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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-12-31 16:13:26 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-12-31 16:13:26 +0000
commit4bc50170da8e6e2863a5ae3b27ebd37921ece171 (patch)
tree1967c03ad44b5687d7c9bd31bc25eb3226094406 /channels/chan_console.c
parent1321156549b0805eb84f062331db1143ef83cd7c (diff)
Merge the main set of changes from team/russell/chan_console.
Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_console.c')
-rw-r--r--channels/chan_console.c1094
1 files changed, 1094 insertions, 0 deletions
diff --git a/channels/chan_console.c b/channels/chan_console.c
new file mode 100644
index 000000000..54f375225
--- /dev/null
+++ b/channels/chan_console.c
@@ -0,0 +1,1094 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2006 - 2007, Digium, Inc.
+ *
+ * Russell Bryant <russell@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*!
+ * \file
+ * \brief Cross-platform console channel driver
+ *
+ * \author Russell Bryant <russell@digium.com>
+ *
+ * \note Some of the code in this file came from chan_oss and chan_alsa.
+ * chan_oss, Mark Spencer <markster@digium.com>
+ * chan_oss, Luigi Rizzo
+ * chan_alsa, Matthew Fredrickson <creslin@digium.com>
+ *
+ * \ingroup channel_drivers
+ *
+ * \note Since this works with any audio system that libportaudio supports,
+ * including ALSA and OSS, this may someday deprecate chan_alsa and chan_oss.
+ * However, before that can be done, it needs to *at least* have all of the
+ * features that these other channel drivers have. The features implemented
+ * in at least one of the other console channel drivers that are not yet
+ * implemented here are:
+ *
+ * - Multiple device support
+ * - with "active" CLI command
+ * - Set Auto-answer from the dialplan
+ * - transfer CLI command
+ * - boost CLI command and .conf option
+ * - console_video support
+ */
+
+/*** MODULEINFO
+ <depend>portaudio</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <sys/signal.h> /* SIGURG */
+
+#include <portaudio.h>
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/causes.h"
+#include "asterisk/cli.h"
+#include "asterisk/musiconhold.h"
+#include "asterisk/callerid.h"
+
+/*!
+ * \brief The sample rate to request from PortAudio
+ *
+ * \note This should be changed to 16000 once there is a translator for going
+ * between SLINEAR and SLINEAR16. Making it a configuration parameter
+ * would be even better, but 16 kHz should be the default.
+ *
+ * \note If this changes, NUM_SAMPLES will need to change, as well.
+ */
+#define SAMPLE_RATE 8000
+
+/*!
+ * \brief The number of samples to configure the portaudio stream for
+ *
+ * 160 samples (20 ms) is the most common frame size in Asterisk. So, the code
+ * in this module reads 160 sample frames from the portaudio stream and queues
+ * them up on the Asterisk channel. Frames of any sizes can be written to a
+ * portaudio stream, but the portaudio documentation does say that for high
+ * performance applications, the data should be written to Pa_WriteStream in
+ * the same size as what is used to initialize the stream.
+ *
+ * \note This will need to be dynamic once the sample rate can be something
+ * other than 8 kHz.
+ */
+#define NUM_SAMPLES 160
+
+/*! \brief Mono Input */
+#define INPUT_CHANNELS 1
+
+/*! \brief Mono Output */
+#define OUTPUT_CHANNELS 1
+
+/*!
+ * \brief Maximum text message length
+ * \note This should be changed if there is a common definition somewhere
+ * that defines the maximum length of a text message.
+ */
+#define TEXT_SIZE 256
+
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+#ifndef MAX
+#define MAX(a,b) ((a) > (b) ? (a) : (b))
+#endif
+
+/*! \brief Dance, Kirby, Dance! @{ */
+#define V_BEGIN " --- <(\"<) --- "
+#define V_END " --- (>\")> ---\n"
+/*! @} */
+
+static const char config_file[] = "console.conf";
+
+/*!
+ * \brief Console pvt structure
+ *
+ * Currently, this is a singleton object. However, multiple instances will be
+ * needed when this module is updated for multiple device support.
+ */
+static struct console_pvt {
+ AST_DECLARE_STRING_FIELDS(
+ /*! Name of the device */
+ AST_STRING_FIELD(name);
+ /*! Default context for outgoing calls */
+ AST_STRING_FIELD(context);
+ /*! Default extension for outgoing calls */
+ AST_STRING_FIELD(exten);
+ /*! Default CallerID number */
+ AST_STRING_FIELD(cid_num);
+ /*! Default CallerID name */
+ AST_STRING_FIELD(cid_name);
+ /*! Default MOH class to listen to, if:
+ * - No MOH class set on the channel
+ * - Peer channel putting this device on hold did not suggest a class */
+ AST_STRING_FIELD(mohinterpret);
+ /*! Default language */
+ AST_STRING_FIELD(language);
+ );
+ /*! Current channel for this device */
+ struct ast_channel *owner;
+ /*! Current PortAudio stream for this device */
+ PaStream *stream;
+ /*! A frame for preparing to queue on to the channel */
+ struct ast_frame fr;
+ /*! Running = 1, Not running = 0 */
+ unsigned int streamstate:1;
+ /*! On-hook = 0, Off-hook = 1 */
+ unsigned int hookstate:1;
+ /*! Unmuted = 0, Muted = 1 */
+ unsigned int muted:1;
+ /*! Automatically answer incoming calls */
+ unsigned int autoanswer:1;
+ /*! Ignore context in the console dial CLI command */
+ unsigned int overridecontext:1;
+ /*! Lock to protect data in this struct */
+ ast_mutex_t __lock;
+ /*! ID for the stream monitor thread */
+ pthread_t thread;
+} console_pvt = {
+ .__lock = AST_MUTEX_INIT_VALUE,
+ .thread = AST_PTHREADT_NULL,
+};
+
+/*!
+ * \brief Global jitterbuffer configuration
+ *
+ * \note Disabled by default.
+ */
+static struct ast_jb_conf default_jbconf = {
+ .flags = 0,
+ .max_size = -1,
+ .resync_threshold = -1,
+ .impl = ""
+};
+static struct ast_jb_conf global_jbconf;
+
+/*! Channel Technology Callbacks @{ */
+static struct ast_channel *console_request(const char *type, int format,
+ void *data, int *cause);
+static int console_digit_begin(struct ast_channel *c, char digit);
+static int console_digit_end(struct ast_channel *c, char digit, unsigned int duration);
+static int console_text(struct ast_channel *c, const char *text);
+static int console_hangup(struct ast_channel *c);
+static int console_answer(struct ast_channel *c);
+static struct ast_frame *console_read(struct ast_channel *chan);
+static int console_call(struct ast_channel *c, char *dest, int timeout);
+static int console_write(struct ast_channel *chan, struct ast_frame *f);
+static int console_indicate(struct ast_channel *chan, int cond,
+ const void *data, size_t datalen);
+static int console_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+/*! @} */
+
+/*!
+ * \brief Formats natively supported by this module.
+ *
+ * \note Once 16 kHz is supported, AST_FORMAT_SLINEAR16 needs to be added.
+ */
+#define SUPPORTED_FORMATS ( AST_FORMAT_SLINEAR )
+
+static const struct ast_channel_tech console_tech = {
+ .type = "Console",
+ .description = "Console Channel Driver",
+ .capabilities = SUPPORTED_FORMATS,
+ .requester = console_request,
+ .send_digit_begin = console_digit_begin,
+ .send_digit_end = console_digit_end,
+ .send_text = console_text,
+ .hangup = console_hangup,
+ .answer = console_answer,
+ .read = console_read,
+ .call = console_call,
+ .write = console_write,
+ .indicate = console_indicate,
+ .fixup = console_fixup,
+};
+
+/*! \brief lock a console_pvt struct */
+#define console_pvt_lock(pvt) ast_mutex_lock(&(pvt)->__lock)
+
+/*! \brief unlock a console_pvt struct */
+#define console_pvt_unlock(pvt) ast_mutex_unlock(&(pvt)->__lock)
+
+/*!
+ * \brief Stream monitor thread
+ *
+ * \arg data A pointer to the console_pvt structure that contains the portaudio
+ * stream that needs to be monitored.
+ *
+ * This function runs in its own thread to monitor data coming in from a
+ * portaudio stream. When enough data is available, it is queued up to
+ * be read from the Asterisk channel.
+ */
+static void *stream_monitor(void *data)
+{
+ struct console_pvt *pvt = data;
+ char buf[NUM_SAMPLES * sizeof(int16_t)];
+ PaError res;
+ struct ast_frame f = {
+ .frametype = AST_FRAME_VOICE,
+ .subclass = AST_FORMAT_SLINEAR,
+ .src = "console_stream_monitor",
+ .data = buf,
+ .datalen = sizeof(buf),
+ .samples = sizeof(buf) / sizeof(int16_t),
+ };
+
+ for (;;) {
+ pthread_testcancel();
+ res = Pa_ReadStream(pvt->stream, buf, sizeof(buf) / sizeof(int16_t));
+ pthread_testcancel();
+
+ if (res == paNoError)
+ ast_queue_frame(pvt->owner, &f);
+ }
+
+ return NULL;
+}
+
+static int start_stream(struct console_pvt *pvt)
+{
+ PaError res;
+ int ret_val = 0;
+
+ console_pvt_lock(pvt);
+
+ if (pvt->streamstate)
+ goto return_unlock;
+
+ pvt->streamstate = 1;
+ ast_debug(1, "Starting stream\n");
+
+ res = Pa_OpenDefaultStream(&pvt->stream, INPUT_CHANNELS, OUTPUT_CHANNELS,
+ paInt16, SAMPLE_RATE, NUM_SAMPLES, NULL, NULL);
+ if (res != paNoError) {
+ ast_log(LOG_WARNING, "Failed to open default audio device - (%d) %s\n",
+ res, Pa_GetErrorText(res));
+ ret_val = -1;
+ goto return_unlock;
+ }
+
+ res = Pa_StartStream(pvt->stream);
+ if (res != paNoError) {
+ ast_log(LOG_WARNING, "Failed to start stream - (%d) %s\n",
+ res, Pa_GetErrorText(res));
+ ret_val = -1;
+ goto return_unlock;
+ }
+
+ if (ast_pthread_create_background(&pvt->thread, NULL, stream_monitor, pvt)) {
+ ast_log(LOG_ERROR, "Failed to start stream monitor thread\n");
+ ret_val = -1;
+ }
+
+return_unlock:
+ console_pvt_unlock(pvt);
+
+ return ret_val;
+}
+
+static int stop_stream(struct console_pvt *pvt)
+{
+ if (!pvt->streamstate)
+ return 0;
+
+ pthread_cancel(pvt->thread);
+ pthread_kill(pvt->thread, SIGURG);
+ pthread_join(pvt->thread, NULL);
+
+ console_pvt_lock(pvt);
+ Pa_AbortStream(pvt->stream);
+ Pa_CloseStream(pvt->stream);
+ pvt->stream = NULL;
+ pvt->streamstate = 0;
+ console_pvt_unlock(pvt);
+
+ return 0;
+}
+
+/*!
+ * \note Called with the pvt struct locked
+ */
+static struct ast_channel *console_new(struct console_pvt *pvt, const char *ext, const char *ctx, int state)
+{
+ struct ast_channel *chan;
+
+ if (!(chan = ast_channel_alloc(1, state, pvt->cid_num, pvt->cid_name, NULL,
+ ext, ctx, 0, "Console/%s", pvt->name))) {
+ return NULL;
+ }
+
+ chan->tech = &console_tech;
+ chan->nativeformats = AST_FORMAT_SLINEAR;
+ chan->readformat = AST_FORMAT_SLINEAR;
+ chan->writeformat = AST_FORMAT_SLINEAR;
+ chan->tech_pvt = pvt;
+
+ pvt->owner = chan;
+
+ if (!ast_strlen_zero(pvt->language))
+ ast_string_field_set(chan, language, pvt->language);
+
+ ast_jb_configure(chan, &global_jbconf);
+
+ if (state != AST_STATE_DOWN) {
+ if (ast_pbx_start(chan)) {
+ chan->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
+ ast_hangup(chan);
+ chan = NULL;
+ } else
+ start_stream(pvt);
+ }
+
+ return chan;
+}
+
+static struct ast_channel *console_request(const char *type, int format, void *data, int *cause)
+{
+ int oldformat = format;
+ struct ast_channel *chan;
+ struct console_pvt *pvt = &console_pvt;
+
+ format &= SUPPORTED_FORMATS;
+ if (!format) {
+ ast_log(LOG_NOTICE, "Channel requested with unsupported format(s): '%d'\n", oldformat);
+ return NULL;
+ }
+
+ if (pvt->owner) {
+ ast_log(LOG_NOTICE, "Console channel already active!\n");
+ *cause = AST_CAUSE_BUSY;
+ return NULL;
+ }
+
+ console_pvt_lock(pvt);
+ chan = console_new(pvt, NULL, NULL, AST_STATE_DOWN);
+ console_pvt_unlock(pvt);
+
+ if (!chan)
+ ast_log(LOG_WARNING, "Unable to create new Console channel!\n");
+
+ return chan;
+}
+
+static int console_digit_begin(struct ast_channel *c, char digit)
+{
+ ast_verb(1, V_BEGIN "Console Received Beginning of Digit %c" V_END, digit);
+
+ return -1; /* non-zero to request inband audio */
+}
+
+static int console_digit_end(struct ast_channel *c, char digit, unsigned int duration)
+{
+ ast_verb(1, V_BEGIN "Console Received End of Digit %c (duration %u)" V_END,
+ digit, duration);
+
+ return -1; /* non-zero to request inband audio */
+}
+
+static int console_text(struct ast_channel *c, const char *text)
+{
+ ast_verb(1, V_BEGIN "Console Received Text '%s'" V_END, text);
+
+ return 0;
+}
+
+static int console_hangup(struct ast_channel *c)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ ast_verb(1, V_BEGIN "Hangup on Console" V_END);
+
+ pvt->hookstate = 0;
+ c->tech_pvt = NULL;
+ pvt->owner = NULL;
+
+ stop_stream(pvt);
+
+ return 0;
+}
+
+static int console_answer(struct ast_channel *c)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ ast_verb(1, V_BEGIN "Call from Console has been Answered" V_END);
+
+ ast_setstate(c, AST_STATE_UP);
+
+ return start_stream(pvt);
+}
+
+/*
+ * \brief Implementation of the ast_channel_tech read() callback
+ *
+ * Calling this function is harmless. However, if it does get called, it
+ * is an indication that something weird happened that really shouldn't
+ * have and is worth looking into.
+ *
+ * Why should this function not get called? Well, let me explain. There are
+ * a couple of ways to pass on audio that has come from this channel. The way
+ * that this channel driver uses is that once the audio is available, it is
+ * wrapped in an ast_frame and queued onto the channel using ast_queue_frame().
+ *
+ * The other method would be signalling to the core that there is audio waiting,
+ * and that it needs to call the channel's read() callback to get it. The way
+ * the channel gets signalled is that one or more file descriptors are placed
+ * in the fds array on the ast_channel which the core will poll() on. When the
+ * fd indicates that input is available, the read() callback is called. This
+ * is especially useful when there is a dedicated file descriptor where the
+ * audio is read from. An example would be the socket for an RTP stream.
+ */
+static struct ast_frame *console_read(struct ast_channel *chan)
+{
+ ast_debug(1, "I should not be called ...\n");
+
+ return &ast_null_frame;
+}
+
+static int console_call(struct ast_channel *c, char *dest, int timeout)
+{
+ struct ast_frame f = { 0, };
+ struct console_pvt *pvt = &console_pvt;
+
+ ast_verb(1, V_BEGIN "Call to device '%s' on console from '%s' <%s>" V_END,
+ dest, c->cid.cid_name, c->cid.cid_num);
+
+ console_pvt_lock(pvt);
+
+ if (pvt->autoanswer) {
+ ast_verb(1, V_BEGIN "Auto-answered" V_END);
+ pvt->hookstate = 1;
+ f.frametype = AST_FRAME_CONTROL;
+ f.subclass = AST_CONTROL_ANSWER;
+ } else {
+ ast_verb(1, V_BEGIN "Type 'answer' to answer, or use 'autoanswer' "
+ "for future calls" V_END);
+ f.frametype = AST_FRAME_CONTROL;
+ f.subclass = AST_CONTROL_RINGING;
+ }
+
+ console_pvt_unlock(pvt);
+
+ ast_queue_frame(c, &f);
+
+ return start_stream(pvt);
+}
+
+static int console_write(struct ast_channel *chan, struct ast_frame *f)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ Pa_WriteStream(pvt->stream, f->data, f->samples);
+
+ return 0;
+}
+
+static int console_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
+{
+ struct console_pvt *pvt = chan->tech_pvt;
+ int res = 0;
+
+ switch (cond) {
+ case AST_CONTROL_BUSY:
+ case AST_CONTROL_CONGESTION:
+ case AST_CONTROL_RINGING:
+ res = -1; /* Ask for inband indications */
+ break;
+ case AST_CONTROL_PROGRESS:
+ case AST_CONTROL_PROCEEDING:
+ case AST_CONTROL_VIDUPDATE:
+ case -1:
+ break;
+ case AST_CONTROL_HOLD:
+ ast_verb(1, V_BEGIN "Console Has Been Placed on Hold" V_END);
+ ast_moh_start(chan, data, pvt->mohinterpret);
+ break;
+ case AST_CONTROL_UNHOLD:
+ ast_verb(1, V_BEGIN "Console Has Been Retrieved from Hold" V_END);
+ ast_moh_stop(chan);
+ break;
+ default:
+ ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n",
+ cond, chan->name);
+ /* The core will play inband indications for us if appropriate */
+ res = -1;
+ }
+
+ return res;
+}
+
+static int console_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ pvt->owner = newchan;
+
+ return 0;
+}
+
+/*!
+ * split a string in extension-context, returns pointers to malloc'ed
+ * strings.
+ * If we do not have 'overridecontext' then the last @ is considered as
+ * a context separator, and the context is overridden.
+ * This is usually not very necessary as you can play with the dialplan,
+ * and it is nice not to need it because you have '@' in SIP addresses.
+ * Return value is the buffer address.
+ *
+ * \note came from chan_oss
+ */
+static char *ast_ext_ctx(struct console_pvt *pvt, const char *src, char **ext, char **ctx)
+{
+ if (ext == NULL || ctx == NULL)
+ return NULL; /* error */
+
+ *ext = *ctx = NULL;
+
+ if (src && *src != '\0')
+ *ext = ast_strdup(src);
+
+ if (*ext == NULL)
+ return NULL;
+
+ if (!pvt->overridecontext) {
+ /* parse from the right */
+ *ctx = strrchr(*ext, '@');
+ if (*ctx)
+ *(*ctx)++ = '\0';
+ }
+
+ return *ext;
+}
+
+static char *cli_console_autoanswer(struct ast_cli_entry *e, int cmd,
+ struct ast_cli_args *a)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "console set autoanswer [on|off]";
+ e->usage =
+ "Usage: console set 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";
+ return NULL;
+
+ case CLI_GENERATE:
+ return NULL;
+ }
+
+ if (a->argc == e->args - 1) {
+ ast_cli(a->fd, "Auto answer is %s.\n", pvt->autoanswer ? "on" : "off");
+ return CLI_SUCCESS;
+ }
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!pvt) {
+ ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n",
+ pvt->name);
+ return CLI_FAILURE;
+ }
+
+ if (!strcasecmp(a->argv[e->args-1], "on"))
+ pvt->autoanswer = 1;
+ else if (!strcasecmp(a->argv[e->args - 1], "off"))
+ pvt->autoanswer = 0;
+ else
+ return CLI_SHOWUSAGE;
+
+ return CLI_SUCCESS;
+}
+
+static char *cli_console_flash(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH };
+ struct console_pvt *pvt = &console_pvt;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console flash";
+ e->usage =
+ "Usage: console flash\n"
+ " Flashes the call currently placed on the console.\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!pvt->owner) {
+ ast_cli(a->fd, "No call to flash\n");
+ return CLI_FAILURE;
+ }
+
+ pvt->hookstate = 0;
+
+ ast_queue_frame(pvt->owner, &f);
+
+ return CLI_SUCCESS;
+}
+
+static char *cli_console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ char *s = NULL;
+ const char *mye = NULL, *myc = NULL;
+ struct console_pvt *pvt = &console_pvt;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console dial";
+ e->usage =
+ "Usage: console dial [extension[@context]]\n"
+ " Dials a given extension (and context if specified)\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc > e->args + 1)
+ return CLI_SHOWUSAGE;
+
+ if (pvt->owner) { /* already in a call */
+ int i;
+ struct ast_frame f = { AST_FRAME_DTMF, 0 };
+
+ if (a->argc == e->args) { /* argument is mandatory here */
+ ast_cli(a->fd, "Already in a call. You can only dial digits until you hangup.\n");
+ return CLI_FAILURE;
+ }
+ s = a->argv[e->args];
+ /* send the string one char at a time */
+ for (i = 0; i < strlen(s); i++) {
+ f.subclass = s[i];
+ ast_queue_frame(pvt->owner, &f);
+ }
+ return CLI_SUCCESS;
+ }
+
+ /* if we have an argument split it into extension and context */
+ if (a->argc == e->args + 1) {
+ char *ext = NULL, *con = NULL;
+ s = ast_ext_ctx(pvt, a->argv[e->args], &ext, &con);
+ ast_debug(1, "provided '%s', exten '%s' context '%s'\n",
+ a->argv[e->args], mye, myc);
+ mye = ext;
+ myc = con;
+ }
+
+ /* supply default values if needed */
+ if (ast_strlen_zero(mye))
+ mye = pvt->exten;
+ if (ast_strlen_zero(myc))
+ myc = pvt->context;
+
+ if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
+ console_pvt_lock(pvt);
+ pvt->hookstate = 1;
+ console_new(pvt, mye, myc, AST_STATE_RINGING);
+ console_pvt_unlock(pvt);
+ } else
+ ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
+
+ if (s)
+ free(s);
+
+ return CLI_SUCCESS;
+}
+
+static char *cli_console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console hangup";
+ e->usage =
+ "Usage: console hangup\n"
+ " Hangs up any call currently placed on the console.\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!pvt->owner && !pvt->hookstate) {
+ ast_cli(a->fd, "No call to hang up\n");
+ return CLI_FAILURE;
+ }
+
+ pvt->hookstate = 0;
+ if (pvt->owner)
+ ast_queue_hangup(pvt->owner);
+
+ return CLI_SUCCESS;
+}
+
+static char *cli_console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ char *s;
+ struct console_pvt *pvt = &console_pvt;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console {mute|unmute}";
+ e->usage =
+ "Usage: console {mute|unmute}\n"
+ " Mute/unmute the microphone.\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ s = a->argv[e->args-1];
+ if (!strcasecmp(s, "mute"))
+ pvt->muted = 1;
+ else if (!strcasecmp(s, "unmute"))
+ pvt->muted = 0;
+ else
+ return CLI_SHOWUSAGE;
+
+ ast_verb(1, V_BEGIN "The Console is now %s" V_END,
+ pvt->muted ? "Muted" : "Unmuted");
+
+ return CLI_SUCCESS;
+}
+
+static char *cli_list_devices(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ PaDeviceIndex index, num, def_input, def_output;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console list devices";
+ e->usage =
+ "Usage: console list devices\n"
+ " List all available devices.\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ ast_cli(a->fd, "Available Devices:\n---------------------------------\n");
+
+ num = Pa_GetDeviceCount();
+ if (!num) {
+ ast_cli(a->fd, "(None)\n");
+ return CLI_SUCCESS;
+ }
+
+ def_input = Pa_GetDefaultInputDevice();
+ def_output = Pa_GetDefaultOutputDevice();
+ for (index = 0; index < num; index++) {
+ const PaDeviceInfo *dev = Pa_GetDeviceInfo(index);
+ if (!dev)
+ continue;
+ ast_cli(a->fd, "Device Name: %s\n", dev->name);
+ if (index == def_input)
+ ast_cli(a->fd, " ---> Default Input Device\n");
+ if (index == def_output)
+ ast_cli(a->fd, " ---> Default Output Device\n");
+ }
+
+ return CLI_SUCCESS;
+}
+
+/*!
+ * \brief answer command from the console
+ */
+static char *cli_console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
+ struct console_pvt *pvt = &console_pvt;
+
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "console answer";
+ e->usage =
+ "Usage: console answer\n"
+ " Answers an incoming call on the console channel.\n";
+ return NULL;
+
+ case CLI_GENERATE:
+ return NULL; /* no completion */
+ }
+
+ if (a->argc != e->args)
+ return CLI_SHOWUSAGE;
+
+ if (!pvt->owner) {
+ ast_cli(a->fd, "No one is calling us\n");
+ return CLI_FAILURE;
+ }
+
+ pvt->hookstate = 1;
+ ast_queue_frame(pvt->owner, &f);
+
+ return CLI_SUCCESS;
+}
+
+/*!
+ * \brief Console send text CLI command
+ *
+ * \note concatenate all arguments into a single string. argv is NULL-terminated
+ * so we can use it right away
+ */
+static char *cli_console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+ char buf[TEXT_SIZE];
+ struct console_pvt *pvt = &console_pvt;
+ struct ast_frame f = {
+ .frametype = AST_FRAME_TEXT,
+ .data = buf,
+ .src = "console_send_text",
+ };
+ int len;
+
+ if (cmd == CLI_INIT) {
+ e->command = "console send text";
+ e->usage =
+ "Usage: console send text <message>\n"
+ " Sends a text message for display on the remote terminal.\n";
+ return NULL;
+ } else if (cmd == CLI_GENERATE)
+ return NULL;
+
+ if (a->argc < e->args + 1)
+ return CLI_SHOWUSAGE;
+
+ if (!pvt->owner) {
+ ast_cli(a->fd, "Not in a call\n");
+ return CLI_FAILURE;
+ }
+
+ ast_join(buf, sizeof(buf) - 1, a->argv + e->args);
+ if (ast_strlen_zero(buf))
+ return CLI_SHOWUSAGE;
+
+ len = strlen(buf);
+ buf[len] = '\n';
+ f.datalen = len + 1;
+
+ ast_queue_frame(pvt->owner, &f);
+
+ return CLI_SUCCESS;
+}
+
+static struct ast_cli_entry cli_console[] = {
+ AST_CLI_DEFINE(cli_console_dial, "Dial an extension from the console"),
+ AST_CLI_DEFINE(cli_console_hangup, "Hangup a call on the console"),
+ AST_CLI_DEFINE(cli_console_mute, "Disable/Enable mic input"),
+ AST_CLI_DEFINE(cli_console_answer, "Answer an incoming console call"),
+ AST_CLI_DEFINE(cli_console_sendtext, "Send text to a connected party"),
+ AST_CLI_DEFINE(cli_console_flash, "Send a flash to the connected party"),
+ AST_CLI_DEFINE(cli_console_autoanswer, "Turn autoanswer on or off"),
+ AST_CLI_DEFINE(cli_list_devices, "List available devices"),
+};
+
+/*!
+ * \brief Set default values for a pvt struct
+ *
+ * \note This function expects the pvt lock to be held.
+ */
+static void set_pvt_defaults(struct console_pvt *pvt, int reload)
+{
+ if (!reload) {
+ /* This should be changed for multiple device support. Right now,
+ * there is no way to change the name of a device. The default
+ * input and output sound devices are the only ones supported. */
+ ast_string_field_set(pvt, name, "default");
+ }
+
+ ast_string_field_set(pvt, mohinterpret, "default");
+ ast_string_field_set(pvt, context, "default");
+ ast_string_field_set(pvt, exten, "s");
+ ast_string_field_set(pvt, language, "");
+ ast_string_field_set(pvt, cid_num, "");
+ ast_string_field_set(pvt, cid_name, "");
+
+ pvt->overridecontext = 0;
+ pvt->autoanswer = 0;
+}
+
+static void store_callerid(struct console_pvt *pvt, const char *value)
+{
+ char cid_name[256];
+ char cid_num[256];
+
+ ast_callerid_split(value, cid_name, sizeof(cid_name),
+ cid_num, sizeof(cid_num));
+
+ ast_string_field_set(pvt, cid_name, cid_name);
+ ast_string_field_set(pvt, cid_num, cid_num);
+}
+
+/*!
+ * \brief Store a configuration parameter in a pvt struct
+ *
+ * \note This function expects the pvt lock to be held.
+ */
+static void store_config_core(struct console_pvt *pvt, const char *var, const char *value)
+{
+ if (!ast_jb_read_conf(&global_jbconf, var, value))
+ return;
+
+ CV_START(var, value);
+
+ CV_STRFIELD("context", pvt, context);
+ CV_STRFIELD("extension", pvt, exten);
+ CV_STRFIELD("mohinterpret", pvt, mohinterpret);
+ CV_STRFIELD("language", pvt, language);
+ CV_F("callerid", store_callerid(pvt, value));
+ CV_BOOL("overridecontext", pvt->overridecontext);
+ CV_BOOL("autoanswer", pvt->autoanswer);
+
+ ast_log(LOG_WARNING, "Unknown option '%s'\n", var);
+
+ CV_END;
+}
+
+/*!
+ * \brief Load the configuration
+ * \param reload if this was called due to a reload
+ * \retval 0 succcess
+ * \retval -1 failure
+ */
+static int load_config(int reload)
+{
+ struct ast_config *cfg;
+ struct ast_variable *v;
+ struct console_pvt *pvt = &console_pvt;
+ struct ast_flags config_flags = { 0 };
+ int res = -1;
+
+ /* default values */
+ memcpy(&global_jbconf, &default_jbconf, sizeof(global_jbconf));
+
+ console_pvt_lock(pvt);
+
+ set_pvt_defaults(pvt, reload);
+
+ if (!(cfg = ast_config_load(config_file, config_flags))) {
+ ast_log(LOG_NOTICE, "Unable to open configuration file %s!\n", config_file);
+ goto return_unlock;
+ }
+
+ for (v = ast_variable_browse(cfg, "general"); v; v = v->next)
+ store_config_core(pvt, v->name, v->value);
+
+ ast_config_destroy(cfg);
+
+ res = 0;
+
+return_unlock:
+ console_pvt_unlock(pvt);
+ return res;
+}
+
+static int init_pvt(struct console_pvt *pvt)
+{
+ if (ast_string_field_init(pvt, 32))
+ return -1;
+
+ if (ast_mutex_init(&pvt->__lock)) {
+ ast_log(LOG_ERROR, "Failed to initialize mutex\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static void destroy_pvt(struct console_pvt *pvt)
+{
+ ast_string_field_free_memory(pvt);
+
+ ast_mutex_destroy(&pvt->__lock);
+}
+
+static int unload_module(void)
+{
+ struct console_pvt *pvt = &console_pvt;
+
+ if (pvt->hookstate)
+ stop_stream(pvt);
+
+ Pa_Terminate();
+
+ ast_channel_unregister(&console_tech);
+ ast_cli_unregister_multiple(cli_console, ARRAY_LEN(cli_console));
+
+ destroy_pvt(pvt);
+
+ return 0;
+}
+
+static int load_module(void)
+{
+ PaError res;
+ struct console_pvt *pvt = &console_pvt;
+
+ if (init_pvt(pvt))
+ goto return_error;
+
+ if (load_config(0))
+ goto return_error;
+
+ res = Pa_Initialize();
+ if (res != paNoError) {
+ ast_log(LOG_WARNING, "Failed to initialize audio system - (%d) %s\n",
+ res, Pa_GetErrorText(res));
+ goto return_error_pa_init;
+ }
+
+ if (ast_channel_register(&console_tech)) {
+ ast_log(LOG_ERROR, "Unable to register channel type 'Console'\n");
+ goto return_error_chan_reg;
+ }
+
+ if (ast_cli_register_multiple(cli_console, ARRAY_LEN(cli_console)))
+ goto return_error_cli_reg;
+
+ return AST_MODULE_LOAD_SUCCESS;
+
+return_error_cli_reg:
+ ast_cli_unregister_multiple(cli_console, ARRAY_LEN(cli_console));
+return_error_chan_reg:
+ ast_channel_unregister(&console_tech);
+return_error_pa_init:
+ Pa_Terminate();
+return_error:
+ destroy_pvt(pvt);
+
+ return AST_MODULE_LOAD_DECLINE;
+}
+
+static int reload(void)
+{
+ return load_config(1);
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Console Channel Driver",
+ .load = load_module,
+ .unload = unload_module,
+ .reload = reload,
+);