension (and context if specified)\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc > e->args + 1)
return CLI_SHOWUSAGE;
if (o->owner) { /* already in a call */
int i;
struct ast_frame f = { AST_FRAME_DTMF, { 0 } };
const char *s;
if (a->argc == e->args) { /* argument is mandatory here */
ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
return CLI_FAILURE;
}
s = a->argv[e->args];
/* send the string one char at a time */
for (i = 0; i < strlen(s); i++) {
f.subclass.integer = s[i];
ast_queue_frame(o->owner, &f);
}
return CLI_SUCCESS;
}
/* if we have an argument split it into extension and context */
if (a->argc == e->args + 1)
s = ast_ext_ctx(a->argv[e->args], &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, NULL);
} else
ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
if (s)
ast_free(s);
return CLI_SUCCESS;
}
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
const char *s;
int toggle = 0;
if (cmd == CLI_INIT) {
e->command = "console {mute|unmute} [toggle]";
e->usage =
"Usage: console {mute|unmute} [toggle]\n"
" Mute/unmute the microphone.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc > e->args)
return CLI_SHOWUSAGE;
if (a->argc == e->args) {
if (strcasecmp(a->argv[e->args-1], "toggle"))
return CLI_SHOWUSAGE;
toggle = 1;
}
s = a->argv[e->args-2];
if (!strcasecmp(s, "mute"))
o->mute = toggle ? !o->mute : 1;
else if (!strcasecmp(s, "unmute"))
o->mute = toggle ? !o->mute : 0;
else
return CLI_SHOWUSAGE;
ast_cli(a->fd, "Console mic is %s\n", o->mute ? "off" : "on");
return CLI_SUCCESS;
}
static char *console_transfer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
struct ast_channel *b = NULL;
char *tmp, *ext, *ctx;
switch (cmd) {
case CLI_INIT:
e->command = "console transfer";
e->usage =
"Usage: console transfer <extension>[@context]\n"
" Transfers the currently connected call to the given extension (and\n"
" context if specified)\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
if (o == NULL)
return CLI_FAILURE;
if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) {
ast_cli(a->fd, "There is no call to transfer\n");
return CLI_SUCCESS;
}
tmp = ast_ext_ctx(a->argv[2], &ext, &ctx);
if (ctx == NULL) /* supply default context if needed */
ctx = o->owner->context;
if (!ast_exists_extension(b, ctx, ext, 1,
S_COR(b->caller.id.number.valid, b->caller.id.number.str, NULL))) {
ast_cli(a->fd, "No such extension exists\n");
} else {
ast_cli(a->fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx);
if (ast_async_goto(b, ctx, ext, 1))
ast_cli(a->fd, "Failed to transfer :(\n");
}
if (tmp)
ast_free(tmp);
return CLI_SUCCESS;
}
static char *console_active(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "console {set|show} active [<device>]";
e->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";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == 3)
ast_cli(a->fd, "active console is [%s]\n", oss_active);
else if (a->argc != 4)
return CLI_SHOWUSAGE;
else {
struct chan_oss_pvt *o;
if (strcmp(a->argv[3], "show") == 0) {
for (o = oss_default.next; o; o = o->next)
ast_cli(a->fd, "device [%s] exists\n", o->name);
return CLI_SUCCESS;
}
o = find_desc(a->argv[3]);
if (o == NULL)
ast_cli(a->fd, "No device [%s] exists\n", a->argv[3]);
else
oss_active = o->name;
}
return CLI_SUCCESS;
}
/*!
* \brief store the boost factor
*/
static void store_boost(struct chan_oss_pvt *o, const char *s)
{
double boost = 0;
if (sscanf(s, "%30lf", &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 char *console_boost(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct chan_oss_pvt *o = find_desc(oss_active);
switch (cmd) {
case CLI_INIT:
e->command = "console boost";
e->usage =
"Usage: console boost [boost in dB]\n"
" Sets or display mic boost in dB\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == 2)
ast_cli(a->fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE)));
else if (a->argc == 3)
store_boost(o, a->argv[2]);
return CLI_SUCCESS;
}
static struct ast_cli_entry cli_oss[] = {
AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
AST_CLI_DEFINE(console_flash, "Flash a call on the console"),
AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
AST_CLI_DEFINE(console_transfer, "Transfer a call to a different extension"),
AST_CLI_DEFINE(console_cmd, "Generic console command"),
AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
AST_CLI_DEFINE(console_boost, "Sets/displays mic boost in dB"),
AST_CLI_DEFINE(console_active, "Sets/displays active console"),
};
/*!
* 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, const char *s)
{
int i;
for (i = 0; i < strlen(s); i++) {
if (!isalnum(s[i]) && strchr(" \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)
ast_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, const char *s)
{
ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num));
}
static void store_config_core(struct chan_oss_pvt *o, const char *var, const char *value)
{
CV_START(var, value);
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, var, value))
return;
if (!console_video_config(&o->env, var, value))
return; /* matched there */
CV_BOOL("autoanswer", o->autoanswer);
CV_BOOL("autohangup", o->autohangup);
CV_BOOL("overridecontext", o->overridecontext);
CV_STR("device", o->device);
CV_UINT("frags", o->frags);
CV_UINT("debug", oss_debug);
CV_UINT("queuesize", o->queuesize);
CV_STR("context", o->ctx);
CV_STR("language", o->language);
CV_STR("mohinterpret", o->mohinterpret);
CV_STR("extension", o->ext);
CV_F("mixer", store_mixer(o, value));
CV_F("callerid", store_callerid(o, value)) ;
CV_F("boost", store_boost(o, value));
CV_END;
}
/*!
* 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) {
store_config_core(o, v->name, v->value);
}
if (ast_strlen_zero(o->device))
ast_copy_string(o->device, DEV_DSP, sizeof(o->device));
if (o->mixer_cmd) {
char *cmd;
if (asprintf(&cmd, "mixer %s", o->mixer_cmd) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
} else {
ast_log(LOG_WARNING, "running [%s]\n", cmd);
if (system(cmd) < 0) {
ast_log(LOG_WARNING, "system() failed: %s\n", strerror(errno));
}
ast_free(cmd);
}
}
/* if the config file requested to start the GUI, do it */
if (get_gui_startup(o->env))
console_video_start(o->env, NULL);
if (o == &oss_default) /* we are done with the default */
return NULL;
openit:
#ifdef TRYOPEN
if (setformat(o, O_RDWR) < 0) { /* open device */
ast_verb(1, "Device %s not detected\n", ctg);
ast_verb(1, "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 */
/* link into list of devices */
if (o != &oss_default) {
o->next = oss_default.next;
oss_default.next = o;
}
return o;
#ifdef TRYOPEN
error:
if (o != &oss_default)
ast_free(o);
return NULL;
#endif
}
static int load_module(void)
{
struct ast_config *cfg = NULL;
char *ctg = NULL;
struct ast_flags config_flags = { 0 };
struct ast_format tmpfmt;
/* 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, config_flags))) {
ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
return AST_MODULE_LOAD_DECLINE;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Config file %s is in an invalid format. Aborting.\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 (!(oss_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_FAILURE;
}
ast_format_cap_add(oss_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0));
/* TODO XXX CONSOLE VIDEO IS DISABLE UNTIL IT HAS A MAINTAINER
* add console_video_formats to oss_tech.capabilities once this occurs. */
if (ast_channel_register(&oss_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n");
return AST_MODULE_LOAD_DECLINE;
}
ast_cli_register_multiple(cli_oss, ARRAY_LEN(cli_oss));
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
struct chan_oss_pvt *o, *next;
ast_channel_unregister(&oss_tech);
ast_cli_unregister_multiple(cli_oss, ARRAY_LEN(cli_oss));
o = oss_default.next;
while (o) {
close(o->sounddev);
if (o->owner)
ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD);
if (o->owner)
return -1;
next = o->next;
ast_free(o->name);
ast_free(o);
o = next;
}
oss_tech.capabilities = ast_format_cap_destroy(oss_tech.capabilities);
return 0;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver");
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