Move to rizzo's new chan_oss, but leave the old one just in case... (bug #4379 with...
authorMark Spencer <markster@digium.com>
Wed, 3 Aug 2005 04:11:52 +0000 (04:11 +0000)
committerMark Spencer <markster@digium.com>
Wed, 3 Aug 2005 04:11:52 +0000 (04:11 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6263 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_oss.c
channels/chan_oss_old.c [new file with mode: 0755]

index 8b61abf..b8f0fb6 100755 (executable)
@@ -1,28 +1,27 @@
 /*
  * 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 - 2005, Digium, Inc.
  *
  * Mark Spencer <markster@digium.com>
  *
  * This program is free software, distributed under the terms of
  * the GNU General Public License
+ *
+ * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25
+ * note-this code best seen with ts=8 (8-spaces tabs) in the editor
  */
 
+#include <stdio.h>
+#include <ctype.h>     /* for isalnum */
+#include <string.h>
 #include <unistd.h>
-#include <fcntl.h>
-#include <errno.h>
 #include <sys/ioctl.h>
+#include <fcntl.h>
 #include <sys/time.h>
-#include <string.h>
 #include <stdlib.h>
-#include <stdio.h>
+#include <errno.h>
+
 
 #ifdef __linux
 #include <linux/soundcard.h>
@@ -44,16 +43,132 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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"
 
+/* ringtones we use */
 #include "busy.h"
 #include "ringtone.h"
 #include "ring10.h"
 #include "answer.h"
 
+/*
+ * 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
+
+[general]
+; general config options, default values are shown
+; all but debug can go also in the device-specific sections.
+; debug=0x0            ; misc debug flags, default is 0
+
+[card1]
+; autoanswer = no      ; no autoanswer on call
+; autohangup = yes     ; hangup when other party closes
+; extension=s          ; default extension to call
+; context=default      ; default context
+; language=""          ; default language
+; overridecontext=no   ; the whole dial string is considered an extension.
+                       ; if yes, the last @ will start the context
+
+; device=/dev/dsp      ; device to open
+; mixer="-f /dev/mixer0 pcm 80 ; mixer command to run on start
+; queuesize=10         ; frames in device driver
+; frags=8              ; argument to SETFRAGMENT
+
+.. 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"
@@ -61,42 +176,29 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #define DEV_DSP "/dev/dsp"
 #endif
 
-/* 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 * 10) << 16) | (0x0006);
-
-/* Don't switch between read/write modes faster than every 300 ms */
-#define MIN_SWITCH_TIME 600
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+#ifndef MAX
+#define MAX(a,b) ((a) > (b) ? (a) : (b))
+#endif
 
-static struct timeval lasttime;
 
 static int usecnt;
-static int silencesuppression = 0;
-static int silencethreshold = 1000;
-static int playbackonly = 0;
-
-
 AST_MUTEX_DEFINE_STATIC(usecnt_lock);
 
-static const char type[] = "Console";
-static const char desc[] = "OSS Console Channel Driver";
-static const char tdesc[] = "OSS Console Channel Driver";
-static const char config[] = "oss.conf";
-
-static char context[AST_MAX_CONTEXT] = "default";
-static char language[MAX_LANGUAGE] = "";
-static char exten[AST_MAX_EXTENSION] = "s";
+static char *config = "oss.conf";      /* default config file */
 
-static int hookstate=0;
-
-static short silence[FRAME_SIZE] = {0, };
+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;
@@ -105,25 +207,99 @@ struct sound {
 };
 
 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 },
+       { 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 */
 };
 
-/* Sound command pipe */
-static int sndcmd[2];
 
-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. */
+/*
+ * 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;
+       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;
+       char device[64];        /* device to open */
+
+       pthread_t sthread;
+
        struct ast_channel *owner;
-       char exten[AST_MAX_EXTENSION];
-       char context[AST_MAX_CONTEXT];
-} oss;
+       char ext[AST_MAX_EXTENSION];
+       char ctx[AST_MAX_CONTEXT];
+       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,
+       .ext = "s",
+       .ctx = "default",
+       .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */
+       .lastopen = { 0, 0 },
+};
 
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause);
+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(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);
@@ -135,704 +311,649 @@ 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 = type,
-       .description = tdesc,
-       .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,
+       .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,
 };
 
-static int time_has_passed(void)
+/*
+ * returns a pointer to the descriptor with the given name
+ */
+static struct chan_oss_pvt *find_desc(char *dev)
 {
-       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. */
+       struct chan_oss_pvt *o;
 
-static pthread_t sthread;
+       for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next)
+               ;
+       if (o == NULL)
+               ast_log(LOG_WARNING, "could not find <%s>\n", dev);
+       return o;
+}
 
-#define MAX_BUFFER_SIZE 100
-static int buffersize = 3;
+/*
+ * split a string in extension-context, returns pointers to malloc'ed
+ * strings.
+ * If we 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 = strdup(src);
+       if (*ext == NULL)
+               return NULL;
+       if (!o->overridecontext) {
+               /* parse from the right */
+               *ctx = strrchr(*ext, '@');
+               if (*ctx)
+                       *(*ctx)++ = '\0';
+       }
+       return *ext;
+}
 
-static int full_duplex = 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;
 
-/* Are we reading or writing (simulated full duplex) */
-static int readmode = 1;
+       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;
+}
 
-/* File descriptor for sound device */
-static int sounddev = -1;
+/* Write an exactly FRAME_SIZE sized frame */
+static int soundcard_writeframe(struct chan_oss_pvt *o, short *data)
+{      
+       int res;
 
-static int autoanswer = 1;
-#if 0
-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];
+       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;
        }
-       sum = sum/FRAME_SIZE;
-       return sum;
+       o->w_errors = 0;
+       return write(o->sounddev, ((void *)data), FRAME_SIZE * 2);
 }
-#endif
-
-static int cursound = -1;
-static int sampsent = 0;
-static int silencelen=0;
-static int offset=0;
-static int nosound=0;
 
-static int send_sound(void)
+/*
+ * 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 total = FRAME_SIZE;
-       short *frame = NULL;
-       int amt=0;
-       int res;
-       int myoff;
-       audio_buf_info abi;
-       if (cursound > -1) {
-               res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
-               if (res) {
-                       ast_log(LOG_WARNING, "Unable to read output space\n");
-                       return -1;
-               }
-               /* Calculate how many samples we can send, max */
-               if (total > (abi.fragments * abi.fragsize / 2)) 
-                       total = abi.fragments * abi.fragsize / 2;
-               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;
+       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;
                        }
                }
-               if (frame)
-                       res = write(sounddev, frame, res * 2);
-               if (res > 0)
-                       return 0;
-               return res;
        }
-       return 0;
+       l = soundcard_writeframe(o, myframe);
+       if (l > 0)
+               o->sampsent = l_sampsent;       /* update status */
 }
 
-static void *sound_thread(void *unused)
+static void *sound_thread(void *arg)
 {
-       fd_set rfds;
-       fd_set wfds;
-       int max;
-       int res;
        char ign[4096];
-       if (read(sounddev, ign, sizeof(sounddev)) < 0)
-               ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
-       for(;;) {
+       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);
-               max = sndcmd[0];
-               FD_SET(sndcmd[0], &rfds);
-               if (!oss.owner) {
-                       FD_SET(sounddev, &rfds);
-                       if (sounddev > max)
-                               max = sounddev;
-               }
-               if (cursound > -1) {
-                       FD_SET(sounddev, &wfds);
-                       if (sounddev > max)
-                               max = sounddev;
+               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);
+                       }
                }
-               res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+               /* 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(sndcmd[0], &rfds)) {
-                       read(sndcmd[0], &cursound, sizeof(cursound));
-                       silencelen = 0;
-                       offset = 0;
-                       sampsent = 0;
+               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(sounddev, &rfds)) {
-                       /* Ignore read */
-                       if (read(sounddev, ign, sizeof(ign)) < 0)
-                               ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+               if (o->sounddev > -1) {
+                       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);
                }
-               if (FD_ISSET(sounddev, &wfds))
-                       if (send_sound())
-                               ast_log(LOG_WARNING, "Failed to write sound\n");
        }
-       /* Never reached */
-       return NULL;
+       return NULL; /* Never reached */
 }
 
-#if 0
-static int silence_suppress(short *buf)
+/*
+ * 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)
 {
-#define SILBUF 3
-       int loudness;
-       static int silentframes = 0;
-       static char silbuf[FRAME_SIZE * 2 * SILBUF];
-       static int silbufcnt=0;
-       if (!silencesuppression)
+       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;
-       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(sounddev, silbuf, silbufcnt * FRAME_SIZE);
-                       silbufcnt = 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;
        }
-       return 0;
-}
-#endif
-
-static int setformat(void)
-{
-       int fmt, desired, res, fd = sounddev;
-       static int warnedalready = 0;
-       static int warnedalready2 = 0;
+       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;
        }
-       res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
-       
-       /* Check to see if duplex set (FreeBSD Bug)*/
-       res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
-       
-       if ((fmt & DSP_CAP_DUPLEX) && !res) {
-               if (option_verbose > 1) 
-                       ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
-               full_duplex = -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;
+       fmt = desired = 8000; /* 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 (!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
-       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;
-       readmode = 0;
-       if (force || time_has_passed()) {
-               ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-               /* 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 |O_NONBLOCK);
-               if (fd < 0) {
-                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-                       return -1;
+               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;
                }
-               /* 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));
-                       return -1;
-               }
-               if (setformat()) {
-                       return -1;
-               }
-               return 0;
        }
-       return 1;
-}
-
-static int soundcard_setinput(int force)
-{
-       int fd = sounddev;
-       if (full_duplex || (readmode && !force))
-               return 0;
-       readmode = -1;
-       if (force || time_has_passed()) {
-               ioctl(sounddev, SNDCTL_DSP_RESET, 0);
-               close(sounddev);
-               /* dup2(0, sound); */
-               fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
-               if (fd < 0) {
-                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
-                       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));
-                       return -1;
-               }
-               if (setformat()) {
-                       return -1;
+       /*
+        * 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;
+                       }
                }
-               return 0;
        }
-       return 1;
-}
-
-static int soundcard_init(void)
-{
-       /* Assume it's full duplex for starters */
-       int fd = open(DEV_DSP,  O_RDWR | O_NONBLOCK);
-       if (fd < 0) {
-               ast_log(LOG_WARNING, "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;
+       /* 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(struct ast_channel *c, char digit)
 {
+       /* no better use for received digits than print them */
        ast_verbose( " << Console Received digit %c >> \n", digit);
        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)
+{
+       write(o->sndcmd[1], &x, sizeof(x));
+}
+
+
+/*
+ * handler for incoming calls. Either autoanswer, or start ringing
+ */
 static int oss_call(struct ast_channel *c, char *dest, int timeout)
 {
-       int res = 3;
+       struct chan_oss_pvt *o = c->tech_pvt;
        struct ast_frame f = { 0, };
-       ast_verbose( " << Call placed to '%s' on console >> \n", dest);
-       if (autoanswer) {
+
+       ast_verbose(" << Call to '%s' on console from <%s><%s><%s> >>\n",
+               dest, c->cid.cid_dnid, c->cid.cid_num, c->cid.cid_name);
+       if (o->autoanswer) {
                ast_verbose( " << Auto-answered >> \n" );
                f.frametype = AST_FRAME_CONTROL;
                f.subclass = AST_CONTROL_ANSWER;
                ast_queue_frame(c, &f);
        } else {
-               nosound = 1;
-               ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+               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);
-               write(sndcmd[1], &res, sizeof(res));
+               ring(o, AST_CONTROL_RING);
        }
        return 0;
 }
 
-static void answer_sound(void)
-{
-       int res;
-       nosound = 1;
-       res = 4;
-       write(sndcmd[1], &res, sizeof(res));
-       
-}
-
+/*
+ * 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");
-       answer_sound();
+#if 0
+       /* play an answer tone (XXX do we really need it ?) */
+       ring(o, AST_CONTROL_ANSWER);
+#endif
        ast_setstate(c, AST_STATE_UP);
-       cursound = -1;
-       nosound=0;
+       o->cursound = -1;
+       o->nosound=0;
        return 0;
 }
 
 static int oss_hangup(struct ast_channel *c)
 {
-       int res = 0;
-       cursound = -1;
+       struct chan_oss_pvt *o = c->tech_pvt;
+
+       o->cursound = -1;
+       o->nosound = 0;
        c->tech_pvt = NULL;
-       oss.owner = NULL;
+       o->owner = NULL;
        ast_verbose( " << Hangup on console >> \n");
-       ast_mutex_lock(&usecnt_lock);
+       ast_mutex_lock(&usecnt_lock);   /* XXX not sure why */
        usecnt--;
        ast_mutex_unlock(&usecnt_lock);
-       if (hookstate) {
-               if (autoanswer) {
+       if (o->hookstate) {
+               if (o->autoanswer || o->autohangup) {
                        /* Assume auto-hangup too */
-                       hookstate = 0;
+                       o->hookstate = 0;
+                       setformat(o, O_CLOSE);
                } else {
                        /* Make congestion noise */
-                       res = 2;
-                       write(sndcmd[1], &res, sizeof(res));
+                       ring(o, AST_CONTROL_CONGESTION);
                }
        }
        return 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;
-}
-
-
-static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+/* used for data coming from the network */
+static int oss_write(struct ast_channel *c, struct ast_frame *f)
 {
-       int res;
-       static char sizbuf[8000];
-       static int sizpos = 0;
-       int len = sizpos;
-       int pos;
+       int src;
+       struct chan_oss_pvt *o = c->tech_pvt;
+
        /* Immediately return if no sound is enabled */
-       if (nosound)
+       if (o->nosound)
                return 0;
        /* Stop any currently playing sound */
-       cursound = -1;
-       if (!full_duplex && !playbackonly) {
-               /* If we're half duplex, we have to switch to read mode
-                  to honor immediate needs if necessary.  But if we are in play
-                  back only mode, then we don't switch because the console
-                  is only being used one way -- just to playback something. */
-               res = soundcard_setinput(1);
-               if (res < 0) {
-                       ast_log(LOG_WARNING, "Unable to set device to input mode\n");
-                       return -1;
+       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;
-       }
-       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 *oss_read(struct ast_channel *c)
 {
-       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.samples = 0;
-       f.datalen = 0;
-       f.data = NULL;
-       f.offset = 0;
-       f.src = type;
-       f.mallocd = 0;
-       f.delivery.tv_sec = 0;
-       f.delivery.tv_usec = 0;
-       
-       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(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
-       if (res < 0) {
-               ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
-#if 0
-               CRASH;
-#endif         
-               return NULL;
-       }
-       readpos += res;
-       
-       if (readpos >= FRAME_SIZE * 2) {
-               /* A real frame */
-               readpos = 0;
-               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 + AST_FRIENDLY_OFFSET;
-               f.offset = AST_FRIENDLY_OFFSET;
-               f.src = type;
-               f.mallocd = 0;
-               f.delivery.tv_sec = 0;
-               f.delivery.tv_usec = 0;
-#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;
+       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 = 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;
+       f->offset = AST_FRIENDLY_OFFSET;
+       return f;
 }
 
 static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
 {
-       struct chan_oss_pvt *p = newchan->tech_pvt;
-       p->owner = newchan;
+       struct chan_oss_pvt *o = newchan->tech_pvt;
+       o->owner = newchan;
        return 0;
 }
 
-static int oss_indicate(struct ast_channel *chan, int cond)
+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:
-               res = 1;
-               break;
        case AST_CONTROL_CONGESTION:
-               res = 2;
-               break;
        case AST_CONTROL_RINGING:
-               res = 0;
+               res = cond;
                break;
+
        case -1:
-               cursound = -1;
+               o->cursound = -1;
                return 0;
+
        default:
-               ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
+               ast_log(LOG_WARNING,
+                   "Don't know how to display condition %d on %s\n",
+                   cond, c->name);
                return -1;
        }
-       if (res > -1) {
-               write(sndcmd[1], &res, sizeof(res));
-       }
+       if (res > -1)
+               ring(o, res);
        return 0;       
 }
 
-static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+/*
+ * allocate a new channel.
+ */
+static struct ast_channel *oss_new(struct chan_oss_pvt *o,
+       char *ext, char *ctx, int state)
 {
-       struct ast_channel *tmp;
-       tmp = ast_channel_alloc(1);
-       if (tmp) {
-               tmp->tech = &oss_tech;
-               snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
-               tmp->type = type;
-               tmp->fds[0] = sounddev;
-               tmp->nativeformats = AST_FORMAT_SLINEAR;
-               tmp->readformat = AST_FORMAT_SLINEAR;
-               tmp->writeformat = AST_FORMAT_SLINEAR;
-               tmp->tech_pvt = p;
-               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;
-                       }
+       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; /* -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 (ctx && !ast_strlen_zero(ctx))
+               ast_copy_string(c->context, ctx, sizeof(c->context));
+       if (ext && !ast_strlen_zero(ext))
+               ast_copy_string(c->exten, ext, sizeof(c->exten));
+       if (o->language && !ast_strlen_zero(o->language))
+               ast_copy_string(c->language, o->language, sizeof(c->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 tmp;
+       return c;
 }
 
-static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause)
+static struct ast_channel *oss_request(const char *type,
+       int format, void *data, int *cause)
 {
-       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);
+       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 (oss.owner) {
-               ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+       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;
        }
-       tmp= oss_new(&oss, AST_STATE_DOWN);
-       if (!tmp) {
+       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 tmp;
+       return c;
 }
 
 static int console_autoanswer(int fd, int argc, char *argv[])
 {
-       if ((argc != 1) && (argc != 2))
-               return RESULT_SHOWUSAGE;
+       struct chan_oss_pvt *o = find_desc(oss_active);
+
        if (argc == 1) {
-               ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+               ast_cli(fd, "Auto answer is %s.\n", o->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;
        }
+       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 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 (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+               if (l && !strncasecmp(word, "on", MIN(l, 2)))
                        return strdup("on");
        case 1:
-               if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+               if (l && !strncasecmp(word, "off", MIN(l, 3)))
                        return strdup("off");
        default:
                return NULL;
@@ -846,19 +967,28 @@ static char autoanswer_usage[] =
 "       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(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 (!oss.owner) {
+       if (!o->owner) {
                ast_cli(fd, "No one is calling us\n");
                return RESULT_FAILURE;
        }
-       hookstate = 1;
-       cursound = -1;
-       ast_queue_frame(oss.owner, &f);
-       answer_sound();
+       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;
 }
 
@@ -866,30 +996,34 @@ static char sendtext_usage[] =
 "Usage: send text <message>\n"
 "       Sends a text message for display on the remote terminal.\n";
 
+/*
+ * concatenate all arguments into a single string
+ */
 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] = "";
+       char text2send[TEXT_SIZE] = "";
        struct ast_frame f = { 0, };
+
        if (argc < 2)
                return RESULT_SHOWUSAGE;
-       if (!oss.owner) {
-               ast_cli(fd, "No one is calling us\n");
+       if (!o->owner) {
+               ast_cli(fd, "Not in a call\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);
+       while (tmparg < argc) {
+               strncat(text2send, argv[tmparg++],
+                       sizeof(text2send) - strlen(text2send) - 1);
+               strncat(text2send, " ",
+                       sizeof(text2send) - strlen(text2send) - 1);
        }
-       if (strlen(text2send)) {
+       if (!ast_strlen_zero(text2send)) {
                f.frametype = AST_FRAME_TEXT;
                f.subclass = 0;
                f.data = text2send;
                f.datalen = strlen(text2send);
-               ast_queue_frame(oss.owner, &f);
+               ast_queue_frame(o->owner, &f);
        }
        return RESULT_SUCCESS;
 }
@@ -900,86 +1034,91 @@ static char answer_usage[] =
 
 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;
-       cursound = -1;
-       if (!oss.owner && !hookstate) {
+       o->cursound = -1;
+       o->nosound = 0;
+       if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */
                ast_cli(fd, "No call to hangup up\n");
                return RESULT_FAILURE;
        }
-       hookstate = 0;
-       if (oss.owner) {
-               ast_queue_hangup(oss.owner);
-       }
+       o->hookstate = 0;
+       if (o->owner)
+               ast_queue_hangup(o->owner);
+       setformat(o, O_CLOSE);
        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;
-       cursound = -1;
-       if (!oss.owner) {
+       o->cursound = -1;
+       if (!o->owner) { /* XXX maybe !o->hookstate too ? */
                ast_cli(fd, "No call to flash\n");
                return RESULT_FAILURE;
        }
-       hookstate = 0;
-       if (oss.owner) {
-               ast_queue_frame(oss.owner, &f);
-       }
+       o->hookstate = 0;
+       if (o->owner) /* XXX must be true, right ? */
+               ast_queue_frame(o->owner, &f);
        return RESULT_SUCCESS;
 }
 
-static char hangup_usage[] =
-"Usage: hangup\n"
-"       Hangs up any call currently placed on the console.\n";
-
 
 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[256], *tmp2;
-       char *mye, *myc;
-       int x;
-       struct ast_frame f = { AST_FRAME_DTMF, 0 };
-       if ((argc != 1) && (argc != 2))
+       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 (oss.owner) {
-               if (argc == 2) {
-                       for (x=0;x<strlen(argv[1]);x++) {
-                               f.subclass = argv[1][x];
-                               ast_queue_frame(oss.owner, &f);
-                       }
-               } else {
-                       ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
+       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;
        }
-       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 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)) {
-               strncpy(oss.exten, mye, sizeof(oss.exten)-1);
-               strncpy(oss.context, myc, sizeof(oss.context)-1);
-               hookstate = 1;
-               oss_new(&oss, AST_STATE_RINGING);
+               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;
 }
 
@@ -987,31 +1126,60 @@ static char dial_usage[] =
 "Usage: dial [extension[@context]]\n"
 "       Dials a given extensison (and context if specified)\n";
 
+static char mute_usage[] =
+"Usage: mute\nMutes the microphone\n";
+
+static char unmute_usage[] =
+"Usage: unmute\nUnmutes the microphone\n";
+
+static int console_mute(int fd, int argc, char *argv[])
+{
+       struct chan_oss_pvt *o = find_desc(oss_active);
+
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       o->mute = 1;
+       return RESULT_SUCCESS;
+}
+
+static int console_unmute(int fd, int argc, char *argv[])
+{
+       struct chan_oss_pvt *o = find_desc(oss_active);
+
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       o->mute = 0;
+       return RESULT_SUCCESS;
+}
+
 static int console_transfer(int fd, int argc, char *argv[])
 {
-       char tmp[256];
-       char *context;
+       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 (oss.owner && ast_bridged_channel(oss.owner)) {
-               strncpy(tmp, argv[1], sizeof(tmp) - 1);
-               context = strchr(tmp, '@');
-               if (context) {
-                       *context = '\0';
-                       context++;
-               } else
-                       context = oss.owner->context;
-               if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
-                       ast_cli(fd, "Whee, transferring %s to %s@%s.\n", 
-                                       ast_bridged_channel(oss.owner)->name, tmp, context);
-                       if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
-                               ast_cli(fd, "Failed to transfer :(\n");
-               } else {
-                       ast_cli(fd, "No such extension exists\n");
-               }
-       } else {
+       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;
 }
 
@@ -1020,93 +1188,211 @@ static char transfer_usage[] =
 "       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 },
+       { { "mute", NULL }, console_mute, "Disable mic input", mute_usage },
+       { { "unmute", NULL }, console_unmute, "Enable mic input", unmute_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 }
+       { { "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"}
 };
 
-int load_module()
+/*
+ * 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 res;
-       int x;
-       struct ast_config *cfg;
-       struct ast_variable *v;
-       res = pipe(sndcmd);
-       if (res) {
-               ast_log(LOG_ERROR, "Unable to create pipe\n");
-               return -1;
+       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;
+               }
        }
-       res = soundcard_init();
-       if (res < 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 OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
+       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;
+               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;
                }
-               return 0;
+               o->name = strdup(ctg);
        }
-       if (!full_duplex)
-               ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
-       res = ast_channel_register(&oss_tech);
-       if (res < 0) {
-               ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
-               return -1;
+
+       /* fill other fields from configuration */
+       for (v = ast_variable_browse(cfg, ctg);v; v=v->next) {
+               M_START(v->name, v->value);
+
+               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("extension", o->ext)
+               M_F("mixer", store_mixer(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;
+
+               asprintf(&cmd, "mixer %s", o->mixer_cmd);
+               ast_log(LOG_WARNING, "running [%s]\n", cmd);
+               system(cmd);
+               free(cmd);
        }
-       for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
-               ast_cli_register(myclis + x);
-       if ((cfg = ast_config_load(config))) {
-               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)-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, "playbackonly"))
-                               playbackonly = ast_true(v->value);
-                       v=v->next;
+       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(&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(void)
+{
+       int i;
+       struct ast_config *cfg;
+
+       /* load config file */
+       cfg = ast_config_load(config);
+       if (cfg != NULL) {
+               char *ctg = NULL;       /* first pass is 'general' */
+
+               do {
+                       store_config(cfg, ctg);
+               } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL);
                ast_config_destroy(cfg);
        }
-       ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+       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 -1;
+       }
+       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;
+       }
+       ast_cli_register_multiple(myclis, sizeof(myclis)/sizeof(struct ast_cli_entry));
        return 0;
 }
 
 
-
 int unload_module()
 {
-       int x;
+       struct chan_oss_pvt *o;
 
        ast_channel_unregister(&oss_tech);
-       for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
-               ast_cli_unregister(myclis + x);
-       close(sounddev);
-       if (sndcmd[0] > 0) {
-               close(sndcmd[0]);
-               close(sndcmd[1]);
+       ast_cli_unregister_multiple(myclis,
+               sizeof(myclis)/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 ? */
        }
-       if (oss.owner)
-               ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
-       if (oss.owner)
-               return -1;
        return 0;
 }
 
 char *description()
 {
-       return (char *) desc;
+       return (char *)oss_tech.description;
 }
 
 int usecount()
diff --git a/channels/chan_oss_old.c b/channels/chan_oss_old.c
new file mode 100755 (executable)
index 0000000..8b61abf
--- /dev/null
@@ -0,0 +1,1120 @@
+/*
+ * 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 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#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>
+
+#ifdef __linux
+#include <linux/soundcard.h>
+#elif defined(__FreeBSD__)
+#include <sys/soundcard.h>
+#else
+#include <soundcard.h>
+#endif
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#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"
+#include "asterisk/cli.h"
+#include "asterisk/utils.h"
+#include "asterisk/causes.h"
+#include "asterisk/endian.h"
+
+#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
+
+/* 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 * 10) << 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 silencesuppression = 0;
+static int silencethreshold = 1000;
+static int playbackonly = 0;
+
+
+AST_MUTEX_DEFINE_STATIC(usecnt_lock);
+
+static const char type[] = "Console";
+static const char desc[] = "OSS Console Channel Driver";
+static const char tdesc[] = "OSS Console Channel Driver";
+static const char config[] = "oss.conf";
+
+static char context[AST_MAX_CONTEXT] = "default";
+static char language[MAX_LANGUAGE] = "";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+static 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_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_CONTEXT];
+} oss;
+
+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 = type,
+       .description = tdesc,
+       .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,
+};
+
+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;
+}
+
+/* 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. */
+
+static 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 descriptor for sound device */
+static int sounddev = -1;
+
+static int autoanswer = 1;
+#if 0
+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;
+}
+#endif
+
+static int cursound = -1;
+static int sampsent = 0;
+static int silencelen=0;
+static int offset=0;
+static int nosound=0;
+
+static int send_sound(void)
+{
+       short myframe[FRAME_SIZE];
+       int total = FRAME_SIZE;
+       short *frame = NULL;
+       int amt=0;
+       int res;
+       int myoff;
+       audio_buf_info abi;
+       if (cursound > -1) {
+               res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
+               if (res) {
+                       ast_log(LOG_WARNING, "Unable to read output space\n");
+                       return -1;
+               }
+               /* Calculate how many samples we can send, max */
+               if (total > (abi.fragments * abi.fragsize / 2)) 
+                       total = abi.fragments * abi.fragsize / 2;
+               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;
+                               }
+                       }
+               }
+               if (frame)
+                       res = write(sounddev, frame, res * 2);
+               if (res > 0)
+                       return 0;
+               return res;
+       }
+       return 0;
+}
+
+static void *sound_thread(void *unused)
+{
+       fd_set rfds;
+       fd_set wfds;
+       int max;
+       int res;
+       char ign[4096];
+       if (read(sounddev, ign, sizeof(sounddev)) < 0)
+               ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+       for(;;) {
+               FD_ZERO(&rfds);
+               FD_ZERO(&wfds);
+               max = sndcmd[0];
+               FD_SET(sndcmd[0], &rfds);
+               if (!oss.owner) {
+                       FD_SET(sounddev, &rfds);
+                       if (sounddev > max)
+                               max = sounddev;
+               }
+               if (cursound > -1) {
+                       FD_SET(sounddev, &wfds);
+                       if (sounddev > max)
+                               max = sounddev;
+               }
+               res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
+               if (res < 1) {
+                       ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
+                       continue;
+               }
+               if (FD_ISSET(sndcmd[0], &rfds)) {
+                       read(sndcmd[0], &cursound, sizeof(cursound));
+                       silencelen = 0;
+                       offset = 0;
+                       sampsent = 0;
+               }
+               if (FD_ISSET(sounddev, &rfds)) {
+                       /* Ignore read */
+                       if (read(sounddev, ign, sizeof(ign)) < 0)
+                               ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno));
+               }
+               if (FD_ISSET(sounddev, &wfds))
+                       if (send_sound())
+                               ast_log(LOG_WARNING, "Failed to write sound\n");
+       }
+       /* Never reached */
+       return NULL;
+}
+
+#if 0
+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(sounddev, silbuf, silbufcnt * FRAME_SIZE);
+                       silbufcnt = 0;
+               }
+       }
+       return 0;
+}
+#endif
+
+static int setformat(void)
+{
+       int fmt, desired, res, fd = sounddev;
+       static int warnedalready = 0;
+       static int warnedalready2 = 0;
+
+#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;
+       }
+       res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+       
+       /* Check to see if duplex set (FreeBSD Bug)*/
+       res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt);
+       
+       if ((fmt & DSP_CAP_DUPLEX) && !res) {
+               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");
+               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 (!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
+       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;
+       readmode = 0;
+       if (force || time_has_passed()) {
+               ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+               /* 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 |O_NONBLOCK);
+               if (fd < 0) {
+                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+                       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));
+                       return -1;
+               }
+               if (setformat()) {
+                       return -1;
+               }
+               return 0;
+       }
+       return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+       int fd = sounddev;
+       if (full_duplex || (readmode && !force))
+               return 0;
+       readmode = -1;
+       if (force || time_has_passed()) {
+               ioctl(sounddev, SNDCTL_DSP_RESET, 0);
+               close(sounddev);
+               /* dup2(0, sound); */
+               fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
+               if (fd < 0) {
+                       ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+                       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));
+                       return -1;
+               }
+               if (setformat()) {
+                       return -1;
+               }
+               return 0;
+       }
+       return 1;
+}
+
+static int soundcard_init(void)
+{
+       /* Assume it's full duplex for starters */
+       int fd = open(DEV_DSP,  O_RDWR | O_NONBLOCK);
+       if (fd < 0) {
+               ast_log(LOG_WARNING, "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_text(struct ast_channel *c, const char *text)
+{
+       ast_verbose( " << Console Received text %s >> \n", text);
+       return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+       int res = 3;
+       struct ast_frame f = { 0, };
+       ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+       if (autoanswer) {
+               ast_verbose( " << Auto-answered >> \n" );
+               f.frametype = AST_FRAME_CONTROL;
+               f.subclass = AST_CONTROL_ANSWER;
+               ast_queue_frame(c, &f);
+       } else {
+               nosound = 1;
+               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);
+               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 oss_answer(struct ast_channel *c)
+{
+       ast_verbose( " << Console call has been answered >> \n");
+       answer_sound();
+       ast_setstate(c, AST_STATE_UP);
+       cursound = -1;
+       nosound=0;
+       return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+       int res = 0;
+       cursound = -1;
+       c->tech_pvt = NULL;
+       oss.owner = NULL;
+       ast_verbose( " << Hangup on console >> \n");
+       ast_mutex_lock(&usecnt_lock);
+       usecnt--;
+       ast_mutex_unlock(&usecnt_lock);
+       if (hookstate) {
+               if (autoanswer) {
+                       /* Assume auto-hangup too */
+                       hookstate = 0;
+               } else {
+                       /* Make congestion noise */
+                       res = 2;
+                       write(sndcmd[1], &res, sizeof(res));
+               }
+       }
+       return 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;
+}
+
+
+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;
+       /* Immediately return if no sound is enabled */
+       if (nosound)
+               return 0;
+       /* Stop any currently playing sound */
+       cursound = -1;
+       if (!full_duplex && !playbackonly) {
+               /* If we're half duplex, we have to switch to read mode
+                  to honor immediate needs if necessary.  But if we are in play
+                  back only mode, then we don't switch because the console
+                  is only being used one way -- just to playback something. */
+               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.samples = 0;
+       f.datalen = 0;
+       f.data = NULL;
+       f.offset = 0;
+       f.src = type;
+       f.mallocd = 0;
+       f.delivery.tv_sec = 0;
+       f.delivery.tv_usec = 0;
+       
+       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(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+       if (res < 0) {
+               ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno));
+#if 0
+               CRASH;
+#endif         
+               return NULL;
+       }
+       readpos += res;
+       
+       if (readpos >= FRAME_SIZE * 2) {
+               /* A real frame */
+               readpos = 0;
+               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 + AST_FRIENDLY_OFFSET;
+               f.offset = AST_FRIENDLY_OFFSET;
+               f.src = type;
+               f.mallocd = 0;
+               f.delivery.tv_sec = 0;
+               f.delivery.tv_usec = 0;
+#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 oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+       struct chan_oss_pvt *p = newchan->tech_pvt;
+       p->owner = newchan;
+       return 0;
+}
+
+static int oss_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;
+       case -1:
+               cursound = -1;
+               return 0;
+       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 *oss_new(struct chan_oss_pvt *p, int state)
+{
+       struct ast_channel *tmp;
+       tmp = ast_channel_alloc(1);
+       if (tmp) {
+               tmp->tech = &oss_tech;
+               snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+               tmp->type = type;
+               tmp->fds[0] = sounddev;
+               tmp->nativeformats = AST_FORMAT_SLINEAR;
+               tmp->readformat = AST_FORMAT_SLINEAR;
+               tmp->writeformat = AST_FORMAT_SLINEAR;
+               tmp->tech_pvt = p;
+               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 *oss_request(const char *type, int format, void *data, int *cause)
+{
+       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 (oss.owner) {
+               ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+               *cause = AST_CAUSE_BUSY;
+               return NULL;
+       }
+       tmp= oss_new(&oss, AST_STATE_DOWN);
+       if (!tmp) {
+               ast_log(LOG_WARNING, "Unable to create new OSS 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 'oss.conf'.\n";
+
+static int console_answer(int fd, int argc, char *argv[])
+{
+       struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       if (!oss.owner) {
+               ast_cli(fd, "No one is calling us\n");
+               return RESULT_FAILURE;
+       }
+       hookstate = 1;
+       cursound = -1;
+       ast_queue_frame(oss.owner, &f);
+       answer_sound();
+       return RESULT_SUCCESS;
+}
+
+static char sendtext_usage[] =
+"Usage: send text <message>\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;
+       char text2send[256] = "";
+       struct ast_frame f = { 0, };
+       if (argc < 2)
+               return RESULT_SHOWUSAGE;
+       if (!oss.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(oss.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[])
+{
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       cursound = -1;
+       if (!oss.owner && !hookstate) {
+               ast_cli(fd, "No call to hangup up\n");
+               return RESULT_FAILURE;
+       }
+       hookstate = 0;
+       if (oss.owner) {
+               ast_queue_hangup(oss.owner);
+       }
+       return RESULT_SUCCESS;
+}
+
+static int console_flash(int fd, int argc, char *argv[])
+{
+       struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+       if (argc != 1)
+               return RESULT_SHOWUSAGE;
+       cursound = -1;
+       if (!oss.owner) {
+               ast_cli(fd, "No call to flash\n");
+               return RESULT_FAILURE;
+       }
+       hookstate = 0;
+       if (oss.owner) {
+               ast_queue_frame(oss.owner, &f);
+       }
+       return RESULT_SUCCESS;
+}
+
+static char hangup_usage[] =
+"Usage: hangup\n"
+"       Hangs up any call currently placed on the console.\n";
+
+
+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[256], *tmp2;
+       char *mye, *myc;
+       int x;
+       struct ast_frame f = { AST_FRAME_DTMF, 0 };
+       if ((argc != 1) && (argc != 2))
+               return RESULT_SHOWUSAGE;
+       if (oss.owner) {
+               if (argc == 2) {
+                       for (x=0;x<strlen(argv[1]);x++) {
+                               f.subclass = argv[1][x];
+                               ast_queue_frame(oss.owner, &f);
+                       }
+               } 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(oss.exten, mye, sizeof(oss.exten)-1);
+               strncpy(oss.context, myc, sizeof(oss.context)-1);
+               hookstate = 1;
+               oss_new(&oss, 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[])
+{
+       char tmp[256];
+       char *context;
+       if (argc != 2)
+               return RESULT_SHOWUSAGE;
+       if (oss.owner && ast_bridged_channel(oss.owner)) {
+               strncpy(tmp, argv[1], sizeof(tmp) - 1);
+               context = strchr(tmp, '@');
+               if (context) {
+                       *context = '\0';
+                       context++;
+               } else
+                       context = oss.owner->context;
+               if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) {
+                       ast_cli(fd, "Whee, transferring %s to %s@%s.\n", 
+                                       ast_bridged_channel(oss.owner)->name, tmp, context);
+                       if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1))
+                               ast_cli(fd, "Failed to transfer :(\n");
+               } else {
+                       ast_cli(fd, "No such extension exists\n");
+               }
+       } else {
+               ast_cli(fd, "There is no call to transfer\n");
+       }
+       return RESULT_SUCCESS;
+}
+
+static char transfer_usage[] =
+"Usage: transfer <extension>[@context]\n"
+"       Transfers the currently connected call to the given extension (and\n"
+"context if specified)\n";
+
+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 }
+};
+
+int load_module()
+{
+       int res;
+       int x;
+       struct ast_config *cfg;
+       struct ast_variable *v;
+       res = pipe(sndcmd);
+       if (res) {
+               ast_log(LOG_ERROR, "Unable to create pipe\n");
+               return -1;
+       }
+       res = soundcard_init();
+       if (res < 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 OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
+               }
+               return 0;
+       }
+       if (!full_duplex)
+               ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+       res = ast_channel_register(&oss_tech);
+       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 = ast_config_load(config))) {
+               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)-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, "playbackonly"))
+                               playbackonly = ast_true(v->value);
+                       v=v->next;
+               }
+               ast_config_destroy(cfg);
+       }
+       ast_pthread_create(&sthread, NULL, sound_thread, NULL);
+       return 0;
+}
+
+
+
+int unload_module()
+{
+       int x;
+
+       ast_channel_unregister(&oss_tech);
+       for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+               ast_cli_unregister(myclis + x);
+       close(sounddev);
+       if (sndcmd[0] > 0) {
+               close(sndcmd[0]);
+               close(sndcmd[1]);
+       }
+       if (oss.owner)
+               ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD);
+       if (oss.owner)
+               return -1;
+       return 0;
+}
+
+char *description()
+{
+       return (char *) desc;
+}
+
+int usecount()
+{
+       return usecnt;
+}
+
+char *key()
+{
+       return ASTERISK_GPL_KEY;
+}