/* * Asterisk -- A telephony toolkit for Linux. * * Copyright (C) 2002, Linux Support Services * * By Matthew Fredrickson * * This program is free software, distributed under the terms of * the GNU General Public License */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include #include "busy.h" #include "ringtone.h" #include "ring10.h" #include "answer.h" #ifdef ALSA_MONITOR #include "alsa-monitor.h" #endif #define DEBUG 0 /* Which device to use */ #define ALSA_INDEV "default" #define ALSA_OUTDEV "default" #define DESIRED_RATE 8000 /* Lets use 160 sample frames, just like GSM. */ #define FRAME_SIZE 160 #define PERIOD_FRAMES 80 /* 80 Frames, at 2 bytes each */ /* When you set the frame size, you have to come up with the right buffer format as well. */ /* 5 64-byte frames = one frame */ #define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006); /* Don't switch between read/write modes faster than every 300 ms */ #define MIN_SWITCH_TIME 600 static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; //static int block = O_NONBLOCK; static char indevname[50] = ALSA_INDEV; static char outdevname[50] = ALSA_OUTDEV; #if 0 static struct timeval lasttime; #endif static int usecnt; static int needanswer = 0; static int needringing = 0; static int needhangup = 0; static int silencesuppression = 0; static int silencethreshold = 1000; static char digits[80] = ""; static char text2send[80] = ""; static ast_mutex_t usecnt_lock = AST_MUTEX_INITIALIZER; static char *type = "Console"; static char *desc = "ALSA Console Channel Driver"; static char *tdesc = "ALSA Console Channel Driver"; static char *config = "alsa.conf"; static char context[AST_MAX_EXTENSION] = "default"; static char language[MAX_LANGUAGE] = ""; static char exten[AST_MAX_EXTENSION] = "s"; /* Command pipe */ static int cmd[2]; int hookstate=0; static short silence[FRAME_SIZE] = {0, }; struct sound { int ind; short *data; int datalen; int samplen; int silencelen; int repeat; }; static struct sound sounds[] = { { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 }, { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 }, { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 }, { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 }, }; /* Sound command pipe */ static int sndcmd[2]; static struct chan_alsa_pvt { /* We only have one ALSA structure -- near sighted perhaps, but it keeps this driver as simple as possible -- as it should be. */ struct ast_channel *owner; char exten[AST_MAX_EXTENSION]; char context[AST_MAX_EXTENSION]; #if 0 snd_pcm_t *card; #endif snd_pcm_t *icard, *ocard; } alsa; #if 0 static int time_has_passed(void) { struct timeval tv; int ms; gettimeofday(&tv, NULL); ms = (tv.tv_sec - lasttime.tv_sec) * 1000 + (tv.tv_usec - lasttime.tv_usec) / 1000; if (ms > MIN_SWITCH_TIME) return -1; return 0; } #endif /* Number of buffers... Each is FRAMESIZE/8 ms long. For example with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, usually plenty. */ pthread_t sthread; #define MAX_BUFFER_SIZE 100 //static int buffersize = 3; //static int full_duplex = 0; /* Are we reading or writing (simulated full duplex) */ //static int readmode = 1; /* File descriptors for sound device */ static int readdev = -1; static int writedev = -1; static int autoanswer = 1; #if 0 static int calc_loudness(short *frame) { int sum = 0; int x; for (x=0;x -1) { res = total; if (sampsent < sounds[cursound].samplen) { myoff=0; while(total) { amt = total; if (amt > (sounds[cursound].datalen - offset)) amt = sounds[cursound].datalen - offset; memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2); total -= amt; offset += amt; sampsent += amt; myoff += amt; if (offset >= sounds[cursound].datalen) offset = 0; } /* Set it up for silence */ if (sampsent >= sounds[cursound].samplen) silencelen = sounds[cursound].silencelen; frame = myframe; } else { if (silencelen > 0) { frame = silence; silencelen -= res; } else { if (sounds[cursound].repeat) { /* Start over */ sampsent = 0; offset = 0; } else { cursound = -1; nosound = 0; } return 0; } } if (res == 0 || !frame) { return 0; } #ifdef ALSA_MONITOR alsa_monitor_write((char *)frame, res * 2); #endif state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) { snd_pcm_prepare(alsa.ocard); } res = snd_pcm_writei(alsa.ocard, frame, res); if (res > 0) return 0; return 0; } return 0; } static void *sound_thread(void *unused) { fd_set rfds; fd_set wfds; int max; int res; for(;;) { FD_ZERO(&rfds); FD_ZERO(&wfds); max = sndcmd[0]; FD_SET(sndcmd[0], &rfds); if (cursound > -1) { FD_SET(writedev, &wfds); if (writedev > max) max = writedev; } #ifdef ALSA_MONITOR if (!alsa.owner) { FD_SET(readdev, &rfds); if (readdev > max) max = readdev; } #endif res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); if (res < 1) { ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); continue; } #ifdef ALSA_MONITOR if (FD_ISSET(readdev, &rfds)) { /* Keep the pipe going with read audio */ snd_pcm_state_t state; short buf[FRAME_SIZE]; int r; state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) { snd_pcm_prepare(alsa.ocard); } r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE); if (r == -EPIPE) { #if DEBUG ast_log(LOG_ERROR, "XRUN read\n"); #endif snd_pcm_prepare(alsa.icard); } else if (r == -ESTRPIPE) { ast_log(LOG_ERROR, "-ESTRPIPE\n"); snd_pcm_prepare(alsa.icard); } else if (r < 0) { ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r)); } else alsa_monitor_read((char *)buf, r * 2); } #endif if (FD_ISSET(sndcmd[0], &rfds)) { read(sndcmd[0], &cursound, sizeof(cursound)); silencelen = 0; offset = 0; sampsent = 0; } if (FD_ISSET(writedev, &wfds)) if (send_sound()) ast_log(LOG_WARNING, "Failed to write sound\n"); } /* Never reached */ return NULL; } static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream) { int err; int direction; snd_pcm_t *handle = NULL; snd_pcm_hw_params_t *hwparams = NULL; snd_pcm_sw_params_t *swparams = NULL; struct pollfd pfd; snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4; //int period_bytes = 0; snd_pcm_uframes_t buffer_size = 0; unsigned int rate = DESIRED_RATE; unsigned int per_min = 1; //unsigned int per_max = 8; snd_pcm_uframes_t start_threshold, stop_threshold; err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK); if (err < 0) { ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err)); return NULL; } else { ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write"); } snd_pcm_hw_params_alloca(&hwparams); snd_pcm_hw_params_any(handle, hwparams); err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err)); } err = snd_pcm_hw_params_set_format(handle, hwparams, format); if (err < 0) { ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err)); } err = snd_pcm_hw_params_set_channels(handle, hwparams, 1); if (err < 0) { ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err)); } direction = 0; err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction); if (rate != DESIRED_RATE) { ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate); } direction = 0; err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction); if (err < 0) { ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err)); } else { ast_log(LOG_DEBUG, "Period size is %d\n", err); } buffer_size = 4096 * 2; //period_size * 16; err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size); if (err < 0) { ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err)); } else { ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err); } #if 0 direction = 0; err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction); if (err < 0) { ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err)); } err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0); if (err < 0) { ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err)); } #endif err = snd_pcm_hw_params(handle, hwparams); if (err < 0) { ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err)); } snd_pcm_sw_params_alloca(&swparams); snd_pcm_sw_params_current(handle, swparams); #if 1 if (stream == SND_PCM_STREAM_PLAYBACK) { start_threshold = period_size; } else { start_threshold = 1; } err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold); if (err < 0) { ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err)); } #endif #if 1 if (stream == SND_PCM_STREAM_PLAYBACK) { stop_threshold = buffer_size; } else { stop_threshold = buffer_size; } err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold); if (err < 0) { ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err)); } #endif #if 0 err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES); if (err < 0) { ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err)); } #endif #if 0 err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold); if (err < 0) { ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err)); } #endif err = snd_pcm_sw_params(handle, swparams); if (err < 0) { ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err)); } err = snd_pcm_poll_descriptors_count(handle); if (err <= 0) { ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err)); } if (err != 1) { ast_log(LOG_DEBUG, "Can't handle more than one device\n"); } snd_pcm_poll_descriptors(handle, &pfd, err); ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd); if (stream == SND_PCM_STREAM_CAPTURE) readdev = pfd.fd; else writedev = pfd.fd; return handle; } static int soundcard_init(void) { alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE); alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK); if (!alsa.icard || !alsa.ocard) { ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n"); return -1; } return readdev; } static int alsa_digit(struct ast_channel *c, char digit) { ast_verbose( " << Console Received digit %c >> \n", digit); return 0; } static int alsa_text(struct ast_channel *c, char *text) { ast_verbose( " << Console Received text %s >> \n", text); return 0; } static int alsa_call(struct ast_channel *c, char *dest, int timeout) { int res = 3; ast_verbose( " << Call placed to '%s' on console >> \n", dest); if (autoanswer) { ast_verbose( " << Auto-answered >> \n" ); needanswer = 1; } else { ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); needringing = 1; write(sndcmd[1], &res, sizeof(res)); } return 0; } static void answer_sound(void) { int res; nosound = 1; res = 4; write(sndcmd[1], &res, sizeof(res)); } static int alsa_answer(struct ast_channel *c) { ast_verbose( " << Console call has been answered >> \n"); answer_sound(); ast_setstate(c, AST_STATE_UP); cursound = -1; return 0; } static int alsa_hangup(struct ast_channel *c) { int res; cursound = -1; c->pvt->pvt = NULL; alsa.owner = NULL; ast_verbose( " << Hangup on console >> \n"); ast_mutex_lock(&usecnt_lock); usecnt--; ast_mutex_unlock(&usecnt_lock); needhangup = 0; needanswer = 0; if (hookstate) { res = 2; write(sndcmd[1], &res, sizeof(res)); } return 0; } #if 0 static int soundcard_writeframe(short *data) { /* Write an exactly FRAME_SIZE sized of frame */ static int bufcnt = 0; static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5]; struct audio_buf_info info; int res; int fd = sounddev; static int warned=0; if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) { if (!warned) ast_log(LOG_WARNING, "Error reading output space\n"); bufcnt = buffersize; warned++; } if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) { /* We've run out of stuff, buffer again */ bufcnt = 0; } if (bufcnt == buffersize) { /* Write sample immediately */ res = write(fd, ((void *)data), FRAME_SIZE * 2); } else { /* Copy the data into our buffer */ res = FRAME_SIZE * 2; memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2); bufcnt++; if (bufcnt == buffersize) { res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize); } } return res; } #endif static int alsa_write(struct ast_channel *chan, struct ast_frame *f) { int res; static char sizbuf[8000]; static int sizpos = 0; int len = sizpos; int pos; //size_t frames = 0; snd_pcm_state_t state; /* Immediately return if no sound is enabled */ if (nosound) return 0; /* Stop any currently playing sound */ if (cursound != -1) { snd_pcm_drop(alsa.ocard); snd_pcm_prepare(alsa.ocard); cursound = -1; } /* We have to digest the frame in 160-byte portions */ if (f->datalen > sizeof(sizbuf) - sizpos) { ast_log(LOG_WARNING, "Frame too large\n"); return -1; } memcpy(sizbuf + sizpos, f->data, f->datalen); len += f->datalen; pos = 0; #ifdef ALSA_MONITOR alsa_monitor_write(sizbuf, len); #endif state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) { snd_pcm_prepare(alsa.ocard); } res = snd_pcm_writei(alsa.ocard, sizbuf, len/2); if (res == -EPIPE) { #if DEBUG ast_log(LOG_DEBUG, "XRUN write\n"); #endif snd_pcm_prepare(alsa.ocard); res = snd_pcm_writei(alsa.ocard, sizbuf, len/2); if (res != len/2) { ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res)); return -1; } else if (res < 0) { ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res)); return -1; } } else { if (res == -ESTRPIPE) { ast_log(LOG_ERROR, "You've got some big problems\n"); } } return 0; } static struct ast_frame *alsa_read(struct ast_channel *chan) { static struct ast_frame f; static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET/2]; short *buf; static int readpos = 0; static int left = FRAME_SIZE; int res; int b; int nonull=0; snd_pcm_state_t state; int r = 0; int off = 0; /* Acknowledge any pending cmd */ res = read(cmd[0], &b, sizeof(b)); if (res > 0) nonull = 1; f.frametype = AST_FRAME_NULL; f.subclass = 0; f.samples = 0; f.datalen = 0; f.data = NULL; f.offset = 0; f.src = type; f.mallocd = 0; if (needringing) { f.frametype = AST_FRAME_CONTROL; f.subclass = AST_CONTROL_RINGING; needringing = 0; return &f; } if (needhangup) { needhangup = 0; return NULL; } if (strlen(text2send)) { f.frametype = AST_FRAME_TEXT; f.subclass = 0; f.data = text2send; f.datalen = strlen(text2send); strcpy(text2send,""); return &f; } if (strlen(digits)) { f.frametype = AST_FRAME_DTMF; f.subclass = digits[0]; for (res=0;res= 0) { off -= r; } /* Update positions */ readpos += r; left -= r; if (readpos >= FRAME_SIZE) { /* A real frame */ readpos = 0; left = FRAME_SIZE; if (chan->_state != AST_STATE_UP) { /* Don't transmit unless it's up */ return &f; } f.frametype = AST_FRAME_VOICE; f.subclass = AST_FORMAT_SLINEAR; f.samples = FRAME_SIZE; f.datalen = FRAME_SIZE * 2; f.data = buf; f.offset = AST_FRIENDLY_OFFSET; f.src = type; f.mallocd = 0; #ifdef ALSA_MONITOR alsa_monitor_read((char *)buf, FRAME_SIZE * 2); #endif #if 0 { static int fd = -1; if (fd < 0) fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT); write(fd, f.data, f.datalen); } #endif } return &f; } static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) { struct chan_alsa_pvt *p = newchan->pvt->pvt; p->owner = newchan; return 0; } static int alsa_indicate(struct ast_channel *chan, int cond) { int res; switch(cond) { case AST_CONTROL_BUSY: res = 1; break; case AST_CONTROL_CONGESTION: res = 2; break; case AST_CONTROL_RINGING: res = 0; break; default: ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); return -1; } if (res > -1) { write(sndcmd[1], &res, sizeof(res)); } return 0; } static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state) { struct ast_channel *tmp; tmp = ast_channel_alloc(1); if (tmp) { snprintf(tmp->name, sizeof(tmp->name), "ALSA/%s", indevname); tmp->type = type; tmp->fds[0] = readdev; tmp->fds[1] = cmd[0]; tmp->nativeformats = AST_FORMAT_SLINEAR; tmp->pvt->pvt = p; tmp->pvt->send_digit = alsa_digit; tmp->pvt->send_text = alsa_text; tmp->pvt->hangup = alsa_hangup; tmp->pvt->answer = alsa_answer; tmp->pvt->read = alsa_read; tmp->pvt->call = alsa_call; tmp->pvt->write = alsa_write; tmp->pvt->indicate = alsa_indicate; tmp->pvt->fixup = alsa_fixup; if (strlen(p->context)) strncpy(tmp->context, p->context, sizeof(tmp->context)-1); if (strlen(p->exten)) strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1); if (strlen(language)) strncpy(tmp->language, language, sizeof(tmp->language)-1); p->owner = tmp; ast_setstate(tmp, 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(tmp)) { ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); ast_hangup(tmp); tmp = NULL; } } } return tmp; } static struct ast_channel *alsa_request(char *type, int format, void *data) { int oldformat = format; struct ast_channel *tmp; format &= AST_FORMAT_SLINEAR; if (!format) { ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat); return NULL; } if (alsa.owner) { ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n"); return NULL; } tmp= alsa_new(&alsa, AST_STATE_DOWN); if (!tmp) { ast_log(LOG_WARNING, "Unable to create new ALSA channel\n"); } return tmp; } static int console_autoanswer(int fd, int argc, char *argv[]) { if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; if (argc == 1) { ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); return RESULT_SUCCESS; } else { if (!strcasecmp(argv[1], "on")) autoanswer = -1; else if (!strcasecmp(argv[1], "off")) 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 switch(state) { case 0: if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2))) return strdup("on"); case 1: if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 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 'alsa.conf'.\n"; static int console_answer(int fd, int argc, char *argv[]) { if (argc != 1) return RESULT_SHOWUSAGE; if (!alsa.owner) { ast_cli(fd, "No one is calling us\n"); return RESULT_FAILURE; } hookstate = 1; cursound = -1; needanswer++; answer_sound(); 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[]) { int tmparg = 2; if (argc < 2) return RESULT_SHOWUSAGE; if (!alsa.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"); strcpy(text2send, ""); while(tmparg <= argc) { strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send)); strncat(text2send, " ", sizeof(text2send) - strlen(text2send)); } needanswer++; return RESULT_SUCCESS; } static char answer_usage[] = "Usage: answer\n" " Answers an incoming call on the console (ALSA) channel.\n"; static int console_hangup(int fd, int argc, char *argv[]) { if (argc != 1) return RESULT_SHOWUSAGE; cursound = -1; if (!alsa.owner && !hookstate) { ast_cli(fd, "No call to hangup up\n"); return RESULT_FAILURE; } hookstate = 0; if (alsa.owner) { ast_queue_hangup(alsa.owner); } return RESULT_SUCCESS; } static char hangup_usage[] = "Usage: hangup\n" " Hangs up any call currently placed on the console.\n"; static int console_dial(int fd, int argc, char *argv[]) { char tmp[256], *tmp2; char *mye, *myc; int b = 0; if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; if (alsa.owner) { if (argc == 2) { strncat(digits, argv[1], sizeof(digits) - strlen(digits)); /* Wake up the polling thread */ write(cmd[1], &b, sizeof(b)); } else { ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n"); return RESULT_FAILURE; } return RESULT_SUCCESS; } mye = exten; myc = context; if (argc == 2) { char *stringp=NULL; strncpy(tmp, argv[1], sizeof(tmp)-1); stringp=tmp; strsep(&stringp, "@"); tmp2 = strsep(&stringp, "@"); if (strlen(tmp)) mye = tmp; if (tmp2 && strlen(tmp2)) myc = tmp2; } if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { strncpy(alsa.exten, mye, sizeof(alsa.exten)-1); strncpy(alsa.context, myc, sizeof(alsa.context)-1); hookstate = 1; alsa_new(&alsa, 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 ("; 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 }, { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_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 } }; int load_module() { int res; int x; int flags; struct ast_config *cfg; struct ast_variable *v; res = pipe(cmd); res = pipe(sndcmd); if (res) { ast_log(LOG_ERROR, "Unable to create pipe\n"); return -1; } flags = fcntl(cmd[0], F_GETFL); fcntl(cmd[0], F_SETFL, flags | O_NONBLOCK); flags = fcntl(cmd[1], F_GETFL); fcntl(cmd[1], F_SETFL, flags | O_NONBLOCK); res = soundcard_init(); if (res < 0) { close(cmd[1]); close(cmd[0]); if (option_verbose > 1) { ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n"); ast_verbose(VERBOSE_PREFIX_2 "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n"); } return 0; } #if 0 if (!full_duplex) ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n"); #endif res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, alsa_request); if (res < 0) { ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type); return -1; } for (x=0;xname, "autoanswer")) autoanswer = ast_true(v->value); else if (!strcasecmp(v->name, "silencesuppression")) silencesuppression = ast_true(v->value); else if (!strcasecmp(v->name, "silencethreshold")) silencethreshold = atoi(v->value); else if (!strcasecmp(v->name, "context")) strncpy(context, v->value, sizeof(context)-1); else if (!strcasecmp(v->name, "language")) strncpy(language, v->value, sizeof(language)-1); else if (!strcasecmp(v->name, "extension")) strncpy(exten, v->value, sizeof(exten)-1); else if (!strcasecmp(v->name, "input_device")) strncpy(indevname, v->value, sizeof(indevname)-1); else if (!strcasecmp(v->name, "output_device")) strncpy(outdevname, v->value, sizeof(outdevname)-1); v=v->next; } ast_destroy(cfg); } pthread_create(&sthread, NULL, sound_thread, NULL); #ifdef ALSA_MONITOR if (alsa_monitor_start()) { ast_log(LOG_ERROR, "Problem starting Monitoring\n"); } #endif return 0; } int unload_module() { int x; for (x=0;x 0) { close(cmd[0]); close(cmd[1]); } if (sndcmd[0] > 0) { close(sndcmd[0]); close(sndcmd[1]); } if (alsa.owner) ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD); if (alsa.owner) return -1; return 0; } char *description() { return desc; } int usecount() { int res; ast_mutex_lock(&usecnt_lock); res = usecnt; ast_mutex_unlock(&usecnt_lock); return res; } char *key() { return ASTERISK_GPL_KEY; }