Version 0.1.2 from FTP
authorMark Spencer <markster@digium.com>
Thu, 6 Jan 2000 12:33:15 +0000 (12:33 +0000)
committerMark Spencer <markster@digium.com>
Thu, 6 Jan 2000 12:33:15 +0000 (12:33 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_oss.c [new file with mode: 0755]
configs/oss.conf.sample [new file with mode: 0755]

diff --git a/channels/chan_oss.c b/channels/chan_oss.c
new file mode 100755 (executable)
index 0000000..caf2403
--- /dev/null
@@ -0,0 +1,791 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as a channel, and the console to command it :).
+ *
+ * The full-duplex "simulation" is pretty weak.  This is generally a 
+ * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
+ * writing a driver.
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/options.h>
+#include <asterisk/pbx.h>
+#include <asterisk/config.h>
+#include <asterisk/cli.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <sys/time.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <linux/soundcard.h>
+
+/* Which device to use */
+#define DEV_DSP "/dev/dsp"
+
+/* Lets use 160 sample frames, just like GSM.  */
+#define FRAME_SIZE 160
+
+/* 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 * 5) << 16) | (0x0006);
+
+/* Don't switch between read/write modes faster than every 300 ms */
+#define MIN_SWITCH_TIME 600
+
+static struct timeval lasttime;
+
+static int usecnt;
+static int needanswer = 0;
+static int needhangup = 0;
+static int silencesuppression = 0;
+static int silencethreshold = 1000;
+
+static char digits[80] = "";
+
+static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static char *type = "Console";
+static char *desc = "OSS Console Channel Driver";
+static char *tdesc = "OSS Console Channel Driver";
+static char *config = "oss.conf";
+
+static char context[AST_MAX_EXTENSION] = "default";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+/* Some pipes to prevent overflow */
+static int funnel[2];
+static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_t silly;
+
+static struct chan_oss_pvt {
+       /* We only have one OSS 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];
+} oss;
+
+static int time_has_passed()
+{
+       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;
+}
+
+/* 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. */
+
+
+#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 descriptor for sound device */
+static int sounddev = -1;
+
+static int autoanswer = 1;
+static int calc_loudness(short *frame)
+{
+       int sum = 0;
+       int x;
+       for (x=0;x<FRAME_SIZE;x++) {
+               if (frame[x] < 0)
+                       sum -= frame[x];
+               else
+                       sum += frame[x];
+       }
+       sum = sum/FRAME_SIZE;
+       return sum;
+}
+
+static int silence_suppress(short *buf)
+{
+#define SILBUF 3
+       int loudness;
+       static int silentframes = 0;
+       static char silbuf[FRAME_SIZE * 2 * SILBUF];
+       static int silbufcnt=0;
+       if (!silencesuppression)
+               return 0;
+       loudness = calc_loudness((short *)(buf));
+       if (option_debug)
+               ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
+       if (loudness < silencethreshold) {
+               silentframes++;
+               silbufcnt++;
+               /* Keep track of the last few bits of silence so we can play
+                  them as lead-in when the time is right */
+               if (silbufcnt >= 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(funnel[1], silbuf, silbufcnt * FRAME_SIZE);
+                       silbufcnt = 0;
+               }
+       }
+       return 0;
+}
+
+static void *silly_thread(void *ignore)
+{
+       char buf[FRAME_SIZE * 2];
+       int pos=0;
+       int res=0;
+       /* Read from the sound device, and write to the pipe. */
+       for (;;) {
+               /* Give the writer a better shot at the lock */
+#if 0
+               usleep(1000);
+#endif         
+               pthread_testcancel();
+               pthread_mutex_lock(&sound_lock);
+               res = read(sounddev, buf + pos, FRAME_SIZE * 2 - pos);
+               pthread_mutex_unlock(&sound_lock);
+               if (res > 0) {
+                       pos += res;
+                       if (pos == FRAME_SIZE * 2) {
+                               if (needhangup || needanswer || strlen(digits) || 
+                                   !silence_suppress((short *)buf)) {
+                                       res = write(funnel[1], buf, sizeof(buf));
+                               }
+                               pos = 0;
+                       }
+               } else {
+                       close(funnel[1]);
+                       break;
+               }
+               pthread_testcancel();
+       }
+       return NULL;
+}
+
+static int setformat(void)
+{
+       int fmt, desired, res, fd = sounddev;
+       static int warnedalready = 0;
+       static int warnedalready2 = 0;
+       pthread_mutex_lock(&sound_lock);
+       fmt = AFMT_S16_LE;
+       res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+       if (res < 0) {
+               ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+               pthread_mutex_unlock(&sound_lock);
+               return -1;
+       }
+       res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+       if (res >= 0) {
+               if (option_verbose > 1) 
+                       ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+               full_duplex = -1;
+       }
+       fmt = 0;
+       res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+       if (res < 0) {
+               ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+               pthread_mutex_unlock(&sound_lock);
+               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");
+               pthread_mutex_unlock(&sound_lock);
+               return -1;
+       }
+       if (fmt != desired) {
+               if (!warnedalready++)
+                       ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+       }
+#if 1
+       fmt = BUFFER_FMT;
+       res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+       if (res < 0) {
+               if (!warnedalready2++)
+                       ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+       }
+#endif
+       pthread_mutex_unlock(&sound_lock);
+       return 0;
+}
+
+static int soundcard_setoutput(int force)
+{
+       /* Make sure the soundcard is in output mode.  */
+       int fd = sounddev;
+       if (full_duplex || (!readmode && !force))
+               return 0;
+       pthread_mutex_lock(&sound_lock);
+       readmode = 0;
+       if (force || time_has_passed()) {
+               ioctl(sounddev, SNDCTL_DSP_RESET);
+               /* Keep the same fd reserved by closing the sound device and copying stdin at the same
+                  time. */
+               /* dup2(0, sound); */ 
+               close(sounddev);
+               fd = open(DEV_DSP, O_WRONLY);
+               if (fd < 0) {
+                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               /* dup2 will close the original and make fd be sound */
+               if (dup2(fd, sounddev) < 0) {
+                       ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               if (setformat()) {
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               pthread_mutex_unlock(&sound_lock);
+               return 0;
+       }
+       pthread_mutex_unlock(&sound_lock);
+       return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+       int fd = sounddev;
+       if (full_duplex || (readmode && !force))
+               return 0;
+       pthread_mutex_lock(&sound_lock);
+       readmode = -1;
+       if (force || time_has_passed()) {
+               ioctl(sounddev, SNDCTL_DSP_RESET);
+               close(sounddev);
+               /* dup2(0, sound); */
+               fd = open(DEV_DSP, O_RDONLY);
+               if (fd < 0) {
+                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               /* dup2 will close the original and make fd be sound */
+               if (dup2(fd, sounddev) < 0) {
+                       ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               if (setformat()) {
+                       pthread_mutex_unlock(&sound_lock);
+                       return -1;
+               }
+               pthread_mutex_unlock(&sound_lock);
+               return 0;
+       }
+       pthread_mutex_unlock(&sound_lock);
+       return 1;
+}
+
+static int soundcard_init()
+{
+       /* Assume it's full duplex for starters */
+       int fd = open(DEV_DSP,  O_RDWR);
+       if (fd < 0) {
+               ast_log(LOG_ERROR, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+               return fd;
+       }
+       gettimeofday(&lasttime, NULL);
+       sounddev = fd;
+       setformat();
+       if (!full_duplex) 
+               soundcard_setinput(1);
+       return sounddev;
+}
+
+static int oss_digit(struct ast_channel *c, char digit)
+{
+       ast_verbose( " << Console Received digit %c >> \n", digit);
+       return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+       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");
+       }
+       return 0;
+}
+
+static int oss_answer(struct ast_channel *c)
+{
+       ast_verbose( " << Console call has been answered >> \n");
+       c->state = AST_STATE_UP;
+       return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+       c->pvt->pvt = NULL;
+       oss.owner = NULL;
+       ast_verbose( " << Hangup on console >> \n");
+       pthread_mutex_lock(&usecnt_lock);
+       usecnt--;
+       pthread_mutex_unlock(&usecnt_lock);
+       needhangup = 0;
+       needanswer = 0;
+       return 0;
+}
+
+static int soundcard_writeframe(short *data)
+{      
+       /* Write an exactly FRAME_SIZE sized of frame */
+       static int bufcnt = 0;
+       static char buffer[FRAME_SIZE * 2 * MAX_BUFFER_SIZE * 5];
+       struct audio_buf_info info;
+       int res;
+       int fd = sounddev;
+       static int warned=0;
+       pthread_mutex_lock(&sound_lock);
+       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 * 2), data, FRAME_SIZE * 2);
+               bufcnt++;
+               if (bufcnt == buffersize) {
+                       res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
+               }
+       }
+       pthread_mutex_unlock(&sound_lock);
+       return res;
+}
+
+
+static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+{
+       int res;
+       static char sizbuf[8000];
+       static int sizpos = 0;
+       int len = sizpos;
+       int pos;
+       if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
+               /* If we're half duplex, we have to switch to read mode
+                  to honor immediate needs if necessary */
+               res = soundcard_setinput(1);
+               if (res < 0) {
+                       ast_log(LOG_WARNING, "Unable to set device to input mode\n");
+                       return -1;
+               }
+               return 0;
+       }
+       res = soundcard_setoutput(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 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;
+       while(len - pos > FRAME_SIZE * 2) {
+               soundcard_writeframe((short *)(sizbuf + pos));
+               pos += FRAME_SIZE * 2;
+       }
+       if (len - pos) 
+               memmove(sizbuf, sizbuf + pos, len - pos);
+       sizpos = len - pos;
+       return 0;
+}
+
+static struct ast_frame *oss_read(struct ast_channel *chan)
+{
+       static struct ast_frame f;
+       static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
+       static int readpos = 0;
+       int res;
+       
+#if 0
+       ast_log(LOG_DEBUG, "oss_read()\n");
+#endif
+       
+       f.frametype = AST_FRAME_NULL;
+       f.subclass = 0;
+       f.timelen = 0;
+       f.datalen = 0;
+       f.data = NULL;
+       f.offset = 0;
+       f.src = type;
+       f.mallocd = 0;
+       
+       if (needhangup) {
+               return NULL;
+       }
+       if (strlen(digits)) {
+               f.frametype = AST_FRAME_DTMF;
+               f.subclass = digits[0];
+               for (res=0;res<strlen(digits);res++)
+                       digits[res] = digits[res + 1];
+               return &f;
+       }
+       
+       if (needanswer) {
+               needanswer = 0;
+               f.frametype = AST_FRAME_CONTROL;
+               f.subclass = AST_CONTROL_ANSWER;
+               chan->state = AST_STATE_UP;
+               return &f;
+       }
+       
+       res = soundcard_setinput(0);
+       if (res < 0) {
+               ast_log(LOG_WARNING, "Unable to set input mode\n");
+               return NULL;
+       }
+       if (res > 0) {
+               /* Theoretically shouldn't happen, but anyway, return a NULL frame */
+               return &f;
+       }
+       res = read(funnel[0], buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+       if (res < 0) {
+               ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
+               return NULL;
+       }
+       readpos += res;
+       
+       if (readpos == FRAME_SIZE * 2) {
+               /* A real frame */
+               readpos = 0;
+               f.frametype = AST_FRAME_VOICE;
+               f.subclass = AST_FORMAT_SLINEAR;
+               f.timelen = FRAME_SIZE / 8;
+               f.datalen = FRAME_SIZE * 2;
+               f.data = buf + AST_FRIENDLY_OFFSET;
+               f.offset = AST_FRIENDLY_OFFSET;
+               f.src = type;
+               f.mallocd = 0;
+       }
+       return &f;
+}
+
+static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+{
+       struct ast_channel *tmp;
+       tmp = ast_channel_alloc();
+       if (tmp) {
+               snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+               tmp->type = type;
+               tmp->fd = funnel[0];
+               tmp->format = AST_FORMAT_SLINEAR;
+               tmp->pvt->pvt = p;
+               tmp->pvt->send_digit = oss_digit;
+               tmp->pvt->hangup = oss_hangup;
+               tmp->pvt->answer = oss_answer;
+               tmp->pvt->read = oss_read;
+               tmp->pvt->write = oss_write;
+               if (strlen(p->context))
+                       strncpy(tmp->context, p->context, sizeof(tmp->context));
+               if (strlen(p->exten))
+                       strncpy(tmp->exten, p->exten, sizeof(tmp->exten));
+               p->owner = tmp;
+               tmp->state = state;
+               pthread_mutex_lock(&usecnt_lock);
+               usecnt++;
+               pthread_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 *oss_request(char *type, int format, void *data)
+{
+       int oldformat = format;
+       format &= AST_FORMAT_SLINEAR;
+       if (!format) {
+               ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
+               return NULL;
+       }
+       if (oss.owner) {
+               ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+               return NULL;
+       }
+       return oss_new(&oss, AST_STATE_DOWN);
+}
+
+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 'oss.conf'.\n";
+
+static int console_answer(int fd, int argc, char *argv[])
+{
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       if (!oss.owner) {
+               ast_cli(fd, "No one is calling us\n");
+               return RESULT_FAILURE;
+       }
+       needanswer++;
+       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[])
+{
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       if (!oss.owner) {
+               ast_cli(fd, "No call to hangup up\n");
+               return RESULT_FAILURE;
+       }
+       needhangup++;
+       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;
+       if ((argc != 1) && (argc != 2))
+               return RESULT_SHOWUSAGE;
+       if (oss.owner) {
+               if (argc == 2)
+                       strncat(digits, argv[1], sizeof(digits) - strlen(digits));
+               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) {
+               strncpy(tmp, argv[1], sizeof(tmp));
+               strtok(tmp, "@");
+               tmp2 = strtok(NULL, "@");
+               if (strlen(tmp))
+                       mye = tmp;
+               if (tmp2 && strlen(tmp2))
+                       myc = tmp2;
+       }
+       if (ast_exists_extension(NULL, myc, mye, 1)) {
+               strncpy(oss.exten, mye, sizeof(oss.exten));
+               strncpy(oss.context, myc, sizeof(oss.context));
+               oss_new(&oss, AST_STATE_UP);
+       } 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 },
+       { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
+};
+
+int load_module()
+{
+       int res;
+       int x;
+       int flags;
+       struct ast_config *cfg = ast_load(config);
+       struct ast_variable *v;
+       res = pipe(funnel);
+       if (res) {
+               ast_log(LOG_ERROR, "Unable to create pipe\n");
+               return -1;
+       }
+       /* We make the funnel so that writes to the funnel don't block...
+          Our "silly" thread can read to its heart content, preventing
+          recording overruns */
+       flags = fcntl(funnel[1], F_GETFL);
+#if 0
+       fcntl(funnel[0], F_SETFL, flags | O_NONBLOCK);
+#endif
+       fcntl(funnel[1], F_SETFL, flags | O_NONBLOCK);
+       res = soundcard_init();
+       if (res < 0) {
+               close(funnel[1]);
+               close(funnel[0]);
+               return -1;
+       }
+       if (!full_duplex)
+               ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+       pthread_create(&silly, NULL, silly_thread, NULL);
+       res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
+       if (res < 0) {
+               ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
+               return -1;
+       }
+       for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+               ast_cli_register(myclis + x);
+       if (cfg) {
+               v = ast_variable_browse(cfg, "general");
+               while(v) {
+                       if (!strcasecmp(v->name, "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));
+                       else if (!strcasecmp(v->name, "extension"))
+                               strncpy(exten, v->value, sizeof(exten));
+                       v=v->next;
+               }
+               ast_destroy(cfg);
+       }
+       return 0;
+}
+
+
+
+int unload_module()
+{
+       int x;
+       for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+               ast_cli_unregister(myclis + x);
+       close(sounddev);
+       if (funnel[0] > 0) {
+               close(funnel[0]);
+               close(funnel[1]);
+       }
+       if (silly) {
+               pthread_cancel(silly);
+               pthread_join(silly, NULL);
+       }
+       if (oss.owner)
+               ast_softhangup(oss.owner);
+       if (oss.owner)
+               return -1;
+       return 0;
+}
+
+char *description()
+{
+       return desc;
+}
+
+int usecount()
+{
+       int res;
+       pthread_mutex_lock(&usecnt_lock);
+       res = usecnt;
+       pthread_mutex_unlock(&usecnt_lock);
+       return res;
+}
diff --git a/configs/oss.conf.sample b/configs/oss.conf.sample
new file mode 100755 (executable)
index 0000000..138a737
--- /dev/null
@@ -0,0 +1,23 @@
+;
+; Open Sound System Console Driver Configuration File
+;
+[general]
+;
+; Automatically answer incoming calls on the console?  Choose yes if
+; for example you want to use this as an intercom.
+;
+autoanswer=yes
+;
+; Default context (is overridden with @context syntax)
+;
+;context=local
+;
+; Default extension to call
+;
+extension=s
+;
+; Silence supression can be enabled when sound is over a certain threshold.
+; The value for the threshold should probably be between 500 and 2000 or so,
+; but your mileage may vary.  Use the echo test to evaluate the best setting.
+;silencesuppression = yes
+;silencethreshold = 1000