Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
1527 lines
44 KiB
C
1527 lines
44 KiB
C
/*
|
|
* 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
|
|
*
|
|
* \ingroup channel_drivers
|
|
*/
|
|
|
|
/*! \li \ref chan_oss.c uses the configuration file \ref oss.conf
|
|
* \addtogroup configuration_file
|
|
*/
|
|
|
|
/*! \page oss.conf oss.conf
|
|
* \verbinclude oss.conf.sample
|
|
*/
|
|
|
|
/*** MODULEINFO
|
|
<depend>oss</depend>
|
|
<support_level>extended</support_level>
|
|
***/
|
|
|
|
#include "asterisk.h"
|
|
|
|
ASTERISK_REGISTER_FILE()
|
|
|
|
#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__) || defined(__sun)
|
|
#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 "asterisk/bridge.h"
|
|
#include "asterisk/format_cache.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 ? ast_channel_tech_pvt(c) : 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_assigned_ids *assignedids, const struct ast_channel *requestor,
|
|
const char *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, const 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 sample rate to %d\n", desired);
|
|
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, const char *dest, int timeout)
|
|
{
|
|
struct chan_oss_pvt *o = ast_channel_tech_pvt(c);
|
|
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(ast_channel_dialed(c)->number.str, ""),
|
|
S_COR(ast_channel_redirecting(c)->from.number.valid, ast_channel_redirecting(c)->from.number.str, ""),
|
|
S_COR(ast_channel_caller(c)->id.name.valid, ast_channel_caller(c)->id.name.str, ""),
|
|
S_COR(ast_channel_caller(c)->id.number.valid, ast_channel_caller(c)->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 = ast_channel_tech_pvt(c);
|
|
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 = ast_channel_tech_pvt(c);
|
|
|
|
ast_channel_tech_pvt_set(c, 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 = ast_channel_tech_pvt(c);
|
|
|
|
/*
|
|
* 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 = ast_channel_tech_pvt(c);
|
|
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 (ast_channel_state(c) != 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.format = ao2_bump(ast_format_slin);
|
|
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 = ast_channel_tech_pvt(newchan);
|
|
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 = ast_channel_tech_pvt(c);
|
|
int res = 0;
|
|
|
|
switch (cond) {
|
|
case AST_CONTROL_INCOMPLETE:
|
|
case AST_CONTROL_BUSY:
|
|
case AST_CONTROL_CONGESTION:
|
|
case AST_CONTROL_RINGING:
|
|
case AST_CONTROL_PVT_CAUSE_CODE:
|
|
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, ast_channel_name(c));
|
|
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 struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
|
|
{
|
|
struct ast_channel *c;
|
|
|
|
c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, assignedids, requestor, 0, "Console/%s", o->device + 5);
|
|
if (c == NULL)
|
|
return NULL;
|
|
ast_channel_tech_set(c, &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_channel_set_readformat(c, ast_format_slin);
|
|
ast_channel_set_writeformat(c, ast_format_slin);
|
|
ast_channel_nativeformats_set(c, oss_tech.capabilities);
|
|
|
|
/* 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; */
|
|
|
|
ast_channel_tech_pvt_set(c, o);
|
|
|
|
if (!ast_strlen_zero(o->language))
|
|
ast_channel_language_set(c, o->language);
|
|
/* Don't use ast_set_callerid() here because it will
|
|
* generate a needless NewCallerID event */
|
|
if (!ast_strlen_zero(o->cid_num)) {
|
|
ast_channel_caller(c)->ani.number.valid = 1;
|
|
ast_channel_caller(c)->ani.number.str = ast_strdup(o->cid_num);
|
|
}
|
|
if (!ast_strlen_zero(ext)) {
|
|
ast_channel_dialed(c)->number.str = ast_strdup(ext);
|
|
}
|
|
|
|
o->owner = c;
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_jb_configure(c, &global_jbconf);
|
|
ast_channel_unlock(c);
|
|
if (state != AST_STATE_DOWN) {
|
|
if (ast_pbx_start(c)) {
|
|
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(c));
|
|
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_assigned_ids *assignedids, const struct ast_channel *requestor, const char *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);
|
|
|
|
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, 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_format(cap, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
struct ast_str *codec_buf = ast_str_alloca(64);
|
|
ast_log(LOG_NOTICE, "Format %s unsupported\n", ast_format_cap_get_names(cap, &codec_buf));
|
|
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, assignedids, requestor);
|
|
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 *digits;
|
|
|
|
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;
|
|
}
|
|
digits = a->argv[e->args];
|
|
/* send the string one char at a time */
|
|
for (i = 0; i < strlen(digits); i++) {
|
|
f.subclass.integer = digits[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, 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);
|
|
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 || !ast_channel_is_bridged(o->owner)) {
|
|
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 = ast_strdupa(ast_channel_context(o->owner));
|
|
}
|
|
if (ast_bridge_transfer_blind(1, o->owner, ext, ctx, NULL, NULL) != AST_BRIDGE_TRANSFER_SUCCESS) {
|
|
ast_log(LOG_WARNING, "Unable to transfer call from channel %s\n", ast_channel_name(o->owner));
|
|
}
|
|
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 (ast_asprintf(&cmd, "mixer %s", o->mixer_cmd) >= 0) {
|
|
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
|
|
}
|
|
|
|
/*!
|
|
* \brief Load the module
|
|
*
|
|
* Module loading including tests for configuration or dependencies.
|
|
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
|
|
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
|
|
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
|
|
* configuration file or other non-critical problem return
|
|
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
|
|
*/
|
|
static int load_module(void)
|
|
{
|
|
struct ast_config *cfg = NULL;
|
|
char *ctg = NULL;
|
|
struct ast_flags config_flags = { 0 };
|
|
|
|
/* 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(0))) {
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
ast_format_cap_append(oss_tech.capabilities, ast_format_slin, 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;
|
|
}
|
|
ao2_cleanup(oss_tech.capabilities);
|
|
oss_tech.capabilities = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO_STANDARD_EXTENDED(ASTERISK_GPL_KEY, "OSS Console Channel Driver");
|
|
|