diff options
| -rw-r--r-- | CHANGES | 22 | ||||
| -rw-r--r-- | channels/chan_console.c | 1094 | ||||
| -rw-r--r-- | configs/console.conf.sample | 68 | ||||
| -rw-r--r-- | configs/modules.conf.sample | 5 |
4 files changed, 1178 insertions, 11 deletions
@@ -58,6 +58,8 @@ Dialplan functions * Added EXTENSION_STATE() dialplan function which allows retrieving the state of any extension. * Added SYSINFO() dialplan function which allows retrieval of system information + * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for + the existence of a dialplan target. CLI Changes ----------- @@ -136,11 +138,15 @@ MGCP changes ------------ * Added separate settings for media QoS in mgcp.conf -OSS Channel changes +Console Channel Driver changes ------------------- - * Added experimental support for video send&receive. - Requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 - to act as video source. + * Added experimental support for video send & receive to chan_oss. + This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as + a video source. + * Added a new channel driver, chan_console, which uses portaudio as a cross + platform audio interface. It was written as a channel driver that would + work with Mac CoreAudio, but portaudio supports a number of other audio + interfaces, as well. Phone channel changes (chan_phone) ---------------------------------- @@ -175,8 +181,8 @@ Zaptel channel driver (chan_zap) Changes * Added two new options: mwimonitor and mwimonitornotify. These options allow you to enable MWI monitoring on FXO lines. When the MWI state changes, the script specified in the mwimonitornotify option is executed. An internal - event indicating the new state of the mailbox is also generated, so that - the normal MWI facilities in Asterisk work as usual. + event indicating the new state of the mailbox is also generated, so that + the normal MWI facilities in Asterisk work as usual. A new channel driver: Unistim ----------------------------- @@ -324,7 +330,7 @@ Music On Hold Changes * Support for realtime music on hold has been added. * In conjunction with the realtime music on hold, a general section has been added to musiconhold.conf, its sole variable is cachertclasses. If this - is set, then music on hold classes found in realtime will be cached in memory. + is set, then music on hold classes found in realtime will be cached in memory. AEL Changes ----------- @@ -423,5 +429,3 @@ Miscellaneous * A new option when starting a remote asterisk (rasterisk, asterisk -r) for specifying which socket to use to connect to the running Asterisk daemon (-s) - * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for - the existence of a dialplan target. 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, +); diff --git a/configs/console.conf.sample b/configs/console.conf.sample new file mode 100644 index 000000000..820a04dd9 --- /dev/null +++ b/configs/console.conf.sample @@ -0,0 +1,68 @@ +; +; Configuration for chan_console, a cross-platform console channel driver. +; + +[general] + +; Set this option to "yes" to enable automatically answering calls on the +; console. This is very useful if the console is used as an intercom. +; The default value is "no". +; +;autoanswer = no + +; Set the default context to use for outgoing calls. This can be overridden by +; dialing some extension@context, unless the overridecontext option is enabled. +; The default is "default". +; +;context = default + +; Set the default extension to use for outgoing calls. The default is "s". +; +;extension = s + +; Set the default CallerID for created channels. +; +;callerid = MyName Here <(256) 428-6000> + +; Set the default language for created channels. +; +;language = en + +; If you set overridecontext to 'yes', then the whole dial string +; will be interpreted as an extension, which is extremely useful +; to dial SIP, IAX and other extensions which use the '@' character. +; The default is "no". +; +;overridecontext = no ; if 'no', the last @ will start the context + ; if 'yes' the whole string is an extension. + + +; Default Music on Hold class to use when this channel is placed on hold in +; the case that the music class is not set on the channel with +; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel +; putting this one on hold did not suggest a class to use. +; +;mohinterpret=default + +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an + ; Console channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The Console channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive Console side will always + ; be used if the sending side can create jitter. + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Console + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". +;----------------------------------------------------------------------------------- diff --git a/configs/modules.conf.sample b/configs/modules.conf.sample index 0c92e1d0c..8610dbca5 100644 --- a/configs/modules.conf.sample +++ b/configs/modules.conf.sample @@ -31,8 +31,9 @@ noload => pbx_kdeconsole.so ; load => res_musiconhold.so ; -; Load either OSS or ALSA, not both -; By default, load OSS only (automatically) and do not load ALSA +; Load one of: chan_oss, alsa, or console (portaudio). +; By default, load chan_oss only (automatically). ; noload => chan_alsa.so ;noload => chan_oss.so +;noload => chan_console.so |