diff options
Diffstat (limited to 'channels/chan_oss.c')
| -rw-r--r-- | channels/chan_oss.c | 1899 |
1 files changed, 1899 insertions, 0 deletions
diff --git a/channels/chan_oss.c b/channels/chan_oss.c new file mode 100644 index 000000000..ac62f2005 --- /dev/null +++ b/channels/chan_oss.c @@ -0,0 +1,1899 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 1999 - 2005, Digium, Inc. + * + * Mark Spencer <markster@digium.com> + * + * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25 + * note-this code best seen with ts=8 (8-spaces tabs) in the editor + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +/*! \file + * + * \brief Channel driver for OSS sound cards + * + * \author Mark Spencer <markster@digium.com> + * \author Luigi Rizzo + * + * \par See also + * \arg \ref Config_oss + * + * \ingroup channel_drivers + */ + +/*** MODULEINFO + <depend>ossaudio</depend> + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include <stdio.h> +#include <ctype.h> +#include <math.h> +#include <string.h> +#include <unistd.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <sys/time.h> +#include <stdlib.h> +#include <errno.h> + +#ifdef __linux +#include <linux/soundcard.h> +#elif defined(__FreeBSD__) +#include <sys/soundcard.h> +#else +#include <soundcard.h> +#endif + +#include "asterisk/lock.h" +#include "asterisk/frame.h" +#include "asterisk/logger.h" +#include "asterisk/callerid.h" +#include "asterisk/channel.h" +#include "asterisk/module.h" +#include "asterisk/options.h" +#include "asterisk/pbx.h" +#include "asterisk/config.h" +#include "asterisk/cli.h" +#include "asterisk/utils.h" +#include "asterisk/causes.h" +#include "asterisk/endian.h" +#include "asterisk/stringfields.h" +#include "asterisk/abstract_jb.h" +#include "asterisk/musiconhold.h" + +/* ringtones we use */ +#include "busy_tone.h" +#include "ring_tone.h" +#include "ring10.h" +#include "answer.h" + +/*! Global jitterbuffer configuration - by default, jb is disabled */ +static struct ast_jb_conf default_jbconf = +{ + .flags = 0, + .max_size = -1, + .resync_threshold = -1, + .impl = "", +}; +static struct ast_jb_conf global_jbconf; + +/* + * Basic mode of operation: + * + * we have one keyboard (which receives commands from the keyboard) + * and multiple headset's connected to audio cards. + * Cards/Headsets are named as the sections of oss.conf. + * The section called [general] contains the default parameters. + * + * At any time, the keyboard is attached to one card, and you + * can switch among them using the command 'console foo' + * where 'foo' is the name of the card you want. + * + * oss.conf parameters are +START_CONFIG + +[general] + ; General config options, with default values shown. + ; You should use one section per device, with [general] being used + ; for the first device and also as a template for other devices. + ; + ; All but 'debug' can go also in the device-specific sections. + ; + ; debug = 0x0 ; misc debug flags, default is 0 + + ; Set the device to use for I/O + ; device = /dev/dsp + + ; Optional mixer command to run upon startup (e.g. to set + ; volume levels, mutes, etc. + ; mixer = + + ; Software mic volume booster (or attenuator), useful for sound + ; cards or microphones with poor sensitivity. The volume level + ; is in dB, ranging from -20.0 to +20.0 + ; boost = n ; mic volume boost in dB + + ; Set the callerid for outgoing calls + ; callerid = John Doe <555-1234> + + ; autoanswer = no ; no autoanswer on call + ; autohangup = yes ; hangup when other party closes + ; extension = s ; default extension to call + ; context = default ; default context for outgoing calls + ; language = "" ; default language + + ; Default Music on Hold class to use when this channel is placed on hold in + ; the case that the music class is not set on the channel with + ; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel + ; putting this one on hold did not suggest a class to use. + ; + ; mohinterpret=default + + ; If you set overridecontext to 'yes', then the whole dial string + ; will be interpreted as an extension, which is extremely useful + ; to dial SIP, IAX and other extensions which use the '@' character. + ; The default is 'no' just for backward compatibility, but the + ; suggestion is to change it. + ; overridecontext = no ; if 'no', the last @ will start the context + ; if 'yes' the whole string is an extension. + + ; low level device parameters in case you have problems with the + ; device driver on your operating system. You should not touch these + ; unless you know what you are doing. + ; queuesize = 10 ; frames in device driver + ; frags = 8 ; argument to SETFRAGMENT + + ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- + ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an + ; OSS channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The OSS channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive OSS side will always + ; be used if the sending side can create jitter. + + ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + + ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usualy sent from exotic devices + ; and programs. Defaults to 1000. + + ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS + ; channel. Two implementations are currenlty available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + + ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". + ;----------------------------------------------------------------------------------- + +[card1] + ; device = /dev/dsp1 ; alternate device + +END_CONFIG + +.. and so on for the other cards. + + */ + +/* + * Helper macros to parse config arguments. They will go in a common + * header file if their usage is globally accepted. In the meantime, + * we define them here. Typical usage is as below. + * Remember to open a block right before M_START (as it declares + * some variables) and use the M_* macros WITHOUT A SEMICOLON: + * + * { + * M_START(v->name, v->value) + * + * M_BOOL("dothis", x->flag1) + * M_STR("name", x->somestring) + * M_F("bar", some_c_code) + * M_END(some_final_statement) + * ... other code in the block + * } + * + * XXX NOTE these macros should NOT be replicated in other parts of asterisk. + * Likely we will come up with a better way of doing config file parsing. + */ +#define M_START(var, val) \ + char *__s = var; char *__val = val; +#define M_END(x) x; +#define M_F(tag, f) if (!strcasecmp((__s), tag)) { f; } else +#define M_BOOL(tag, dst) M_F(tag, (dst) = ast_true(__val) ) +#define M_UINT(tag, dst) M_F(tag, (dst) = strtoul(__val, NULL, 0) ) +#define M_STR(tag, dst) M_F(tag, ast_copy_string(dst, __val, sizeof(dst))) + +/* + * The following parameters are used in the driver: + * + * FRAME_SIZE the size of an audio frame, in samples. + * 160 is used almost universally, so you should not change it. + * + * FRAGS the argument for the SETFRAGMENT ioctl. + * Overridden by the 'frags' parameter in oss.conf + * + * Bits 0-7 are the base-2 log of the device's block size, + * bits 16-31 are the number of blocks in the driver's queue. + * There are a lot of differences in the way this parameter + * is supported by different drivers, so you may need to + * experiment a bit with the value. + * A good default for linux is 30 blocks of 64 bytes, which + * results in 6 frames of 320 bytes (160 samples). + * FreeBSD works decently with blocks of 256 or 512 bytes, + * leaving the number unspecified. + * Note that this only refers to the device buffer size, + * this module will then try to keep the lenght of audio + * buffered within small constraints. + * + * QUEUE_SIZE The max number of blocks actually allowed in the device + * driver's buffer, irrespective of the available number. + * Overridden by the 'queuesize' parameter in oss.conf + * + * Should be >=2, and at most as large as the hw queue above + * (otherwise it will never be full). + */ + +#define FRAME_SIZE 160 +#define QUEUE_SIZE 10 + +#if defined(__FreeBSD__) +#define FRAGS 0x8 +#else +#define FRAGS ( ( (6 * 5) << 16 ) | 0x6 ) +#endif + +/* + * XXX text message sizes are probably 256 chars, but i am + * not sure if there is a suitable definition anywhere. + */ +#define TEXT_SIZE 256 + +#if 0 +#define TRYOPEN 1 /* try to open on startup */ +#endif +#define O_CLOSE 0x444 /* special 'close' mode for device */ +/* Which device to use */ +#if defined( __OpenBSD__ ) || defined( __NetBSD__ ) +#define DEV_DSP "/dev/audio" +#else +#define DEV_DSP "/dev/dsp" +#endif + +#ifndef MIN +#define MIN(a,b) ((a) < (b) ? (a) : (b)) +#endif +#ifndef MAX +#define MAX(a,b) ((a) > (b) ? (a) : (b)) +#endif + +static char *config = "oss.conf"; /* default config file */ + +static int oss_debug; + +/* + * Each sound is made of 'datalen' samples of sound, repeated as needed to + * generate 'samplen' samples of data, then followed by 'silencelen' samples + * of silence. The loop is repeated if 'repeat' is set. + */ +struct sound { + int ind; + char *desc; + short *data; + int datalen; + int samplen; + int silencelen; + int repeat; +}; + +static struct sound sounds[] = { + { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, + { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 }, + { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 }, + { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 }, + { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 }, + { -1, NULL, 0, 0, 0, 0 }, /* end marker */ +}; + + +/* + * descriptor for one of our channels. + * There is one used for 'default' values (from the [general] entry in + * the configuration file), and then one instance for each device + * (the default is cloned from [general], others are only created + * if the relevant section exists). + */ +struct chan_oss_pvt { + struct chan_oss_pvt *next; + + char *name; + /* + * cursound indicates which in struct sound we play. -1 means nothing, + * any other value is a valid sound, in which case sampsent indicates + * the next sample to send in [0..samplen + silencelen] + * nosound is set to disable the audio data from the channel + * (so we can play the tones etc.). + */ + int sndcmd[2]; /* Sound command pipe */ + int cursound; /* index of sound to send */ + int sampsent; /* # of sound samples sent */ + int nosound; /* set to block audio from the PBX */ + + int total_blocks; /* total blocks in the output device */ + int sounddev; + enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex; + int autoanswer; + int autohangup; + int hookstate; + char *mixer_cmd; /* initial command to issue to the mixer */ + unsigned int queuesize; /* max fragments in queue */ + unsigned int frags; /* parameter for SETFRAGMENT */ + + int warned; /* various flags used for warnings */ +#define WARN_used_blocks 1 +#define WARN_speed 2 +#define WARN_frag 4 + int w_errors; /* overfull in the write path */ + struct timeval lastopen; + + int overridecontext; + int mute; + + /* boost support. BOOST_SCALE * 10 ^(BOOST_MAX/20) must + * be representable in 16 bits to avoid overflows. + */ +#define BOOST_SCALE (1<<9) +#define BOOST_MAX 40 /* slightly less than 7 bits */ + int boost; /* input boost, scaled by BOOST_SCALE */ + char device[64]; /* device to open */ + + pthread_t sthread; + + struct ast_channel *owner; + char ext[AST_MAX_EXTENSION]; + char ctx[AST_MAX_CONTEXT]; + char language[MAX_LANGUAGE]; + char cid_name[256]; /*XXX */ + char cid_num[256]; /*XXX */ + char mohinterpret[MAX_MUSICCLASS]; + + /* buffers used in oss_write */ + char oss_write_buf[FRAME_SIZE * 2]; + int oss_write_dst; + /* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers + * plus enough room for a full frame + */ + char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET]; + int readpos; /* read position above */ + struct ast_frame read_f; /* returned by oss_read */ +}; + +static struct chan_oss_pvt oss_default = { + .cursound = -1, + .sounddev = -1, + .duplex = M_UNSET, /* XXX check this */ + .autoanswer = 1, + .autohangup = 1, + .queuesize = QUEUE_SIZE, + .frags = FRAGS, + .ext = "s", + .ctx = "default", + .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */ + .lastopen = { 0, 0 }, + .boost = BOOST_SCALE, +}; + +static char *oss_active; /* the active device */ + +static int setformat(struct chan_oss_pvt *o, int mode); + +static struct ast_channel *oss_request(const char *type, int format, void *data +, int *cause); +static int oss_digit_begin(struct ast_channel *c, char digit); +static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration); +static int oss_text(struct ast_channel *c, const char *text); +static int oss_hangup(struct ast_channel *c); +static int oss_answer(struct ast_channel *c); +static struct ast_frame *oss_read(struct ast_channel *chan); +static int oss_call(struct ast_channel *c, char *dest, int timeout); +static int oss_write(struct ast_channel *chan, struct ast_frame *f); +static int oss_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen); +static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); +static char tdesc[] = "OSS Console Channel Driver"; + +static const struct ast_channel_tech oss_tech = { + .type = "Console", + .description = tdesc, + .capabilities = AST_FORMAT_SLINEAR, + .requester = oss_request, + .send_digit_begin = oss_digit_begin, + .send_digit_end = oss_digit_end, + .send_text = oss_text, + .hangup = oss_hangup, + .answer = oss_answer, + .read = oss_read, + .call = oss_call, + .write = oss_write, + .indicate = oss_indicate, + .fixup = oss_fixup, +}; + +/* + * returns a pointer to the descriptor with the given name + */ +static struct chan_oss_pvt *find_desc(char *dev) +{ + struct chan_oss_pvt *o = NULL; + + if (!dev) + ast_log(LOG_WARNING, "null dev\n"); + + for (o = oss_default.next; o && o->name && dev && strcmp(o->name, dev) != 0; o = o->next); + + if (!o) + ast_log(LOG_WARNING, "could not find <%s>\n", dev ? dev : "--no-device--"); + + return o; +} + +/* + * split a string in extension-context, returns pointers to malloc'ed + * strings. + * If we do not have 'overridecontext' then the last @ is considered as + * a context separator, and the context is overridden. + * This is usually not very necessary as you can play with the dialplan, + * and it is nice not to need it because you have '@' in SIP addresses. + * Return value is the buffer address. + */ +static char *ast_ext_ctx(const char *src, char **ext, char **ctx) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (ext == NULL || ctx == NULL) + return NULL; /* error */ + + *ext = *ctx = NULL; + + if (src && *src != '\0') + *ext = ast_strdup(src); + + if (*ext == NULL) + return NULL; + + if (!o->overridecontext) { + /* parse from the right */ + *ctx = strrchr(*ext, '@'); + if (*ctx) + *(*ctx)++ = '\0'; + } + + return *ext; +} + +/* + * Returns the number of blocks used in the audio output channel + */ +static int used_blocks(struct chan_oss_pvt *o) +{ + struct audio_buf_info info; + + if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) { + if (!(o->warned & WARN_used_blocks)) { + ast_log(LOG_WARNING, "Error reading output space\n"); + o->warned |= WARN_used_blocks; + } + return 1; + } + + if (o->total_blocks == 0) { + if (0) /* debugging */ + ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", info.fragstotal, info.fragsize, info.fragments); + o->total_blocks = info.fragments; + } + + return o->total_blocks - info.fragments; +} + +/* Write an exactly FRAME_SIZE sized frame */ +static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) +{ + int res; + + if (o->sounddev < 0) + setformat(o, O_RDWR); + if (o->sounddev < 0) + return 0; /* not fatal */ + /* + * Nothing complex to manage the audio device queue. + * If the buffer is full just drop the extra, otherwise write. + * XXX in some cases it might be useful to write anyways after + * a number of failures, to restart the output chain. + */ + res = used_blocks(o); + if (res > o->queuesize) { /* no room to write a block */ + if (o->w_errors++ == 0 && (oss_debug & 0x4)) + ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, o->w_errors); + return 0; + } + o->w_errors = 0; + return write(o->sounddev, ((void *) data), FRAME_SIZE * 2); +} + +/* + * Handler for 'sound writable' events from the sound thread. + * Builds a frame from the high level description of the sounds, + * and passes it to the audio device. + * The actual sound is made of 1 or more sequences of sound samples + * (s->datalen, repeated to make s->samplen samples) followed by + * s->silencelen samples of silence. The position in the sequence is stored + * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen. + * In case we fail to write a frame, don't update o->sampsent. + */ +static void send_sound(struct chan_oss_pvt *o) +{ + short myframe[FRAME_SIZE]; + int ofs, l, start; + int l_sampsent = o->sampsent; + struct sound *s; + + if (o->cursound < 0) /* no sound to send */ + return; + + s = &sounds[o->cursound]; + + for (ofs = 0; ofs < FRAME_SIZE; ofs += l) { + l = s->samplen - l_sampsent; /* # of available samples */ + if (l > 0) { + start = l_sampsent % s->datalen; /* source offset */ + if (l > FRAME_SIZE - ofs) /* don't overflow the frame */ + l = FRAME_SIZE - ofs; + if (l > s->datalen - start) /* don't overflow the source */ + l = s->datalen - start; + bcopy(s->data + start, myframe + ofs, l * 2); + if (0) + ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", l_sampsent, l, s->samplen, ofs); + l_sampsent += l; + } else { /* end of samples, maybe some silence */ + static const short silence[FRAME_SIZE] = { 0, }; + + l += s->silencelen; + if (l > 0) { + if (l > FRAME_SIZE - ofs) + l = FRAME_SIZE - ofs; + bcopy(silence, myframe + ofs, l * 2); + l_sampsent += l; + } else { /* silence is over, restart sound if loop */ + if (s->repeat == 0) { /* last block */ + o->cursound = -1; + o->nosound = 0; /* allow audio data */ + if (ofs < FRAME_SIZE) /* pad with silence */ + bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs) * 2); + } + l_sampsent = 0; + } + } + } + l = soundcard_writeframe(o, myframe); + if (l > 0) + o->sampsent = l_sampsent; /* update status */ +} + +static void *sound_thread(void *arg) +{ + char ign[4096]; + struct chan_oss_pvt *o = (struct chan_oss_pvt *) arg; + + /* + * Just in case, kick the driver by trying to read from it. + * Ignore errors - this read is almost guaranteed to fail. + */ + if (read(o->sounddev, ign, sizeof(ign)) < 0) { + } + for (;;) { + fd_set rfds, wfds; + int maxfd, res; + + FD_ZERO(&rfds); + FD_ZERO(&wfds); + FD_SET(o->sndcmd[0], &rfds); + maxfd = o->sndcmd[0]; /* pipe from the main process */ + if (o->cursound > -1 && o->sounddev < 0) + setformat(o, O_RDWR); /* need the channel, try to reopen */ + else if (o->cursound == -1 && o->owner == NULL) + setformat(o, O_CLOSE); /* can close */ + if (o->sounddev > -1) { + if (!o->owner) { /* no one owns the audio, so we must drain it */ + FD_SET(o->sounddev, &rfds); + maxfd = MAX(o->sounddev, maxfd); + } + if (o->cursound > -1) { + FD_SET(o->sounddev, &wfds); + maxfd = MAX(o->sounddev, maxfd); + } + } + /* ast_select emulates linux behaviour in terms of timeout handling */ + res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL); + if (res < 1) { + ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); + sleep(1); + continue; + } + if (FD_ISSET(o->sndcmd[0], &rfds)) { + /* read which sound to play from the pipe */ + int i, what = -1; + + if (read(o->sndcmd[0], &what, sizeof(what)) != sizeof(what)) { + ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno)); + continue; + } + for (i = 0; sounds[i].ind != -1; i++) { + if (sounds[i].ind == what) { + o->cursound = i; + o->sampsent = 0; + o->nosound = 1; /* block audio from pbx */ + break; + } + } + if (sounds[i].ind == -1) + ast_log(LOG_WARNING, "invalid sound index: %d\n", what); + } + if (o->sounddev > -1) { + if (FD_ISSET(o->sounddev, &rfds)) /* read and ignore errors */ + if (read(o->sounddev, ign, sizeof(ign)) < 0) { + } + if (FD_ISSET(o->sounddev, &wfds)) + send_sound(o); + } + } + return NULL; /* Never reached */ +} + +/* + * reset and close the device if opened, + * then open and initialize it in the desired mode, + * trigger reads and writes so we can start using it. + */ +static int setformat(struct chan_oss_pvt *o, int mode) +{ + int fmt, desired, res, fd; + + if (o->sounddev >= 0) { + ioctl(o->sounddev, SNDCTL_DSP_RESET, 0); + close(o->sounddev); + o->duplex = M_UNSET; + o->sounddev = -1; + } + if (mode == O_CLOSE) /* we are done */ + return 0; + if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000) + return -1; /* don't open too often */ + o->lastopen = ast_tvnow(); + fd = o->sounddev = open(o->device, mode | O_NONBLOCK); + if (fd < 0) { + ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", o->device, strerror(errno)); + return -1; + } + if (o->owner) + o->owner->fds[0] = fd; + +#if __BYTE_ORDER == __LITTLE_ENDIAN + fmt = AFMT_S16_LE; +#else + fmt = AFMT_S16_BE; +#endif + res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt); + if (res < 0) { + ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n"); + return -1; + } + switch (mode) { + case O_RDWR: + res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); + /* Check to see if duplex set (FreeBSD Bug) */ + res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt); + if (res == 0 && (fmt & DSP_CAP_DUPLEX)) { + if (option_verbose > 1) + ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n"); + o->duplex = M_FULL; + }; + break; + case O_WRONLY: + o->duplex = M_WRITE; + break; + case O_RDONLY: + o->duplex = M_READ; + break; + } + + fmt = 0; + res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt); + if (res < 0) { + ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); + return -1; + } + fmt = desired = DEFAULT_SAMPLE_RATE; /* 8000 Hz desired */ + res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt); + + if (res < 0) { + ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); + return -1; + } + if (fmt != desired) { + if (!(o->warned & WARN_speed)) { + ast_log(LOG_WARNING, + "Requested %d Hz, got %d Hz -- sound may be choppy\n", + desired, fmt); + o->warned |= WARN_speed; + } + } + /* + * on Freebsd, SETFRAGMENT does not work very well on some cards. + * Default to use 256 bytes, let the user override + */ + if (o->frags) { + fmt = o->frags; + res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt); + if (res < 0) { + if (!(o->warned & WARN_frag)) { + ast_log(LOG_WARNING, + "Unable to set fragment size -- sound may be choppy\n"); + o->warned |= WARN_frag; + } + } + } + /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */ + res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT; + res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res); + /* it may fail if we are in half duplex, never mind */ + return 0; +} + +/* + * some of the standard methods supported by channels. + */ +static int oss_digit_begin(struct ast_channel *c, char digit) +{ + return 0; +} + +static int oss_digit_end(struct ast_channel *c, char digit, unsigned int duration) +{ + /* no better use for received digits than print them */ + ast_verbose(" << Console Received digit %c of duration %u ms >> \n", + digit, duration); + return 0; +} + +static int oss_text(struct ast_channel *c, const char *text) +{ + /* print received messages */ + ast_verbose(" << Console Received text %s >> \n", text); + return 0; +} + +/* Play ringtone 'x' on device 'o' */ +static void ring(struct chan_oss_pvt *o, int x) +{ + if (write(o->sndcmd[1], &x, sizeof(x)) < 0) { + ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno)); + } +} + + +/* + * handler for incoming calls. Either autoanswer, or start ringing + */ +static int oss_call(struct ast_channel *c, char *dest, int timeout) +{ + struct chan_oss_pvt *o = c->tech_pvt; + struct ast_frame f = { 0, }; + + ast_verbose(" << Call to device '%s' dnid '%s' rdnis '%s' on console from '%s' <%s> >>\n", dest, c->cid.cid_dnid, c->cid.cid_rdnis, c->cid.cid_name, c->cid.cid_num); + if (o->autoanswer) { + ast_verbose(" << Auto-answered >> \n"); + f.frametype = AST_FRAME_CONTROL; + f.subclass = AST_CONTROL_ANSWER; + ast_queue_frame(c, &f); + } else { + ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); + f.frametype = AST_FRAME_CONTROL; + f.subclass = AST_CONTROL_RINGING; + ast_queue_frame(c, &f); + ring(o, AST_CONTROL_RING); + } + return 0; +} + +/* + * remote side answered the phone + */ +static int oss_answer(struct ast_channel *c) +{ + struct chan_oss_pvt *o = c->tech_pvt; + + ast_verbose(" << Console call has been answered >> \n"); +#if 0 + /* play an answer tone (XXX do we really need it ?) */ + ring(o, AST_CONTROL_ANSWER); +#endif + ast_setstate(c, AST_STATE_UP); + o->cursound = -1; + o->nosound = 0; + return 0; +} + +static int oss_hangup(struct ast_channel *c) +{ + struct chan_oss_pvt *o = c->tech_pvt; + + o->cursound = -1; + o->nosound = 0; + c->tech_pvt = NULL; + o->owner = NULL; + ast_verbose(" << Hangup on console >> \n"); + ast_module_unref(ast_module_info->self); + if (o->hookstate) { + if (o->autoanswer || o->autohangup) { + /* Assume auto-hangup too */ + o->hookstate = 0; + setformat(o, O_CLOSE); + } else { + /* Make congestion noise */ + ring(o, AST_CONTROL_CONGESTION); + } + } + return 0; +} + +/* used for data coming from the network */ +static int oss_write(struct ast_channel *c, struct ast_frame *f) +{ + int src; + struct chan_oss_pvt *o = c->tech_pvt; + + /* Immediately return if no sound is enabled */ + if (o->nosound) + return 0; + /* Stop any currently playing sound */ + o->cursound = -1; + /* + * we could receive a block which is not a multiple of our + * FRAME_SIZE, so buffer it locally and write to the device + * in FRAME_SIZE chunks. + * Keep the residue stored for future use. + */ + src = 0; /* read position into f->data */ + while (src < f->datalen) { + /* Compute spare room in the buffer */ + int l = sizeof(o->oss_write_buf) - o->oss_write_dst; + + if (f->datalen - src >= l) { /* enough to fill a frame */ + memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l); + soundcard_writeframe(o, (short *) o->oss_write_buf); + src += l; + o->oss_write_dst = 0; + } else { /* copy residue */ + l = f->datalen - src; + memcpy(o->oss_write_buf + o->oss_write_dst, f->data + src, l); + src += l; /* but really, we are done */ + o->oss_write_dst += l; + } + } + return 0; +} + +static struct ast_frame *oss_read(struct ast_channel *c) +{ + int res; + struct chan_oss_pvt *o = c->tech_pvt; + struct ast_frame *f = &o->read_f; + + /* XXX can be simplified returning &ast_null_frame */ + /* prepare a NULL frame in case we don't have enough data to return */ + bzero(f, sizeof(struct ast_frame)); + f->frametype = AST_FRAME_NULL; + f->src = oss_tech.type; + + res = read(o->sounddev, o->oss_read_buf + o->readpos, sizeof(o->oss_read_buf) - o->readpos); + if (res < 0) /* audio data not ready, return a NULL frame */ + return f; + + o->readpos += res; + if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */ + return f; + + if (o->mute) + return f; + + o->readpos = AST_FRIENDLY_OFFSET; /* reset read pointer for next frame */ + if (c->_state != AST_STATE_UP) /* drop data if frame is not up */ + return f; + /* ok we can build and deliver the frame to the caller */ + f->frametype = AST_FRAME_VOICE; + f->subclass = AST_FORMAT_SLINEAR; + f->samples = FRAME_SIZE; + f->datalen = FRAME_SIZE * 2; + f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET; + if (o->boost != BOOST_SCALE) { /* scale and clip values */ + int i, x; + int16_t *p = (int16_t *) f->data; + for (i = 0; i < f->samples; i++) { + x = (p[i] * o->boost) / BOOST_SCALE; + if (x > 32767) + x = 32767; + else if (x < -32768) + x = -32768; + p[i] = x; + } + } + + f->offset = AST_FRIENDLY_OFFSET; + return f; +} + +static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) +{ + struct chan_oss_pvt *o = newchan->tech_pvt; + o->owner = newchan; + return 0; +} + +static int oss_indicate(struct ast_channel *c, int cond, const void *data, size_t datalen) +{ + struct chan_oss_pvt *o = c->tech_pvt; + int res = -1; + + switch (cond) { + case AST_CONTROL_BUSY: + case AST_CONTROL_CONGESTION: + case AST_CONTROL_RINGING: + res = cond; + break; + + case -1: + o->cursound = -1; + o->nosound = 0; /* when cursound is -1 nosound must be 0 */ + return 0; + + case AST_CONTROL_VIDUPDATE: + res = -1; + break; + case AST_CONTROL_HOLD: + ast_verbose(" << Console Has Been Placed on Hold >> \n"); + ast_moh_start(c, data, o->mohinterpret); + break; + case AST_CONTROL_UNHOLD: + ast_verbose(" << Console Has Been Retrieved from Hold >> \n"); + ast_moh_stop(c); + break; + case AST_CONTROL_SRCUPDATE: + break; + default: + ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, c->name); + return -1; + } + + if (res > -1) + ring(o, res); + + return 0; +} + +/* + * allocate a new channel. + */ +static struct ast_channel *oss_new(struct chan_oss_pvt *o, char *ext, char *ctx, int state) +{ + struct ast_channel *c; + + c = ast_channel_alloc(1, state, o->cid_num, o->cid_name, "", ext, ctx, 0, "Console/%s", o->device + 5); + if (c == NULL) + return NULL; + c->tech = &oss_tech; + if (o->sounddev < 0) + setformat(o, O_RDWR); + c->fds[0] = o->sounddev; /* -1 if device closed, override later */ + c->nativeformats = AST_FORMAT_SLINEAR; + c->readformat = AST_FORMAT_SLINEAR; + c->writeformat = AST_FORMAT_SLINEAR; + c->tech_pvt = o; + + if (!ast_strlen_zero(o->language)) + ast_string_field_set(c, language, o->language); + /* Don't use ast_set_callerid() here because it will + * generate a needless NewCallerID event */ + c->cid.cid_ani = ast_strdup(o->cid_num); + if (!ast_strlen_zero(ext)) + c->cid.cid_dnid = ast_strdup(ext); + + o->owner = c; + ast_module_ref(ast_module_info->self); + ast_jb_configure(c, &global_jbconf); + if (state != AST_STATE_DOWN) { + if (ast_pbx_start(c)) { + ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name); + ast_hangup(c); + o->owner = c = NULL; + /* XXX what about the channel itself ? */ + /* XXX what about usecnt ? */ + } + } + + return c; +} + +static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause) +{ + struct ast_channel *c; + struct chan_oss_pvt *o = find_desc(data); + + ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", type, data, (char *) data); + if (o == NULL) { + ast_log(LOG_NOTICE, "Device %s not found\n", (char *) data); + /* XXX we could default to 'dsp' perhaps ? */ + return NULL; + } + if ((format & AST_FORMAT_SLINEAR) == 0) { + ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format); + return NULL; + } + if (o->owner) { + ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner); + *cause = AST_CAUSE_BUSY; + return NULL; + } + c = oss_new(o, NULL, NULL, AST_STATE_DOWN); + if (c == NULL) { + ast_log(LOG_WARNING, "Unable to create new OSS channel\n"); + return NULL; + } + return c; +} + +static int console_autoanswer_deprecated(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc == 1) { + ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); + return RESULT_SUCCESS; + } + if (argc != 2) + return RESULT_SHOWUSAGE; + if (o == NULL) { + ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n", oss_active); + return RESULT_FAILURE; + } + if (!strcasecmp(argv[1], "on")) + o->autoanswer = -1; + else if (!strcasecmp(argv[1], "off")) + o->autoanswer = 0; + else + return RESULT_SHOWUSAGE; + return RESULT_SUCCESS; +} + +static int console_autoanswer(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc == 2) { + ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); + return RESULT_SUCCESS; + } + if (argc != 3) + return RESULT_SHOWUSAGE; + if (o == NULL) { + ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n", + oss_active); + return RESULT_FAILURE; + } + if (!strcasecmp(argv[2], "on")) + o->autoanswer = -1; + else if (!strcasecmp(argv[2], "off")) + o->autoanswer = 0; + else + return RESULT_SHOWUSAGE; + return RESULT_SUCCESS; +} + +static char *autoanswer_complete_deprecated(const char *line, const char *word, int pos, int state) +{ + static char *choices[] = { "on", "off", NULL }; + + return (pos != 2) ? NULL : ast_cli_complete(word, choices, state); +} + +static char *autoanswer_complete(const char *line, const char *word, int pos, int state) +{ + static char *choices[] = { "on", "off", NULL }; + + return (pos != 3) ? NULL : ast_cli_complete(word, choices, state); +} + +static char autoanswer_usage[] = + "Usage: console autoanswer [on|off]\n" + " Enables or disables autoanswer feature. If used without\n" + " argument, displays the current on/off status of autoanswer.\n" + " The default value of autoanswer is in 'oss.conf'.\n"; + +/* + * answer command from the console + */ +static int console_answer_deprecated(int fd, int argc, char *argv[]) +{ + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 1) + return RESULT_SHOWUSAGE; + if (!o->owner) { + ast_cli(fd, "No one is calling us\n"); + return RESULT_FAILURE; + } + o->hookstate = 1; + o->cursound = -1; + o->nosound = 0; + ast_queue_frame(o->owner, &f); +#if 0 + /* XXX do we really need it ? considering we shut down immediately... */ + ring(o, AST_CONTROL_ANSWER); +#endif + return RESULT_SUCCESS; +} + +static int console_answer(int fd, int argc, char *argv[]) +{ + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 2) + return RESULT_SHOWUSAGE; + if (!o->owner) { + ast_cli(fd, "No one is calling us\n"); + return RESULT_FAILURE; + } + o->hookstate = 1; + o->cursound = -1; + o->nosound = 0; + ast_queue_frame(o->owner, &f); +#if 0 + /* XXX do we really need it ? considering we shut down immediately... */ + ring(o, AST_CONTROL_ANSWER); +#endif + return RESULT_SUCCESS; +} + +static char answer_usage[] = + "Usage: console answer\n" + " Answers an incoming call on the console (OSS) channel.\n"; + +/* + * concatenate all arguments into a single string. argv is NULL-terminated + * so we can use it right away + */ +static int console_sendtext_deprecated(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + char buf[TEXT_SIZE]; + + if (argc < 2) + return RESULT_SHOWUSAGE; + if (!o->owner) { + ast_cli(fd, "Not in a call\n"); + return RESULT_FAILURE; + } + ast_join(buf, sizeof(buf) - 1, argv + 2); + if (!ast_strlen_zero(buf)) { + struct ast_frame f = { 0, }; + int i = strlen(buf); + buf[i] = '\n'; + f.frametype = AST_FRAME_TEXT; + f.subclass = 0; + f.data = buf; + f.datalen = i + 1; + ast_queue_frame(o->owner, &f); + } + return RESULT_SUCCESS; +} + +static int console_sendtext(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + char buf[TEXT_SIZE]; + + if (argc < 3) + return RESULT_SHOWUSAGE; + if (!o->owner) { + ast_cli(fd, "Not in a call\n"); + return RESULT_FAILURE; + } + ast_join(buf, sizeof(buf) - 1, argv + 3); + if (!ast_strlen_zero(buf)) { + struct ast_frame f = { 0, }; + int i = strlen(buf); + buf[i] = '\n'; + f.frametype = AST_FRAME_TEXT; + f.subclass = 0; + f.data = buf; + f.datalen = i + 1; + ast_queue_frame(o->owner, &f); + } + return RESULT_SUCCESS; +} + +static char sendtext_usage[] = + "Usage: console send text <message>\n" + " Sends a text message for display on the remote terminal.\n"; + +static int console_hangup_deprecated(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 1) + return RESULT_SHOWUSAGE; + o->cursound = -1; + o->nosound = 0; + if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */ + ast_cli(fd, "No call to hang up\n"); + return RESULT_FAILURE; + } + o->hookstate = 0; + if (o->owner) + ast_queue_hangup(o->owner); + setformat(o, O_CLOSE); + return RESULT_SUCCESS; +} + +static int console_hangup(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 2) + return RESULT_SHOWUSAGE; + o->cursound = -1; + o->nosound = 0; + if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */ + ast_cli(fd, "No call to hang up\n"); + return RESULT_FAILURE; + } + o->hookstate = 0; + if (o->owner) + ast_queue_hangup(o->owner); + setformat(o, O_CLOSE); + return RESULT_SUCCESS; +} + +static char hangup_usage[] = + "Usage: console hangup\n" + " Hangs up any call currently placed on the console.\n"; + +static int console_flash_deprecated(int fd, int argc, char *argv[]) +{ + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH }; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 1) + return RESULT_SHOWUSAGE; + o->cursound = -1; + o->nosound = 0; /* when cursound is -1 nosound must be 0 */ + if (!o->owner) { /* XXX maybe !o->hookstate too ? */ + ast_cli(fd, "No call to flash\n"); + return RESULT_FAILURE; + } + o->hookstate = 0; + if (o->owner) /* XXX must be true, right ? */ + ast_queue_frame(o->owner, &f); + return RESULT_SUCCESS; +} + +static int console_flash(int fd, int argc, char *argv[]) +{ + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH }; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 2) + return RESULT_SHOWUSAGE; + o->cursound = -1; + o->nosound = 0; /* when cursound is -1 nosound must be 0 */ + if (!o->owner) { /* XXX maybe !o->hookstate too ? */ + ast_cli(fd, "No call to flash\n"); + return RESULT_FAILURE; + } + o->hookstate = 0; + if (o->owner) /* XXX must be true, right ? */ + ast_queue_frame(o->owner, &f); + return RESULT_SUCCESS; +} + +static char flash_usage[] = + "Usage: console flash\n" + " Flashes the call currently placed on the console.\n"; + +static int console_dial_deprecated(int fd, int argc, char *argv[]) +{ + char *s = NULL, *mye = NULL, *myc = NULL; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 1 && argc != 2) + return RESULT_SHOWUSAGE; + if (o->owner) { /* already in a call */ + int i; + struct ast_frame f = { AST_FRAME_DTMF, 0 }; + + if (argc == 1) { /* argument is mandatory here */ + ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n"); + return RESULT_FAILURE; + } + s = argv[1]; + /* send the string one char at a time */ + for (i = 0; i < strlen(s); i++) { + f.subclass = s[i]; + ast_queue_frame(o->owner, &f); + } + return RESULT_SUCCESS; + } + /* if we have an argument split it into extension and context */ + if (argc == 2) + s = ast_ext_ctx(argv[1], &mye, &myc); + /* supply default values if needed */ + if (mye == NULL) + mye = o->ext; + if (myc == NULL) + myc = o->ctx; + if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { + o->hookstate = 1; + oss_new(o, mye, myc, AST_STATE_RINGING); + } else + ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); + if (s) + free(s); + return RESULT_SUCCESS; +} + +static int console_dial(int fd, int argc, char *argv[]) +{ + char *s = NULL, *mye = NULL, *myc = NULL; + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc != 2 && argc != 3) + return RESULT_SHOWUSAGE; + if (o->owner) { /* already in a call */ + int i; + struct ast_frame f = { AST_FRAME_DTMF, 0 }; + + if (argc == 2) { /* argument is mandatory here */ + ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n"); + return RESULT_FAILURE; + } + s = argv[2]; + /* send the string one char at a time */ + for (i = 0; i < strlen(s); i++) { + f.subclass = s[i]; + ast_queue_frame(o->owner, &f); + } + return RESULT_SUCCESS; + } + /* if we have an argument split it into extension and context */ + if (argc == 3) + s = ast_ext_ctx(argv[2], &mye, &myc); + /* supply default values if needed */ + if (mye == NULL) + mye = o->ext; + if (myc == NULL) + myc = o->ctx; + if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { + o->hookstate = 1; + oss_new(o, mye, myc, AST_STATE_RINGING); + } else + ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); + if (s) + free(s); + return RESULT_SUCCESS; +} + +static char dial_usage[] = + "Usage: console dial [extension[@context]]\n" + " Dials a given extension (and context if specified)\n"; + +static int __console_mute_unmute(int mute) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + o->mute = mute; + return RESULT_SUCCESS; +} + +static int console_mute_deprecated(int fd, int argc, char *argv[]) +{ + if (argc != 1) + return RESULT_SHOWUSAGE; + + return __console_mute_unmute(1); +} + +static int console_mute(int fd, int argc, char *argv[]) +{ + if (argc != 2) + return RESULT_SHOWUSAGE; + + return __console_mute_unmute(1); +} + +static char mute_usage[] = + "Usage: console mute\nMutes the microphone\n"; + +static int console_unmute_deprecated(int fd, int argc, char *argv[]) +{ + if (argc != 1) + return RESULT_SHOWUSAGE; + + return __console_mute_unmute(0); +} + +static int console_unmute(int fd, int argc, char *argv[]) +{ + if (argc != 2) + return RESULT_SHOWUSAGE; + + return __console_mute_unmute(0); +} + +static char unmute_usage[] = + "Usage: console unmute\nUnmutes the microphone\n"; + +static int console_transfer_deprecated(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + struct ast_channel *b = NULL; + char *tmp, *ext, *ctx; + + if (argc != 2) + return RESULT_SHOWUSAGE; + if (o == NULL) + return RESULT_FAILURE; + if (o->owner ==NULL || (b = ast_bridged_channel(o->owner)) == NULL) { + ast_cli(fd, "There is no call to transfer\n"); + return RESULT_SUCCESS; + } + + tmp = ast_ext_ctx(argv[1], &ext, &ctx); + if (ctx == NULL) /* supply default context if needed */ + ctx = o->owner->context; + if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num)) + ast_cli(fd, "No such extension exists\n"); + else { + ast_cli(fd, "Whee, transferring %s to %s@%s.\n", + b->name, ext, ctx); + if (ast_async_goto(b, ctx, ext, 1)) + ast_cli(fd, "Failed to transfer :(\n"); + } + if (tmp) + free(tmp); + return RESULT_SUCCESS; +} + +static int console_transfer(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + struct ast_channel *b = NULL; + char *tmp, *ext, *ctx; + + if (argc != 3) + return RESULT_SHOWUSAGE; + if (o == NULL) + return RESULT_FAILURE; + if (o->owner == NULL || (b = ast_bridged_channel(o->owner)) == NULL) { + ast_cli(fd, "There is no call to transfer\n"); + return RESULT_SUCCESS; + } + + tmp = ast_ext_ctx(argv[2], &ext, &ctx); + if (ctx == NULL) /* supply default context if needed */ + ctx = o->owner->context; + if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num)) + ast_cli(fd, "No such extension exists\n"); + else { + ast_cli(fd, "Whee, transferring %s to %s@%s.\n", b->name, ext, ctx); + if (ast_async_goto(b, ctx, ext, 1)) + ast_cli(fd, "Failed to transfer :(\n"); + } + if (tmp) + free(tmp); + return RESULT_SUCCESS; +} + +static char transfer_usage[] = + "Usage: console transfer <extension>[@context]\n" + " Transfers the currently connected call to the given extension (and\n" + "context if specified)\n"; + +static int console_active_deprecated(int fd, int argc, char *argv[]) +{ + if (argc == 1) + ast_cli(fd, "active console is [%s]\n", oss_active); + else if (argc != 2) + return RESULT_SHOWUSAGE; + else { + struct chan_oss_pvt *o; + if (strcmp(argv[1], "show") == 0) { + for (o = oss_default.next; o; o = o->next) + ast_cli(fd, "device [%s] exists\n", o->name); + return RESULT_SUCCESS; + } + o = find_desc(argv[1]); + if (o == NULL) + ast_cli(fd, "No device [%s] exists\n", argv[1]); + else + oss_active = o->name; + } + return RESULT_SUCCESS; +} + +static int console_active(int fd, int argc, char *argv[]) +{ + if (argc == 2) + ast_cli(fd, "active console is [%s]\n", oss_active); + else if (argc != 3) + return RESULT_SHOWUSAGE; + else { + struct chan_oss_pvt *o; + if (strcmp(argv[2], "show") == 0) { + for (o = oss_default.next; o; o = o->next) + ast_cli(fd, "device [%s] exists\n", o->name); + return RESULT_SUCCESS; + } + o = find_desc(argv[2]); + if (o == NULL) + ast_cli(fd, "No device [%s] exists\n", argv[2]); + else + oss_active = o->name; + } + return RESULT_SUCCESS; +} + +static char active_usage[] = + "Usage: console active [device]\n" + " If used without a parameter, displays which device is the current\n" + "console. If a device is specified, the console sound device is changed to\n" + "the device specified.\n"; + +/* + * store the boost factor + */ +static void store_boost(struct chan_oss_pvt *o, char *s) +{ + double boost = 0; + if (sscanf(s, "%lf", &boost) != 1) { + ast_log(LOG_WARNING, "invalid boost <%s>\n", s); + return; + } + if (boost < -BOOST_MAX) { + ast_log(LOG_WARNING, "boost %s too small, using %d\n", s, -BOOST_MAX); + boost = -BOOST_MAX; + } else if (boost > BOOST_MAX) { + ast_log(LOG_WARNING, "boost %s too large, using %d\n", s, BOOST_MAX); + boost = BOOST_MAX; + } + boost = exp(log(10) * boost / 20) * BOOST_SCALE; + o->boost = boost; + ast_log(LOG_WARNING, "setting boost %s to %d\n", s, o->boost); +} + +static int do_boost(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + if (argc == 2) + ast_cli(fd, "boost currently %5.1f\n", 20 * log10(((double) o->boost / (double) BOOST_SCALE))); + else if (argc == 3) + store_boost(o, argv[2]); + return RESULT_SUCCESS; +} + +static struct ast_cli_entry cli_oss_answer_deprecated = { + { "answer", NULL }, + console_answer_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_hangup_deprecated = { + { "hangup", NULL }, + console_hangup_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_flash_deprecated = { + { "flash", NULL }, + console_flash_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_dial_deprecated = { + { "dial", NULL }, + console_dial_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_mute_deprecated = { + { "mute", NULL }, + console_mute_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_unmute_deprecated = { + { "unmute", NULL }, + console_unmute_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_transfer_deprecated = { + { "transfer", NULL }, + console_transfer_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_send_text_deprecated = { + { "send", "text", NULL }, + console_sendtext_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_autoanswer_deprecated = { + { "autoanswer", NULL }, + console_autoanswer_deprecated, NULL, + NULL, autoanswer_complete_deprecated }; + +static struct ast_cli_entry cli_oss_boost_deprecated = { + { "oss", "boost", NULL }, + do_boost, NULL, + NULL }; + +static struct ast_cli_entry cli_oss_active_deprecated = { + { "console", NULL }, + console_active_deprecated, NULL, + NULL }; + +static struct ast_cli_entry cli_oss[] = { + { { "console", "answer", NULL }, + console_answer, "Answer an incoming console call", + answer_usage, NULL, &cli_oss_answer_deprecated }, + + { { "console", "hangup", NULL }, + console_hangup, "Hangup a call on the console", + hangup_usage, NULL, &cli_oss_hangup_deprecated }, + + { { "console", "flash", NULL }, + console_flash, "Flash a call on the console", + flash_usage, NULL, &cli_oss_flash_deprecated }, + + { { "console", "dial", NULL }, + console_dial, "Dial an extension on the console", + dial_usage, NULL, &cli_oss_dial_deprecated }, + + { { "console", "mute", NULL }, + console_mute, "Disable mic input", + mute_usage, NULL, &cli_oss_mute_deprecated }, + + { { "console", "unmute", NULL }, + console_unmute, "Enable mic input", + unmute_usage, NULL, &cli_oss_unmute_deprecated }, + + { { "console", "transfer", NULL }, + console_transfer, "Transfer a call to a different extension", + transfer_usage, NULL, &cli_oss_transfer_deprecated }, + + { { "console", "send", "text", NULL }, + console_sendtext, "Send text to the remote device", + sendtext_usage, NULL, &cli_oss_send_text_deprecated }, + + { { "console", "autoanswer", NULL }, + console_autoanswer, "Sets/displays autoanswer", + autoanswer_usage, autoanswer_complete, &cli_oss_autoanswer_deprecated }, + + { { "console", "boost", NULL }, + do_boost, "Sets/displays mic boost in dB", + NULL, NULL, &cli_oss_boost_deprecated }, + + { { "console", "active", NULL }, + console_active, "Sets/displays active console", + active_usage, NULL, &cli_oss_active_deprecated }, +}; + +/* + * store the mixer argument from the config file, filtering possibly + * invalid or dangerous values (the string is used as argument for + * system("mixer %s") + */ +static void store_mixer(struct chan_oss_pvt *o, char *s) +{ + int i; + + for (i = 0; i < strlen(s); i++) { + if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) { + ast_log(LOG_WARNING, "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s); + return; + } + } + if (o->mixer_cmd) + free(o->mixer_cmd); + o->mixer_cmd = ast_strdup(s); + ast_log(LOG_WARNING, "setting mixer %s\n", s); +} + +/* + * store the callerid components + */ +static void store_callerid(struct chan_oss_pvt *o, char *s) +{ + ast_callerid_split(s, o->cid_name, sizeof(o->cid_name), o->cid_num, sizeof(o->cid_num)); +} + +/* + * grab fields from the config file, init the descriptor and open the device. + */ +static struct chan_oss_pvt *store_config(struct ast_config *cfg, char *ctg) +{ + struct ast_variable *v; + struct chan_oss_pvt *o; + + if (ctg == NULL) { + o = &oss_default; + ctg = "general"; + } else { + if (!(o = ast_calloc(1, sizeof(*o)))) + return NULL; + *o = oss_default; + /* "general" is also the default thing */ + if (strcmp(ctg, "general") == 0) { + o->name = ast_strdup("dsp"); + oss_active = o->name; + goto openit; + } + o->name = ast_strdup(ctg); + } + + strcpy(o->mohinterpret, "default"); + + o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */ + /* fill other fields from configuration */ + for (v = ast_variable_browse(cfg, ctg); v; v = v->next) { + M_START(v->name, v->value); + + /* handle jb conf */ + if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) + continue; + + M_BOOL("autoanswer", o->autoanswer) + M_BOOL("autohangup", o->autohangup) + M_BOOL("overridecontext", o->overridecontext) + M_STR("device", o->device) + M_UINT("frags", o->frags) + M_UINT("debug", oss_debug) + M_UINT("queuesize", o->queuesize) + M_STR("context", o->ctx) + M_STR("language", o->language) + M_STR("mohinterpret", o->mohinterpret) + M_STR("extension", o->ext) + M_F("mixer", store_mixer(o, v->value)) + M_F("callerid", store_callerid(o, v->value)) + M_F("boost", store_boost(o, v->value)) + M_END(; + ); + } + if (ast_strlen_zero(o->device)) + ast_copy_string(o->device, DEV_DSP, sizeof(o->device)); + if (o->mixer_cmd) { + char *cmd; + + 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)); + } + free(cmd); + } + } + if (o == &oss_default) /* we are done with the default */ + return NULL; + + openit: +#if TRYOPEN + if (setformat(o, O_RDWR) < 0) { /* open device */ + if (option_verbose > 0) { + ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg); + ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding " "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n"); + } + goto error; + } + if (o->duplex != M_FULL) + ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n"); +#endif /* TRYOPEN */ + if (pipe(o->sndcmd) != 0) { + ast_log(LOG_ERROR, "Unable to create pipe\n"); + goto error; + } + ast_pthread_create_background(&o->sthread, NULL, sound_thread, o); + /* link into list of devices */ + if (o != &oss_default) { + o->next = oss_default.next; + oss_default.next = o; + } + return o; + + error: + if (o != &oss_default) + free(o); + return NULL; +} + +static int load_module(void) +{ + struct ast_config *cfg = NULL; + char *ctg = NULL; + + /* Copy the default jb config over global_jbconf */ + memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf)); + + /* load config file */ + if (!(cfg = ast_config_load(config))) { + ast_log(LOG_NOTICE, "Unable to load config %s\n", config); + return AST_MODULE_LOAD_DECLINE; + } + + do { + store_config(cfg, ctg); + } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL); + + ast_config_destroy(cfg); + + if (find_desc(oss_active) == NULL) { + ast_log(LOG_NOTICE, "Device %s not found\n", oss_active); + /* XXX we could default to 'dsp' perhaps ? */ + /* XXX should cleanup allocated memory etc. */ + return AST_MODULE_LOAD_FAILURE; + } + + if (ast_channel_register(&oss_tech)) { + ast_log(LOG_ERROR, "Unable to register channel type 'OSS'\n"); + return AST_MODULE_LOAD_FAILURE; + } + + ast_cli_register_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry)); + + return AST_MODULE_LOAD_SUCCESS; +} + + +static int unload_module(void) +{ + struct chan_oss_pvt *o; + + ast_channel_unregister(&oss_tech); + ast_cli_unregister_multiple(cli_oss, sizeof(cli_oss) / sizeof(struct ast_cli_entry)); + + for (o = oss_default.next; o; o = o->next) { + close(o->sounddev); + if (o->sndcmd[0] > 0) { + close(o->sndcmd[0]); + close(o->sndcmd[1]); + } + if (o->owner) + ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD); + if (o->owner) /* XXX how ??? */ + return -1; + /* XXX what about the thread ? */ + /* XXX what about the memory allocated ? */ + } + return 0; +} + +AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "OSS Console Channel Driver"); |