/* * Asterisk -- A telephony toolkit for Linux. * * Copyright (C) 1999 - 2005, Digium, Inc. * * Mark Spencer * * This program is free software, distributed under the terms of * the GNU General Public License * * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.04 * note-this code best seen with ts=8 (8-spaces tabs) in the editor */ #include "asterisk/lock.h" #include "asterisk/frame.h" #include "asterisk/logger.h" #include "asterisk/channel.h" #include "asterisk/module.h" #include "asterisk/options.h" #include "asterisk/pbx.h" #include "asterisk/config.h" /* * 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, WITHOUT ; on each line. * * { * 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) */ #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, strncpy(dst, __val, sizeof(dst) - 1) ) #include "asterisk/cli.h" #include "asterisk/utils.h" #include "asterisk/causes.h" #include "asterisk/endian.h" #include #include #include #include #include #include #include #include #include /* for isalnum */ #ifdef __linux #include #elif defined(__FreeBSD__) #include #else #include #endif #include "busy.h" #include "ringtone.h" #include "ring10.h" #include "answer.h" /* Which device to use */ #if defined( __OpenBSD__ ) || defined( __NetBSD__ ) #define DEV_DSP "/dev/audio" #else #define DEV_DSP "/dev/dsp" #endif /* * Basic mode of operation: * * we have one keyboard (which receives commands from the keyboard) * and multiple headset's connected to audio cards. Headsets are named as * the sections of oss.conf * * At any time, the keyboard is attached to one headset, and you * can switch among them using the 'console' command. * * The following parameters are important for the configuration of * the device: * * 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 /* Don't switch between read/write modes faster than every 300 ms */ #define MIN_SWITCH_TIME 300 static int usecnt; AST_MUTEX_DEFINE_STATIC(usecnt_lock); 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 *type; /* XXX maybe take the one from oss_tech */ 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; struct timeval lasttime; /* last setformat */ 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 */ int silencesuppression; int silencethreshold; int playbackonly; char device[64]; /* device to open */ pthread_t sthread; struct ast_channel *owner; char ext[AST_MAX_EXTENSION]; char ctx[AST_MAX_EXTENSION]; char language[MAX_LANGUAGE]; /* 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 = { .type = "Console", .cursound = -1, .sounddev = -1, .duplex = M_UNSET, /* XXX check this */ .autoanswer = 1, .autohangup = 1, .queuesize = QUEUE_SIZE, .frags = FRAGS, .silencethreshold = 1000, /* currently unused */ .ext = "s", .ctx = "default", .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */ }; static char *oss_active; /* the active device */ static struct ast_channel *oss_request(const char *type, int format, void *data , int *cause); static int oss_digit(struct ast_channel *c, char digit); static int oss_text(struct ast_channel *c, const char *text); static int oss_hangup(struct ast_channel *c); static int oss_answer(struct ast_channel *c); static struct ast_frame *oss_read(struct ast_channel *chan); static int oss_call(struct ast_channel *c, char *dest, int timeout); static int oss_write(struct ast_channel *chan, struct ast_frame *f); static int oss_indicate(struct ast_channel *chan, int cond); static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); static const struct ast_channel_tech oss_tech = { .type = "Console", .description = "OSS Console Channel Driver", .capabilities = AST_FORMAT_SLINEAR, .requester = oss_request, .send_digit = oss_digit, .send_text = oss_text, .hangup = oss_hangup, .answer = oss_answer, .read = oss_read, .call = oss_call, .write = oss_write, .indicate = oss_indicate, .fixup = oss_fixup, }; /* * returns true if too early to switch */ static int too_early(struct chan_oss_pvt *o) { struct timeval tv; int ms; gettimeofday(&tv, NULL); ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 + (tv.tv_usec - o->lasttime.tv_usec) / 1000; if (ms < MIN_SWITCH_TIME) return -1; return 0; } /* * 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; } static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) { /* Write an exactly FRAME_SIZE sized frame */ int res; /* * 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; res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2); return res; } /* * 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; /* sound available */ 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 { /* no sound, maybe some silence */ static short silence[FRAME_SIZE] = {0, }; l += s->silencelen; if (l > 0) { if (l > FRAME_SIZE - ofs) l = FRAME_SIZE - ofs; bcopy(silence, myframe + ofs, l*2); l_sampsent += l; } else { /* silence is over, restart sound if loop */ if (s->repeat == 0) { /* last block */ o->cursound = -1; o->nosound = 0; /* allow audio data */ if (ofs < FRAME_SIZE) /* pad with silence */ bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2); } l_sampsent = 0; } } } l = soundcard_writeframe(o, myframe); if (l > 0) o->sampsent = l_sampsent; /* update status */ } static void *sound_thread(void *arg) { char ign[4096]; struct chan_oss_pvt *o = (struct chan_oss_pvt *)arg; /* * Just in case, kick the driver by trying to read from it. * Ignore errors - this read is almost guaranteed to fail. */ read(o->sounddev, ign, sizeof(ign)); for(;;) { fd_set rfds, wfds; int maxfd, res; FD_ZERO(&rfds); FD_ZERO(&wfds); maxfd = o->sndcmd[0]; /* pipe from the main process */ FD_SET(o->sndcmd[0], &rfds); if (!o->owner) { /* no one owns the audio, so we must drain it */ FD_SET(o->sounddev, &rfds); if (o->sounddev > maxfd) maxfd = o->sounddev; } if (o->cursound > -1) { FD_SET(o->sounddev, &wfds); if (o->sounddev > maxfd) maxfd = o->sounddev; } /* 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)); continue; } if (FD_ISSET(o->sndcmd[0], &rfds)) { /* read which sound to play from the pipe */ int i, what = -1; read(o->sndcmd[0], &what, sizeof(what)); for (i = 0; sounds[i].ind != -1; i++) { if (sounds[i].ind == what) { o->cursound = i; o->sampsent = 0; o->nosound = 1; /* block audio from pbx */ break; } } if (sounds[i].ind == -1) ast_log(LOG_WARNING, "invalid sound index: %d\n", what); } if (FD_ISSET(o->sounddev, &rfds)) { /* read and ignore errors */ read(o->sounddev, ign, sizeof(ign)); } if (FD_ISSET(o->sounddev, &wfds)) send_sound(o); } /* Never reached */ return NULL; } #if 0 static int calc_loudness(short *frame) { int sum = 0; int x; for (x=0;x= SILBUF) { /* Make way for more buffer */ memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1)); silbufcnt--; } memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2); if (silentframes > 10) { /* We've had plenty of silence, so compress it now */ return 1; } } else { silentframes=0; /* Write any buffered silence we have, it may have something important */ if (silbufcnt) { write(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE); silbufcnt = 0; } } return 0; } #endif /* * 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; } fd = o->sounddev = open(o->device, mode |O_NONBLOCK); if (o->sounddev < 0) { ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno)); return -1; } gettimeofday(&o->lasttime, NULL); #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; } /* 8000 Hz desired */ desired = 8000; fmt = desired; res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt); if (res < 0) { ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); return -1; } if (fmt != desired) { if (!(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; } } } /* XXX 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; } /* * make sure output mode is available. Returns 0 if done, * 1 if too early to switch, -1 if error */ static int soundcard_setoutput(struct chan_oss_pvt *o, int force) { if (o->duplex == M_FULL || (o->duplex == M_WRITE && !force)) return 0; if (!force && too_early(o)) return 1; if (setformat(o, O_WRONLY)) return -1; return 0; } /* * make sure input mode is available. Returns 0 if done * 1 if too early to switch, -1 if error */ static int soundcard_setinput(struct chan_oss_pvt *o, int force) { if (o->duplex == M_FULL || (o->duplex == M_READ && !force)) return 0; if (!force && too_early(o)) return 1; if (setformat(o, O_RDONLY)) return -1; return 0; } static int oss_digit(struct ast_channel *c, char digit) { ast_verbose( " << Console Received digit %c >> \n", digit); return 0; } static int oss_text(struct ast_channel *c, const char *text) { ast_verbose( " << Console Received text %s >> \n", text); return 0; } /* request to play a sound on the speaker XXX fix oss. */ #define RING(o, x) { int what = x; write((o)->sndcmd[1], &what, sizeof(what)); } 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 placed to '%s' on console >> \n", dest); 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; } static void answer_sound(struct chan_oss_pvt *o) { RING(o, AST_CONTROL_ANSWER); } static int oss_answer(struct ast_channel *c) { struct chan_oss_pvt *o = c->tech_pvt; ast_verbose( " << Console call has been answered >> \n"); answer_sound(o); /* XXX do we really need it ? considering we shut down immediately... */ 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; c->tech_pvt = NULL; o->owner = NULL; ast_verbose( " << Hangup on console >> \n"); ast_mutex_lock(&usecnt_lock); /* XXX not sure why */ usecnt--; ast_mutex_unlock(&usecnt_lock); if (o->hookstate) { if (o->autoanswer || o->autohangup) { /* Assume auto-hangup too */ o->hookstate = 0; } 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 res; 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; if (o->duplex != M_FULL && !o->playbackonly) { /* XXX check this, looks weird! */ /* If we're half duplex, we have to switch to read mode to honor immediate needs if necessary */ res = soundcard_setinput(o, 1); /* force set if not full_duplex */ if (res < 0) { ast_log(LOG_WARNING, "Unable to set device to input mode\n"); return -1; } return 0; } res = soundcard_setoutput(o, 0); if (res < 0) { ast_log(LOG_WARNING, "Unable to set output device\n"); return -1; } else if (res > 0) { /* The device is still in read mode, and it's too soon to change it, so just pretend we wrote it */ return 0; } /* * we could receive a sample which is not a multiple of our FRAME_SIZE, * so we buffer it locally and write to the device in FRAME_SIZE * chunks, keeping 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; /* 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 = o->type; res = soundcard_setinput(o, 0); if (res < 0) { ast_log(LOG_WARNING, "Unable to set input mode\n"); return NULL; } else if (res > 0) { /* too early to switch ? */ /* Theoretically shouldn't happen, but anyway, return a NULL frame */ return f; } 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; 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; 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) { struct chan_oss_pvt *o = c->tech_pvt; int res; switch(cond) { case AST_CONTROL_BUSY: case AST_CONTROL_CONGESTION: case AST_CONTROL_RINGING: res = cond; break; case -1: o->cursound = -1; return 0; 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; } 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); if (c == NULL) return NULL; c->tech = &oss_tech; snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5); c->type = o->type; c->fds[0] = o->sounddev; c->nativeformats = AST_FORMAT_SLINEAR; c->readformat = AST_FORMAT_SLINEAR; c->writeformat = AST_FORMAT_SLINEAR; c->tech_pvt = o; #define S_OVERRIDE(dst, src) \ { if (src && strlen(src) != 0) \ strncpy((dst), src, sizeof(dst)-1); } S_OVERRIDE(c->context, ctx); S_OVERRIDE(c->exten, ext); S_OVERRIDE(c->language, o->language); o->owner = c; ast_setstate(c, state); ast_mutex_lock(&usecnt_lock); usecnt++; ast_mutex_unlock(&usecnt_lock); ast_update_use_count(); if (state != AST_STATE_DOWN) { if (ast_pbx_start(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; } /* * returns a pointer to the descriptor with the given name */ static struct chan_oss_pvt *find_desc(char *dev) { struct chan_oss_pvt *o; for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next) ; if (o == NULL) ast_log(LOG_WARNING, "%s could not find <%s>\n", __func__, dev); return o; } 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 on the OSS channel\n"); *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(int fd, int argc, char *argv[]) { struct chan_oss_pvt *o = find_desc(oss_active); if ((argc != 1) && (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 (argc == 1) { ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); return RESULT_SUCCESS; } 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 char *autoanswer_complete(char *line, char *word, int pos, int state) { #ifndef MIN #define MIN(a,b) ((a) < (b) ? (a) : (b)) #endif int l = strlen(word); switch(state) { case 0: if (l && !strncasecmp(word, "on", MIN(l, 2))) return strdup("on"); case 1: if (l && !strncasecmp(word, "off", MIN(l, 3))) return strdup("off"); default: return NULL; } return NULL; } static char autoanswer_usage[] = "Usage: autoanswer [on|off]\n" " Enables or disables autoanswer feature. If used without\n" " argument, displays the current on/off status of autoanswer.\n" " The default value of autoanswer is in 'oss.conf'.\n"; static int console_answer(int fd, int argc, char *argv[]) { struct chan_oss_pvt *o = find_desc(oss_active); struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; 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; ast_queue_frame(o->owner, &f); answer_sound(o); return RESULT_SUCCESS; } static char sendtext_usage[] = "Usage: send text \n" " Sends a text message for display on the remote terminal.\n"; static int console_sendtext(int fd, int argc, char *argv[]) { struct chan_oss_pvt *o = find_desc(oss_active); int tmparg = 2; char text2send[256] = ""; struct ast_frame f = { 0, }; if (argc < 2) return RESULT_SHOWUSAGE; if (!o->owner) { ast_cli(fd, "No one is calling us\n"); return RESULT_FAILURE; } if (strlen(text2send)) ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n"); text2send[0] = '\0'; while(tmparg < argc) { strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); } if (strlen(text2send)) { f.frametype = AST_FRAME_TEXT; f.subclass = 0; f.data = text2send; f.datalen = strlen(text2send); ast_queue_frame(o->owner, &f); } return RESULT_SUCCESS; } static char answer_usage[] = "Usage: answer\n" " Answers an incoming call on the console (OSS) channel.\n"; static int console_hangup(int fd, int argc, char *argv[]) { struct chan_oss_pvt *o = find_desc(oss_active); if (argc != 1) return RESULT_SHOWUSAGE; o->cursound = -1; if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */ ast_cli(fd, "No call to hangup up\n"); return RESULT_FAILURE; } o->hookstate = 0; if (o->owner) { ast_queue_hangup(o->owner); } return RESULT_SUCCESS; } static char hangup_usage[] = "Usage: hangup\n" " Hangs up any call currently placed on the console.\n"; 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 != 1) return RESULT_SHOWUSAGE; o->cursound = -1; 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: flash\n" " Flashes the call currently placed on the console.\n"; static int console_dial(int fd, int argc, char *argv[]) { char *tmp = NULL, *mye = NULL, *myc = NULL; int i; struct ast_frame f = { AST_FRAME_DTMF, 0 }; struct chan_oss_pvt *o = find_desc(oss_active); if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; if (o->owner) { /* already in a call */ 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; } mye = argv[1]; /* send the string one char at a time */ for (i=0; iowner, &f); } return RESULT_SUCCESS; } /* if we have an argument split it into extension and context */ if (argc == 2) { tmp = myc = strdup(argv[1]); /* make a writable copy */ mye = strsep(&myc, "@"); /* set exten, advance to context */ myc = strsep(&myc, "@"); /* set context */ } /* 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); return RESULT_SUCCESS; } static char dial_usage[] = "Usage: dial [extension[@context]]\n" " Dials a given extensison (and context if specified)\n"; static int console_transfer(int fd, int argc, char *argv[]) { struct chan_oss_pvt *o = find_desc(oss_active); struct ast_channel *b = NULL; char *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; } ext = ctx = strdup(argv[1]); /* make a writable copy */ strsep(&ctx, "@"); /* set exten, advance to context */ ctx = strsep(&ctx, "@"); /* strip trailing @ and the rest */ 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"); } free(ext); return RESULT_SUCCESS; } static char transfer_usage[] = "Usage: transfer [@context]\n" " Transfers the currently connected call to the given extension (and\n" "context if specified)\n"; static int console_active(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 struct ast_cli_entry myclis[] = { { { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage }, { { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage }, { { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage }, { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage }, { { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_usage }, { { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage }, { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }, { { "console", NULL }, console_active, "Sets/displays active console", "console foo sets foo as the 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, 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 = strdup(s); ast_log(LOG_WARNING, "setting mixer %s\n", s); } /* * 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; o->next = NULL; /* XXX needed ? */ ctg = "general"; } else { o = (struct chan_oss_pvt *)malloc(sizeof *o); if (o == NULL) /* fail */ return NULL; *o = oss_default; /* "general" is also the default thing */ if (strcmp(ctg, "general") == 0) { o->name = strdup("dsp"); oss_active = o->name; goto openit; } o->name = strdup(ctg); } ast_log(LOG_WARNING, "found category [%s]\n", ctg); /* fill other fields from configuration */ v = ast_variable_browse(cfg, ctg); while(v) { M_START(v->name, v->value); M_BOOL("autoanswer", o->autoanswer) M_BOOL("autohangup", o->autohangup) M_BOOL("playbackonly", o->playbackonly) M_BOOL("silencesuppression", o->silencesuppression) M_UINT("silencethreshold", o->silencethreshold ) 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("extension", o->ext) M_F("mixer", store_mixer(o, v->value)) M_END(;); v=v->next; } if (!strlen(o->device)) strncpy(o->device, DEV_DSP, sizeof(o->device)-1); if (o->mixer_cmd) { char *cmd; asprintf(&cmd, "mixer %s", o->mixer_cmd); ast_log(LOG_WARNING, "running [%s]\n", cmd); system(cmd); free(cmd); } if (o == &oss_default) /* we are done with the default */ return NULL; openit: if (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; } soundcard_setinput(o, 1); /* force set if not full_duplex */ if (o->duplex != M_FULL) ast_log(LOG_WARNING, "XXX I don't work right with non " "full-duplex sound cards XXX\n"); if ( pipe(o->sndcmd) != 0 ) { ast_log(LOG_ERROR, "Unable to create pipe\n"); goto error; } ast_pthread_create(&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; } int load_module() { int i; struct ast_config *cfg; struct chan_oss_pvt *o; /* load config file */ cfg = ast_config_load(config); if (cfg != NULL) { char *ctg; store_config(cfg, NULL); /* init general category */ ctg = ast_category_browse(cfg, NULL); /* initial category */ while (ctg != NULL) { store_config(cfg, ctg); ctg = ast_category_browse(cfg, ctg); } ast_config_destroy(cfg); } i = ast_channel_register(&oss_tech); if (i < 0) { ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", oss_default.type); /* XXX should cleanup allocated memory etc. */ return -1; } o = find_desc(oss_active); if (o == 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 -1; } for (i=0; inext) { 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; } char *description() { return (char *)oss_tech.description; } int usecount() /* XXX is this per-device or global for the module ? */ { int res; ast_mutex_lock(&usecnt_lock); res = usecnt; ast_mutex_unlock(&usecnt_lock); return res; } char *key() { return ASTERISK_GPL_KEY; }