[asterisk-commits] mmichelson: trunk r100678 - in /trunk: ./ channels/ include/asterisk/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 28 15:07:19 CST 2008
Author: mmichelson
Date: Mon Jan 28 15:07:18 2008
New Revision: 100678
URL: http://svn.digium.com/view/asterisk?view=rev&rev=100678
Log:
Re-inserting chan_vpb into trunk.
Added:
trunk/channels/chan_vpb.cc (with props)
Modified:
trunk/channels/Makefile
trunk/configure
trunk/configure.ac
trunk/include/asterisk/autoconfig.h.in
Modified: trunk/channels/Makefile
URL: http://svn.digium.com/view/asterisk/trunk/channels/Makefile?view=diff&rev=100678&r1=100677&r2=100678
==============================================================================
--- trunk/channels/Makefile (original)
+++ trunk/channels/Makefile Mon Jan 28 15:07:18 2008
@@ -100,4 +100,6 @@
$(if $(filter chan_misdn,$(EMBEDDED_MODS)),modules.link,chan_misdn.so): misdn_config.o misdn/isdn_lib.o misdn/isdn_msg_parser.o
+chan_vpb.oo: ASTCFLAGS:=$(filter-out -Wdeclaration-after-statement,$(ASTCFLAGS))
+
$(if $(filter chan_oss,$(EMBEDDED_MODS)),modules.link,chan_oss.so): console_video.o vgrabbers.o console_board.o
Added: trunk/channels/chan_vpb.cc
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_vpb.cc?view=auto&rev=100678
==============================================================================
--- trunk/channels/chan_vpb.cc (added)
+++ trunk/channels/chan_vpb.cc Mon Jan 28 15:07:18 2008
@@ -1,0 +1,2899 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2003, Paul Bagyenda
+ * Paul Bagyenda <bagyenda at dsmagic.com>
+ * Copyright (C) 2004 - 2005, Ben Kramer
+ * Ben Kramer <ben at voicetronix.com.au>
+ *
+ * Daniel Bichara <daniel at bichara.com.br> - Brazilian CallerID detection (c)2004
+ *
+ * Welber Silveira - welberms at magiclink.com.br - (c)2004
+ * Copying CLID string to propper structure after detection
+ *
+ * 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 VoiceTronix Interface driver
+ *
+ * \ingroup channel_drivers
+ */
+
+/*** MODULEINFO
+ <depend>vpbapi</depend>
+ ***/
+
+extern "C" {
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/lock.h"
+#include "asterisk/utils.h"
+#include "asterisk/channel.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/callerid.h"
+#include "asterisk/dsp.h"
+#include "asterisk/features.h"
+#include "asterisk/musiconhold.h"
+}
+
+#include <sys/socket.h>
+#include <sys/time.h>
+#include <arpa/inet.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <ctype.h>
+
+#include <vpbapi.h>
+#include <assert.h>
+
+#ifdef pthread_create
+#undef pthread_create
+#endif
+
+#define DEFAULT_GAIN 0
+#define DEFAULT_ECHO_CANCEL 1
+
+#define VPB_SAMPLES 160
+#define VPB_MAX_BUF VPB_SAMPLES*4 + AST_FRIENDLY_OFFSET
+
+#define VPB_NULL_EVENT 200
+
+#define VPB_WAIT_TIMEOUT 4000
+
+#define MAX_VPB_GAIN 12.0
+#define MIN_VPB_GAIN -12.0
+
+#define DTMF_CALLERID
+#define DTMF_CID_START 'D'
+#define DTMF_CID_STOP 'C'
+
+/**/
+#if defined(__cplusplus) || defined(c_plusplus)
+ extern "C" {
+#endif
+/**/
+
+static const char desc[] = "VoiceTronix V6PCI/V12PCI/V4PCI API Support";
+static const char tdesc[] = "Standard VoiceTronix API Driver";
+static const char config[] = "vpb.conf";
+
+/* Default context for dialtone mode */
+static char context[AST_MAX_EXTENSION] = "default";
+
+/* Default language */
+static char language[MAX_LANGUAGE] = "";
+
+static int gruntdetect_timeout = 3600000; /* Grunt detect timeout is 1hr. */
+
+static const int prefformat = AST_FORMAT_SLINEAR;
+
+/* Protect the interface list (of vpb_pvt's) */
+AST_MUTEX_DEFINE_STATIC(iflock);
+
+/* Protect the monitoring thread, so only one process can kill or start it, and not
+ when it's doing something critical. */
+AST_MUTEX_DEFINE_STATIC(monlock);
+
+/* This is the thread for the monitor which checks for input on the channels
+ which are not currently in use. */
+static pthread_t monitor_thread;
+
+static int mthreadactive = -1; /* Flag for monitoring monitorthread.*/
+
+
+static int restart_monitor(void);
+
+/* The private structures of the VPB channels are
+ linked for selecting outgoing channels */
+
+#define MODE_DIALTONE 1
+#define MODE_IMMEDIATE 2
+#define MODE_FXO 3
+
+/* Pick a country or add your own! */
+/* These are the tones that are played to the user */
+#define TONES_AU
+/* #define TONES_USA */
+
+#ifdef TONES_AU
+static VPB_TONE Dialtone = {440, 440, 440, -10, -10, -10, 5000, 0 };
+static VPB_TONE Busytone = {470, 0, 0, -10, -100, -100, 5000, 0 };
+static VPB_TONE Ringbacktone = {400, 50, 440, -10, -10, -10, 1400, 800 };
+#endif
+#ifdef TONES_USA
+static VPB_TONE Dialtone = {350, 440, 0, -16, -16, -100, 10000, 0};
+static VPB_TONE Busytone = {480, 620, 0, -10, -10, -100, 500, 500};
+static VPB_TONE Ringbacktone = {440, 480, 0, -20, -20, -100, 2000, 4000};
+#endif
+
+/* grunt tone defn's */
+#if 0
+static VPB_DETECT toned_grunt = { 3, VPB_GRUNT, 1, 2000, 3000, 0, 0, -40, 0, 0, 0, 40, { { VPB_DELAY, 1000, 0, 0 }, { VPB_RISING, 0, 40, 0 }, { 0, 100, 0, 0 } } };
+#endif
+static VPB_DETECT toned_ungrunt = { 2, VPB_GRUNT, 1, 2000, 1, 0, 0, -40, 0, 0, 30, 40, { { 0, 0, 0, 0 } } };
+
+/* Use loop polarity detection for CID */
+static int UsePolarityCID=0;
+
+/* Use loop drop detection */
+static int UseLoopDrop=1;
+
+/* To use or not to use Native bridging */
+static int UseNativeBridge=1;
+
+/* Use Asterisk Indication or VPB */
+static int use_ast_ind=0;
+
+/* Use Asterisk DTMF detection or VPB */
+static int use_ast_dtmfdet=0;
+
+static int relaxdtmf=0;
+
+/* Use Asterisk DTMF play back or VPB */
+static int use_ast_dtmf=0;
+
+/* Break for DTMF on native bridge ? */
+static int break_for_dtmf=1;
+
+/* Set EC suppression threshold */
+static int ec_supp_threshold=-1;
+
+/* Inter Digit Delay for collecting DTMF's */
+static int dtmf_idd = 3000;
+
+#define TIMER_PERIOD_RINGBACK 2000
+#define TIMER_PERIOD_BUSY 700
+#define TIMER_PERIOD_RING 4000
+static int timer_period_ring = TIMER_PERIOD_RING;
+
+#define VPB_EVENTS_ALL (VPB_MRING|VPB_MDIGIT|VPB_MDTMF|VPB_MTONEDETECT|VPB_MTIMEREXP|VPB_MPLAY_UNDERFLOW \
+ |VPB_MRECORD_OVERFLOW|VPB_MSTATION_OFFHOOK|VPB_MSTATION_ONHOOK \
+ |VPB_MRING_OFF|VPB_MDROP|VPB_MSTATION_FLASH)
+#define VPB_EVENTS_NODROP (VPB_MRING|VPB_MDIGIT|VPB_MDTMF|VPB_MTONEDETECT|VPB_MTIMEREXP|VPB_MPLAY_UNDERFLOW \
+ |VPB_MRECORD_OVERFLOW|VPB_MSTATION_OFFHOOK|VPB_MSTATION_ONHOOK \
+ |VPB_MRING_OFF|VPB_MSTATION_FLASH)
+#define VPB_EVENTS_NODTMF (VPB_MRING|VPB_MDIGIT|VPB_MTONEDETECT|VPB_MTIMEREXP|VPB_MPLAY_UNDERFLOW \
+ |VPB_MRECORD_OVERFLOW|VPB_MSTATION_OFFHOOK|VPB_MSTATION_ONHOOK \
+ |VPB_MRING_OFF|VPB_MDROP|VPB_MSTATION_FLASH)
+#define VPB_EVENTS_STAT (VPB_MRING|VPB_MDIGIT|VPB_MDTMF|VPB_MTONEDETECT|VPB_MTIMEREXP|VPB_MPLAY_UNDERFLOW \
+ |VPB_MRECORD_OVERFLOW|VPB_MSTATION_OFFHOOK|VPB_MSTATION_ONHOOK \
+ |VPB_MRING_OFF|VPB_MSTATION_FLASH)
+
+
+/* Dialing parameters for Australia */
+/* #define DIAL_WITH_CALL_PROGRESS */
+VPB_TONE_MAP DialToneMap[] = { { VPB_BUSY_AUST, VPB_CALL_DISCONNECT, 0 },
+ { VPB_DIAL, VPB_CALL_DIALTONE, 0 },
+ { VPB_RINGBACK_308, VPB_CALL_RINGBACK, 0 },
+ { VPB_BUSY_AUST, VPB_CALL_BUSY, 0 },
+ { VPB_GRUNT, VPB_CALL_GRUNT, 0 },
+ { 0, 0, 1 } };
+#define VPB_DIALTONE_WAIT 2000 /* Wait up to 2s for a dialtone */
+#define VPB_RINGWAIT 4000 /* Wait up to 4s for ring tone after dialing */
+#define VPB_CONNECTED_WAIT 4000 /* If no ring tone detected for 4s then consider call connected */
+#define TIMER_PERIOD_NOANSWER 120000 /* Let it ring for 120s before deciding theres noone there */
+
+#define MAX_BRIDGES_V4PCI 2
+#define MAX_BRIDGES_V12PCI 128
+
+/* port states */
+#define VPB_STATE_ONHOOK 0
+#define VPB_STATE_OFFHOOK 1
+#define VPB_STATE_DIALLING 2
+#define VPB_STATE_JOINED 3
+#define VPB_STATE_GETDTMF 4
+#define VPB_STATE_PLAYDIAL 5
+#define VPB_STATE_PLAYBUSY 6
+#define VPB_STATE_PLAYRING 7
+
+#define VPB_GOT_RXHWG 1
+#define VPB_GOT_TXHWG 2
+#define VPB_GOT_RXSWG 4
+#define VPB_GOT_TXSWG 8
+
+typedef struct {
+ int inuse;
+ struct ast_channel *c0, *c1, **rc;
+ struct ast_frame **fo;
+ int flags;
+ ast_mutex_t lock;
+ ast_cond_t cond;
+ int endbridge;
+} vpb_bridge_t;
+
+static vpb_bridge_t * bridges;
+static int max_bridges = MAX_BRIDGES_V4PCI;
+
+AST_MUTEX_DEFINE_STATIC(bridge_lock);
+
+typedef enum {
+ vpb_model_unknown = 0,
+ vpb_model_v4pci,
+ vpb_model_v12pci
+} vpb_model_t;
+
+static struct vpb_pvt {
+
+ ast_mutex_t owner_lock; /* Protect blocks that expect ownership to remain the same */
+ struct ast_channel *owner; /* Channel who owns us, possibly NULL */
+
+ int golock; /* Got owner lock ? */
+
+ int mode; /* fxo/imediate/dialtone*/
+ int handle; /* Handle for vpb interface */
+
+ int state; /* used to keep port state (internal to driver) */
+
+ int group; /* Which group this port belongs to */
+ ast_group_t callgroup; /* Call group */
+ ast_group_t pickupgroup; /* Pickup group */
+
+
+ char dev[256]; /* Device name, eg vpb/1-1 */
+ vpb_model_t vpb_model; /* card model */
+
+ struct ast_frame f, fr; /* Asterisk frame interface */
+ char buf[VPB_MAX_BUF]; /* Static buffer for reading frames */
+
+ int dialtone; /* NOT USED */
+ float txgain, rxgain; /* Hardware gain control */
+ float txswgain, rxswgain; /* Software gain control */
+
+ int wantdtmf; /* Waiting for DTMF. */
+ char context[AST_MAX_EXTENSION]; /* The context for this channel */
+
+ char ext[AST_MAX_EXTENSION]; /* DTMF buffer for the ext[ens] */
+ char language[MAX_LANGUAGE]; /* language being used */
+ char callerid[AST_MAX_EXTENSION]; /* CallerId used for directly connected phone */
+ int callerid_type; /* Caller ID type: 0=>none 1=>vpb 2=>AstV23 3=>AstBell */
+ char cid_num[AST_MAX_EXTENSION];
+ char cid_name[AST_MAX_EXTENSION];
+
+ int dtmf_caller_pos; /* DTMF CallerID detection (Brazil)*/
+
+ int lastoutput; /* Holds the last Audio format output'ed */
+ int lastinput; /* Holds the last Audio format input'ed */
+ int last_ignore_dtmf;
+
+ void *busy_timer; /* Void pointer for busy vpb_timer */
+ int busy_timer_id; /* unique timer ID for busy timer */
+
+ void *ringback_timer; /* Void pointer for ringback vpb_timer */
+ int ringback_timer_id; /* unique timer ID for ringback timer */
+
+ void *ring_timer; /* Void pointer for ring vpb_timer */
+ int ring_timer_id; /* unique timer ID for ring timer */
+
+ void *dtmfidd_timer; /* Void pointer for DTMF IDD vpb_timer */
+ int dtmfidd_timer_id; /* unique timer ID for DTMF IDD timer */
+
+ struct ast_dsp *vad; /* AST Voice Activation Detection dsp */
+
+ struct timeval lastgrunt; /* time stamp of last grunt event */
+
+ ast_mutex_t lock; /* This one just protects bridge ptr below */
+ vpb_bridge_t *bridge;
+
+ int stopreads; /* Stop reading...*/
+ int read_state; /* Read state */
+ int chuck_count; /* a count of packets weve chucked away!*/
+ pthread_t readthread; /* For monitoring read channel. One per owned channel. */
+
+ ast_mutex_t record_lock; /* This one prevents reentering a record_buf block */
+ ast_mutex_t play_lock; /* This one prevents reentering a play_buf block */
+ int play_buf_time; /* How long the last play_buf took */
+ struct timeval lastplay; /* Last play time */
+
+ ast_mutex_t play_dtmf_lock;
+ char play_dtmf[16];
+
+ int faxhandled; /* has a fax tone been handled ? */
+
+ struct vpb_pvt *next; /* Next channel in list */
+
+} *iflist = NULL;
+
+static struct ast_channel *vpb_new(struct vpb_pvt *i, enum ast_channel_state state, char *context);
+static void *do_chanreads(void *pvt);
+static struct ast_channel *vpb_request(const char *type, int format, void *data, int *cause);
+static int vpb_digit_begin(struct ast_channel *ast, char digit);
+static int vpb_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
+static int vpb_call(struct ast_channel *ast, char *dest, int timeout);
+static int vpb_hangup(struct ast_channel *ast);
+static int vpb_answer(struct ast_channel *ast);
+static struct ast_frame *vpb_read(struct ast_channel *ast);
+static int vpb_write(struct ast_channel *ast, struct ast_frame *frame);
+static enum ast_bridge_result ast_vpb_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
+static int vpb_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
+static int vpb_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+
+static struct ast_channel_tech vpb_tech = {
+ type: "vpb",
+ description: tdesc,
+ capabilities: AST_FORMAT_SLINEAR,
+ properties: 0,
+ requester: vpb_request,
+ devicestate: NULL,
+ send_digit_begin: vpb_digit_begin,
+ send_digit_end: vpb_digit_end,
+ call: vpb_call,
+ hangup: vpb_hangup,
+ answer: vpb_answer,
+ read: vpb_read,
+ write: vpb_write,
+ send_text: NULL,
+ send_image: NULL,
+ send_html: NULL,
+ exception: NULL,
+ bridge: ast_vpb_bridge,
+ indicate: vpb_indicate,
+ fixup: vpb_fixup,
+ setoption: NULL,
+ queryoption: NULL,
+ transfer: NULL,
+ write_video: NULL,
+ bridged_channel: NULL
+};
+
+static struct ast_channel_tech vpb_tech_indicate = {
+ type: "vpb",
+ description: tdesc,
+ capabilities: AST_FORMAT_SLINEAR,
+ properties: 0,
+ requester: vpb_request,
+ devicestate: NULL,
+ send_digit_begin: vpb_digit_begin,
+ send_digit_end: vpb_digit_end,
+ call: vpb_call,
+ hangup: vpb_hangup,
+ answer: vpb_answer,
+ read: vpb_read,
+ write: vpb_write,
+ send_text: NULL,
+ send_image: NULL,
+ send_html: NULL,
+ exception: NULL,
+ bridge: ast_vpb_bridge,
+ indicate: NULL,
+ fixup: vpb_fixup,
+ setoption: NULL,
+ queryoption: NULL,
+ transfer: NULL,
+ write_video: NULL,
+ bridged_channel: NULL
+};
+
+/* Can't get ast_vpb_bridge() working on v4pci without either a horrible
+* high pitched feedback noise or bad hiss noise depending on gain settings
+* Get asterisk to do the bridging
+*/
+#define BAD_V4PCI_BRIDGE
+
+/* This one enables a half duplex bridge which may be required to prevent high pitched
+ * feedback when getting asterisk to do the bridging and when using certain gain settings.
+ */
+/* #define HALF_DUPLEX_BRIDGE */
+
+/* This is the Native bridge code, which Asterisk will try before using its own bridging code */
+static enum ast_bridge_result ast_vpb_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
+{
+ struct vpb_pvt *p0 = (struct vpb_pvt *)c0->tech_pvt;
+ struct vpb_pvt *p1 = (struct vpb_pvt *)c1->tech_pvt;
+ int i;
+ int res;
+ struct ast_channel *cs[3];
+ struct ast_channel *who;
+ struct ast_frame *f;
+
+ cs[0] = c0;
+ cs[1] = c1;
+
+ #ifdef BAD_V4PCI_BRIDGE
+ if(p0->vpb_model==vpb_model_v4pci)
+ return AST_BRIDGE_FAILED_NOWARN;
+ #endif
+ if ( UseNativeBridge != 1){
+ return AST_BRIDGE_FAILED_NOWARN;
+ }
+
+/*
+ ast_mutex_lock(&p0->lock);
+ ast_mutex_lock(&p1->lock);
+*/
+
+ /* Bridge channels, check if we can. I believe we always can, so find a slot.*/
+
+ ast_mutex_lock(&bridge_lock); {
+ for (i = 0; i < max_bridges; i++)
+ if (!bridges[i].inuse)
+ break;
+ if (i < max_bridges) {
+ bridges[i].inuse = 1;
+ bridges[i].endbridge = 0;
+ bridges[i].flags = flags;
+ bridges[i].rc = rc;
+ bridges[i].fo = fo;
+ bridges[i].c0 = c0;
+ bridges[i].c1 = c1;
+ }
+ } ast_mutex_unlock(&bridge_lock);
+
+ if (i == max_bridges) {
+ ast_log(LOG_WARNING, "%s: vpb_bridge: Failed to bridge %s and %s!\n", p0->dev, c0->name, c1->name);
+ ast_mutex_unlock(&p0->lock);
+ ast_mutex_unlock(&p1->lock);
+ return AST_BRIDGE_FAILED_NOWARN;
+ } else {
+ /* Set bridge pointers. You don't want to take these locks while holding bridge lock.*/
+ ast_mutex_lock(&p0->lock); {
+ p0->bridge = &bridges[i];
+ } ast_mutex_unlock(&p0->lock);
+
+ ast_mutex_lock(&p1->lock); {
+ p1->bridge = &bridges[i];
+ } ast_mutex_unlock(&p1->lock);
+
+ ast_verb(2, "%s: vpb_bridge: Bridging call entered with [%s, %s]\n",p0->dev, c0->name, c1->name);
+ }
+
+ ast_verb(3, "Native bridging %s and %s\n", c0->name, c1->name);
+
+ #ifdef HALF_DUPLEX_BRIDGE
+
+ ast_verb(2, "%s: vpb_bridge: Starting half-duplex bridge [%s, %s]\n",p0->dev, c0->name, c1->name);
+
+ int dir = 0;
+
+ memset(p0->buf, 0, sizeof p0->buf);
+ memset(p1->buf, 0, sizeof p1->buf);
+
+ vpb_record_buf_start(p0->handle, VPB_ALAW);
+ vpb_record_buf_start(p1->handle, VPB_ALAW);
+
+ vpb_play_buf_start(p0->handle, VPB_ALAW);
+ vpb_play_buf_start(p1->handle, VPB_ALAW);
+
+ while( !bridges[i].endbridge ) {
+ struct vpb_pvt *from, *to;
+ if(++dir%2) {
+ from = p0;
+ to = p1;
+ } else {
+ from = p1;
+ to = p0;
+ }
+ vpb_record_buf_sync(from->handle, from->buf, VPB_SAMPLES);
+ vpb_play_buf_sync(to->handle, from->buf, VPB_SAMPLES);
+ }
+
+ vpb_record_buf_finish(p0->handle);
+ vpb_record_buf_finish(p1->handle);
+
+ vpb_play_buf_finish(p0->handle);
+ vpb_play_buf_finish(p1->handle);
+
+ ast_verb(2, "%s: vpb_bridge: Finished half-duplex bridge [%s, %s]\n",p0->dev, c0->name, c1->name);
+
+ res = VPB_OK;
+
+ #else
+
+ res = vpb_bridge(p0->handle, p1->handle, VPB_BRIDGE_ON, i+1 /* resource 1 & 2 only for V4PCI*/ );
+ if (res == VPB_OK) {
+ /* pthread_cond_wait(&bridges[i].cond, &bridges[i].lock);*/ /* Wait for condition signal. */
+ while( !bridges[i].endbridge ) {
+ /* Are we really ment to be doing nothing ?!?! */
+ who = ast_waitfor_n(cs, 2, &timeoutms);
+ if (!who) {
+ if (!timeoutms) {
+ res = AST_BRIDGE_RETRY;
+ break;
+ }
+ ast_debug(1, "%s: vpb_bridge: Empty frame read...\n",p0->dev);
+ /* check for hangup / whentohangup */
+ if (ast_check_hangup(c0) || ast_check_hangup(c1))
+ break;
+ continue;
+ }
+ f = ast_read(who);
+ if (!f || ((f->frametype == AST_FRAME_DTMF) &&
+ (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
+ ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
+ *fo = f;
+ *rc = who;
+ ast_debug(1, "%s: vpb_bridge: Got a [%s]\n",p0->dev, f ? "digit" : "hangup");
+/*
+ if ((c0->tech_pvt == pvt0) && (!ast_check_hangup(c0))) {
+ if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
+ ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
+ }
+ if ((c1->tech_pvt == pvt1) && (!ast_check_hangup(c1))) {
+ if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
+ ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
+ }
+*/
+ /* That's all we needed */
+ /*return 0; */
+ /* Check if we need to break */
+ if (break_for_dtmf){
+ break;
+ }
+ else if ((f->frametype == AST_FRAME_DTMF) && ((f->subclass == '#')||(f->subclass == '*'))){
+ break;
+ }
+ } else {
+ if ((f->frametype == AST_FRAME_DTMF) ||
+ (f->frametype == AST_FRAME_VOICE) ||
+ (f->frametype == AST_FRAME_VIDEO))
+ {
+ /* Forward voice or DTMF frames if they happen upon us */
+ /* Actually I dont think we want to forward on any frames!
+ if (who == c0) {
+ ast_write(c1, f);
+ } else if (who == c1) {
+ ast_write(c0, f);
+ }
+ */
+ }
+ ast_frfree(f);
+ }
+ /* Swap priority not that it's a big deal at this point */
+ cs[2] = cs[0];
+ cs[0] = cs[1];
+ cs[1] = cs[2];
+ };
+ vpb_bridge(p0->handle, p1->handle, VPB_BRIDGE_OFF, i+1 /* resource 1 & 2 only for V4PCI*/ );
+ }
+
+ #endif
+
+ ast_mutex_lock(&bridge_lock); {
+ bridges[i].inuse = 0;
+ } ast_mutex_unlock(&bridge_lock);
+
+ p0->bridge = NULL;
+ p1->bridge = NULL;
+
+
+ ast_verb(2, "Bridging call done with [%s, %s] => %d\n", c0->name, c1->name, res);
+
+/*
+ ast_mutex_unlock(&p0->lock);
+ ast_mutex_unlock(&p1->lock);
+*/
+ return (res==VPB_OK) ? AST_BRIDGE_COMPLETE : AST_BRIDGE_FAILED;
+}
+
+/* Caller ID can be located in different positions between the rings depending on your Telco
+ * Australian (Telstra) callerid starts 700ms after 1st ring and finishes 1.5s after first ring
+ * Use ANALYSE_CID to record rings and determine location of callerid
+ */
+/* #define ANALYSE_CID */
+#define RING_SKIP 300
+#define CID_MSECS 2000
+
+static void get_callerid(struct vpb_pvt *p)
+{
+ short buf[CID_MSECS*8]; /* 8kHz sampling rate */
+ struct timeval cid_record_time;
+ int rc;
+ struct ast_channel *owner = p->owner;
+/*
+ char callerid[AST_MAX_EXTENSION] = "";
+*/
+#ifdef ANALYSE_CID
+ void * ws;
+ char * file="cidsams.wav";
+#endif
+
+
+ if( ast_mutex_trylock(&p->record_lock) == 0 ) {
+
+ cid_record_time = ast_tvnow();
+ ast_verb(4, "CID record - start\n");
+
+ /* Skip any trailing ringtone */
+ if (UsePolarityCID != 1){
+ vpb_sleep(RING_SKIP);
+ }
+
+ ast_verb(4, "CID record - skipped %dms trailing ring\n",
+ ast_tvdiff_ms(ast_tvnow(), cid_record_time));
+ cid_record_time = ast_tvnow();
+
+ /* Record bit between the rings which contains the callerid */
+ vpb_record_buf_start(p->handle, VPB_LINEAR);
+ rc = vpb_record_buf_sync(p->handle, (char*)buf, sizeof(buf));
+ vpb_record_buf_finish(p->handle);
+#ifdef ANALYSE_CID
+ vpb_wave_open_write(&ws, file, VPB_LINEAR);
+ vpb_wave_write(ws,(char*)buf,sizeof(buf));
+ vpb_wave_close_write(ws);
+#endif
+
+ ast_verb(4, "CID record - recorded %dms between rings\n",
+ ast_tvdiff_ms(ast_tvnow(), cid_record_time));
+
+ ast_mutex_unlock(&p->record_lock);
+
+ if( rc != VPB_OK ) {
+ ast_log(LOG_ERROR, "Failed to record caller id sample on %s\n", p->dev );
+ return;
+ }
+
+ VPB_CID *cli_struct = new VPB_CID;
+ cli_struct->ra_cldn[0]=0;
+ cli_struct->ra_cn[0]=0;
+ /* This decodes FSK 1200baud type callerid */
+ if ((rc=vpb_cid_decode2(cli_struct, buf, CID_MSECS*8)) == VPB_OK ) {
+ /*
+ if (owner->cid.cid_num)
+ ast_free(owner->cid.cid_num);
+ owner->cid.cid_num=NULL;
+ if (owner->cid.cid_name)
+ ast_free(owner->cid.cid_name);
+ owner->cid.cid_name=NULL;
+ */
+
+ if (cli_struct->ra_cldn[0]=='\0'){
+ /*
+ owner->cid.cid_num = ast_strdup(cli_struct->cldn);
+ owner->cid.cid_name = ast_strdup(cli_struct->cn);
+ */
+ if (owner){
+ ast_set_callerid(owner, cli_struct->cldn, cli_struct->cn, cli_struct->cldn);
+ } else {
+ strcpy(p->cid_num, cli_struct->cldn);
+ strcpy(p->cid_name, cli_struct->cn);
+
+ }
+ ast_verb(4, "CID record - got [%s] [%s]\n",owner->cid.cid_num,owner->cid.cid_name );
+ snprintf(p->callerid,sizeof(p->callerid)-1,"%s %s",cli_struct->cldn,cli_struct->cn);
+ }
+ else {
+ ast_log(LOG_ERROR,"CID record - No caller id avalable on %s \n", p->dev);
+ }
+
+ } else {
+ ast_log(LOG_ERROR, "CID record - Failed to decode caller id on %s - %s\n", p->dev, vpb_strerror(rc) );
+ strncpy(p->callerid,"unknown", sizeof(p->callerid) - 1);
+ }
+ delete cli_struct;
+
+ } else
+ ast_log(LOG_ERROR, "CID record - Failed to set record mode for caller id on %s\n", p->dev );
+}
+
+static void get_callerid_ast(struct vpb_pvt *p)
+{
+ struct callerid_state *cs;
+ char buf[1024];
+ char *name=NULL, *number=NULL;
+ int flags;
+ int rc=0,vrc;
+ int sam_count=0;
+ struct ast_channel *owner = p->owner;
+ int which_cid;
+/*
+ float old_gain;
+*/
+#ifdef ANALYSE_CID
+ void * ws;
+ char * file="cidsams.wav";
+#endif
+
+ if(p->callerid_type == 1) {
+ ast_verb(4, "Collected caller ID already\n");
+ return;
+ }
+ else if(p->callerid_type == 2 ) {
+ which_cid=CID_SIG_V23;
+ ast_verb(4, "Collecting Caller ID v23...\n");
+ }
+ else if(p->callerid_type == 3) {
+ which_cid=CID_SIG_BELL;
+ ast_verb(4, "Collecting Caller ID bell...\n");
+ }
+ else {
+ ast_verb(4, "Caller ID disabled\n");
+ return;
+ }
+/* vpb_sleep(RING_SKIP); */
+/* vpb_record_get_gain(p->handle, &old_gain); */
+ cs = callerid_new(which_cid);
+ if (cs){
+#ifdef ANALYSE_CID
+ vpb_wave_open_write(&ws, file, VPB_MULAW);
+ vpb_record_set_gain(p->handle, 3.0);
+ vpb_record_set_hw_gain(p->handle,12.0);
+#endif
+ vpb_record_buf_start(p->handle, VPB_MULAW);
+ while((rc == 0)&&(sam_count<8000*3)){
+ vrc = vpb_record_buf_sync(p->handle, (char*)buf, sizeof(buf));
+ if (vrc != VPB_OK)
+ ast_log(LOG_ERROR, "%s: Caller ID couldnt read audio buffer!\n",p->dev);
+ rc = callerid_feed(cs,(unsigned char *)buf,sizeof(buf),AST_FORMAT_ULAW);
+#ifdef ANALYSE_CID
+ vpb_wave_write(ws,(char*)buf,sizeof(buf));
+#endif
+ sam_count+=sizeof(buf);
+ ast_verb(4, "Collecting Caller ID samples [%d][%d]...\n",sam_count,rc);
+ }
+ vpb_record_buf_finish(p->handle);
+#ifdef ANALYSE_CID
+ vpb_wave_close_write(ws);
+#endif
+ if (rc == 1){
+ callerid_get(cs, &name, &number, &flags);
+ ast_verb(1, "%s: Caller ID name [%s] number [%s] flags [%d]\n",p->dev,name, number,flags);
+ }
+ else {
+ ast_log(LOG_ERROR, "%s: Failed to decode Caller ID \n", p->dev );
+ }
+/* vpb_record_set_gain(p->handle, old_gain); */
+/* vpb_record_set_hw_gain(p->handle,6.0); */
+ }
+ else {
+ ast_log(LOG_ERROR, "%s: Failed to create Caller ID struct\n", p->dev );
+ }
+ if (owner->cid.cid_num) {
+ ast_free(owner->cid.cid_num);
+ owner->cid.cid_num = NULL;
+ }
+ if (owner->cid.cid_name) {
+ ast_free(owner->cid.cid_name);
+ owner->cid.cid_name = NULL;
+ }
+ if (number)
+ ast_shrink_phone_number(number);
+ ast_set_callerid(owner,
+ number, name,
+ owner->cid.cid_ani ? NULL : number);
+ if (!ast_strlen_zero(name)){
+ snprintf(p->callerid,(sizeof(p->callerid)-1),"%s %s",number,name);
+ } else {
+ snprintf(p->callerid,(sizeof(p->callerid)-1),"%s",number);
+ }
+ if (cs)
+ callerid_free(cs);
+}
+
+/* Terminate any tones we are presently playing */
+static void stoptone( int handle)
+{
+ int ret;
+ VPB_EVENT je;
+ while(vpb_playtone_state(handle)!=VPB_OK){
+ vpb_tone_terminate(handle);
+ ret = vpb_get_event_ch_async(handle,&je);
+ if ((ret == VPB_OK)&&(je.type != VPB_DIALEND)){
+ ast_verb(4, "Stop tone collected a wrong event!![%d]\n",je.type);
+/* vpb_put_event(&je); */
+ }
+ vpb_sleep(10);
+ }
+}
+
+/* Safe vpb_playtone_async */
+static int playtone( int handle, VPB_TONE *tone)
+{
+ int ret=VPB_OK;
+ stoptone(handle);
+ ast_verb(4, "[%02d]: Playing tone\n", handle);
+ ret = vpb_playtone_async(handle, tone);
+ return ret;
+}
+
+static inline int monitor_handle_owned(struct vpb_pvt *p, VPB_EVENT *e)
+{
+ struct ast_frame f = {AST_FRAME_CONTROL}; /* default is control, Clear rest. */
+ int endbridge = 0;
+ int res=0;
+
+ ast_verb(4, "%s: handle_owned: got event: [%d=>%d]\n", p->dev, e->type, e->data);
+
+ f.src = "vpb";
+ switch (e->type) {
+ case VPB_RING:
+ if (p->mode == MODE_FXO) {
+ f.subclass = AST_CONTROL_RING;
+ vpb_timer_stop(p->ring_timer);
+ vpb_timer_start(p->ring_timer);
+ } else
+ f.frametype = AST_FRAME_NULL; /* ignore ring on station port. */
+ break;
+
+ case VPB_RING_OFF:
+ f.frametype = AST_FRAME_NULL;
+ break;
+
+ case VPB_TIMEREXP:
+ if (e->data == p->busy_timer_id) {
+ playtone(p->handle,&Busytone);
+ p->state = VPB_STATE_PLAYBUSY;
+ vpb_timer_stop(p->busy_timer);
+ vpb_timer_start(p->busy_timer);
+ f.frametype = AST_FRAME_NULL;
+ } else if (e->data == p->ringback_timer_id) {
+ playtone(p->handle, &Ringbacktone);
+ vpb_timer_stop(p->ringback_timer);
+ vpb_timer_start(p->ringback_timer);
+ f.frametype = AST_FRAME_NULL;
+ } else if (e->data == p->ring_timer_id) {
+ /* We didnt get another ring in time! */
+ if (p->owner->_state != AST_STATE_UP) {
+ /* Assume caller has hung up */
+ vpb_timer_stop(p->ring_timer);
+ f.subclass = AST_CONTROL_HANGUP;
+ } else {
+ vpb_timer_stop(p->ring_timer);
+ f.frametype = AST_FRAME_NULL;
+ }
+
+ } else {
+ f.frametype = AST_FRAME_NULL; /* Ignore. */
+ }
+ break;
+
+ case VPB_DTMF_DOWN:
+ case VPB_DTMF:
+ if (use_ast_dtmfdet){
+ f.frametype = AST_FRAME_NULL;
+ } else if (p->owner->_state == AST_STATE_UP) {
+ f.frametype = AST_FRAME_DTMF;
+ f.subclass = e->data;
+ } else
+ f.frametype = AST_FRAME_NULL;
+ break;
+
+ case VPB_TONEDETECT:
+ if (e->data == VPB_BUSY || e->data == VPB_BUSY_308 || e->data == VPB_BUSY_AUST ) {
+ ast_verb(4, "%s: handle_owned: got event: BUSY\n", p->dev);
+ if (p->owner->_state == AST_STATE_UP) {
+ f.subclass = AST_CONTROL_HANGUP;
+ }
+ else {
+ f.subclass = AST_CONTROL_BUSY;
+ }
+ }
+ else if (e->data == VPB_FAX){
+ if (!p->faxhandled){
+ if (strcmp(p->owner->exten, "fax")) {
+ const char *target_context = S_OR(p->owner->macrocontext, p->owner->context);
+
+ if (ast_exists_extension(p->owner, target_context, "fax", 1, p->owner->cid.cid_num)) {
+ ast_verb(3, "Redirecting %s to fax extension\n", p->owner->name);
+ /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
+ pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", p->owner->exten);
+ if (ast_async_goto(p->owner, target_context, "fax", 1))
+ ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", p->owner->name, target_context);
+ } else
+ ast_log(LOG_NOTICE, "Fax detected, but no fax extension\n");
+ } else
+ ast_debug(1, "Already in a fax extension, not redirecting\n");
+ } else
+ ast_debug(1, "Fax already handled\n");
+
+ }
+ else if (e->data == VPB_GRUNT) {
+ if ( ast_tvdiff_ms(ast_tvnow(), p->lastgrunt) > gruntdetect_timeout ) {
+ /* Nothing heard on line for a very long time
+ * Timeout connection */
+ ast_verb(3, "grunt timeout\n");
+ ast_log(LOG_NOTICE,"%s: Line hangup due of lack of conversation\n",p->dev);
+ f.subclass = AST_CONTROL_HANGUP;
+ } else {
+ p->lastgrunt = ast_tvnow();
+ f.frametype = AST_FRAME_NULL;
+ }
+ }
+ else {
+ f.frametype = AST_FRAME_NULL;
+ }
+ break;
+
+ case VPB_CALLEND:
+ #ifdef DIAL_WITH_CALL_PROGRESS
+ if (e->data == VPB_CALL_CONNECTED)
+ f.subclass = AST_CONTROL_ANSWER;
+ else if (e->data == VPB_CALL_NO_DIAL_TONE || e->data == VPB_CALL_NO_RING_BACK)
+ f.subclass = AST_CONTROL_CONGESTION;
+ else if (e->data == VPB_CALL_NO_ANSWER || e->data == VPB_CALL_BUSY)
+ f.subclass = AST_CONTROL_BUSY;
+ else if (e->data == VPB_CALL_DISCONNECTED)
+ f.subclass = AST_CONTROL_HANGUP;
+ #else
+ ast_log(LOG_NOTICE,"%s: Got call progress callback but blind dialing \n", p->dev);
+ f.frametype = AST_FRAME_NULL;
+ #endif
+ break;
+
+ case VPB_STATION_OFFHOOK:
+ f.subclass = AST_CONTROL_ANSWER;
+ break;
+
+ case VPB_DROP:
+ if ((p->mode == MODE_FXO)&&(UseLoopDrop)){ /* ignore loop drop on stations */
+ if (p->owner->_state == AST_STATE_UP)
+ f.subclass = AST_CONTROL_HANGUP;
+ else
+ f.frametype = AST_FRAME_NULL;
+ }
+ break;
+ case VPB_LOOP_ONHOOK:
+ if (p->owner->_state == AST_STATE_UP)
+ f.subclass = AST_CONTROL_HANGUP;
+ else
+ f.frametype = AST_FRAME_NULL;
+ break;
+ case VPB_STATION_ONHOOK:
+ f.subclass = AST_CONTROL_HANGUP;
+ break;
+
+ case VPB_STATION_FLASH:
+ f.subclass = AST_CONTROL_FLASH;
+ break;
+
+ /* Called when dialing has finished and ringing starts
+ * No indication that call has really been answered when using blind dialing
+ */
+ case VPB_DIALEND:
+ if (p->state < 5){
+ f.subclass = AST_CONTROL_ANSWER;
+ ast_verb(2, "%s: Dialend\n", p->dev);
+ } else {
+ f.frametype = AST_FRAME_NULL;
+ }
+ break;
+
+ case VPB_PLAY_UNDERFLOW:
+ f.frametype = AST_FRAME_NULL;
+ vpb_reset_play_fifo_alarm(p->handle);
+ break;
+
+ case VPB_RECORD_OVERFLOW:
+ f.frametype = AST_FRAME_NULL;
+ vpb_reset_record_fifo_alarm(p->handle);
+ break;
+
+ default:
+ f.frametype = AST_FRAME_NULL;
+ break;
+ }
+
+/*
+ ast_verb(4, "%s: LOCKING in handle_owned [%d]\n", p->dev,res);
+ res = ast_mutex_lock(&p->lock);
+ ast_verb(4, "%s: LOCKING count[%d] owner[%d] \n", p->dev, p->lock.__m_count,p->lock.__m_owner);
+*/
+ {
+ if (p->bridge) { /* Check what happened, see if we need to report it. */
+ switch (f.frametype) {
+ case AST_FRAME_DTMF:
+ if ( !(p->bridge->c0 == p->owner &&
+ (p->bridge->flags & AST_BRIDGE_DTMF_CHANNEL_0) ) &&
+ !(p->bridge->c1 == p->owner &&
+ (p->bridge->flags & AST_BRIDGE_DTMF_CHANNEL_1) ))
+ /* Kill bridge, this is interesting. */
+ endbridge = 1;
+ break;
+
+ case AST_FRAME_CONTROL:
+ if (!(p->bridge->flags & AST_BRIDGE_IGNORE_SIGS))
+ #if 0
+ if (f.subclass == AST_CONTROL_BUSY ||
+ f.subclass == AST_CONTROL_CONGESTION ||
+ f.subclass == AST_CONTROL_HANGUP ||
+ f.subclass == AST_CONTROL_FLASH)
+ #endif
+ endbridge = 1;
+ break;
+
+ default:
+ break;
+ }
+ if (endbridge) {
+ if (p->bridge->fo)
+ *p->bridge->fo = ast_frisolate(&f);
+ if (p->bridge->rc)
+ *p->bridge->rc = p->owner;
+
+ ast_mutex_lock(&p->bridge->lock); {
+ p->bridge->endbridge = 1;
+ ast_cond_signal(&p->bridge->cond);
+ } ast_mutex_unlock(&p->bridge->lock);
+ }
+ }
+ }
+
+ if (endbridge){
+ res = ast_mutex_unlock(&p->lock);
+/*
+ ast_verb(4, "%s: unLOCKING in handle_owned [%d]\n", p->dev,res);
+*/
+ return 0;
+ }
+
+ ast_verb(4, "%s: handle_owned: Prepared frame type[%d]subclass[%d], bridge=%p owner=[%s]\n",
+ p->dev, f.frametype, f.subclass, (void *)p->bridge, p->owner->name);
+
+ /* Trylock used here to avoid deadlock that can occur if we
+ * happen to be in here handling an event when hangup is called
+ * Problem is that hangup holds p->owner->lock
+ */
+ if ((f.frametype >= 0)&& (f.frametype != AST_FRAME_NULL)&&(p->owner)) {
+ if (ast_mutex_trylock(&p->owner->lock)==0) {
+ ast_queue_frame(p->owner, &f);
+ ast_mutex_unlock(&p->owner->lock);
+ ast_verb(4, "%s: handled_owned: Queued Frame to [%s]\n", p->dev,p->owner->name);
+ } else {
+ ast_verbose("%s: handled_owned: Missed event %d/%d \n",
+ p->dev,f.frametype, f.subclass);
+ }
+ }
+ res = ast_mutex_unlock(&p->lock);
+/*
+ ast_verb(4, "%s: unLOCKING in handle_owned [%d]\n", p->dev,res);
+*/
+
+ return 0;
+}
+
+static inline int monitor_handle_notowned(struct vpb_pvt *p, VPB_EVENT *e)
+{
+ char s[2] = {0};
+ struct ast_channel *owner = p->owner;
+ char cid_num[256];
+ char cid_name[256];
+/*
+ struct ast_channel *c;
+*/
+
+ char str[VPB_MAX_STR];
+ vpb_translate_event(e, str);
+ ast_verb(4, "%s: handle_notowned: mode=%d, event[%d][%s]=[%d]\n", p->dev, p->mode, e->type,str, e->data);
+
+ switch(e->type) {
+ case VPB_LOOP_ONHOOK:
+ case VPB_LOOP_POLARITY:
+ if (UsePolarityCID == 1){
+ ast_verb(4, "Polarity reversal\n");
+ if(p->callerid_type == 1) {
+ ast_verb(4, "Using VPB Caller ID\n");
+ get_callerid(p); /* UK CID before 1st ring*/
+ }
+/* get_callerid_ast(p); */ /* Caller ID using the ast functions */
+ }
+ break;
+ case VPB_RING:
+ if (p->mode == MODE_FXO) /* FXO port ring, start * */ {
+ vpb_new(p, AST_STATE_RING, p->context);
+ if (UsePolarityCID != 1){
+ if(p->callerid_type == 1) {
+ ast_verb(4, "Using VPB Caller ID\n");
+ get_callerid(p); /* Australian CID only between 1st and 2nd ring */
+ }
+ get_callerid_ast(p); /* Caller ID using the ast functions */
+ }
+ else {
+ ast_log(LOG_ERROR, "Setting caller ID: %s %s\n",p->cid_num, p->cid_name);
+ ast_set_callerid(p->owner, p->cid_num, p->cid_name, p->cid_num);
+ p->cid_num[0]=0;
+ p->cid_name[0]=0;
+ }
+
+ vpb_timer_stop(p->ring_timer);
+ vpb_timer_start(p->ring_timer);
+ }
+ break;
+
+ case VPB_RING_OFF:
+ break;
+
+ case VPB_STATION_OFFHOOK:
+ if (p->mode == MODE_IMMEDIATE)
+ vpb_new(p,AST_STATE_RING, p->context);
+ else {
+ ast_verb(4, "%s: handle_notowned: playing dialtone\n",p->dev);
+ playtone(p->handle, &Dialtone);
+ p->state=VPB_STATE_PLAYDIAL;
+ p->wantdtmf = 1;
+ p->ext[0] = 0; /* Just to be sure & paranoid.*/
+ }
+ break;
+
+ case VPB_DIALEND:
+ if (p->mode == MODE_DIALTONE){
+ if (p->state == VPB_STATE_PLAYDIAL) {
+ playtone(p->handle, &Dialtone);
+ p->wantdtmf = 1;
+ p->ext[0] = 0; /* Just to be sure & paranoid. */
+ }
+ /* These are not needed as they have timers to restart them
+ else if (p->state == VPB_STATE_PLAYBUSY) {
+ playtone(p->handle, &Busytone);
+ p->wantdtmf = 1;
+ p->ext[0] = 0;
+ }
+ else if (p->state == VPB_STATE_PLAYRING) {
+ playtone(p->handle, &Ringbacktone);
+ p->wantdtmf = 1;
+ p->ext[0] = 0;
+ }
+ */
+ } else {
+ ast_verb(4, "%s: handle_notowned: Got a DIALEND when not really expected\n",p->dev);
+ }
+ break;
+
+ case VPB_STATION_ONHOOK: /* clear ext */
+ stoptone(p->handle);
+ p->wantdtmf = 1 ;
+ p->ext[0] = 0;
+ p->state=VPB_STATE_ONHOOK;
+ break;
+ case VPB_TIMEREXP:
+ if (e->data == p->dtmfidd_timer_id) {
+ if (ast_exists_extension(NULL, p->context, p->ext, 1, p->callerid)){
+ ast_verb(4, "%s: handle_notowned: DTMF IDD timer out, matching on [%s] in [%s]\n", p->dev,p->ext , p->context);
+
+ vpb_new(p,AST_STATE_RING, p->context);
+ }
+ } else if (e->data == p->ring_timer_id) {
+ /* We didnt get another ring in time! */
+ if (p->owner){
+ if (p->owner->_state != AST_STATE_UP) {
+ /* Assume caller has hung up */
+ vpb_timer_stop(p->ring_timer);
+ }
+ } else {
+ /* No owner any more, Assume caller has hung up */
+ vpb_timer_stop(p->ring_timer);
+ }
+ }
+ break;
+
+ case VPB_DTMF:
+ if (p->state == VPB_STATE_ONHOOK){
+ /* DTMF's being passed while on-hook maybe Caller ID */
+ if ( p->mode == MODE_FXO ) {
+ if ( e->data == DTMF_CID_START ) { /* CallerID Start signal */
+ p->dtmf_caller_pos = 0; /* Leaves the first digit out */
+ memset(p->callerid,0,AST_MAX_EXTENSION);
+ }
+ else if ( e->data == DTMF_CID_STOP ) { /* CallerID End signal */
+ p->callerid[p->dtmf_caller_pos] = '\0';
+ ast_verb(3, " %s: DTMF CallerID %s\n",p->dev,p->callerid);
+ if (owner){
+ /*
+ if (owner->cid.cid_num)
+ ast_free(owner->cid.cid_num);
+ owner->cid.cid_num=NULL;
+ if (owner->cid.cid_name)
+ ast_free(owner->cid.cid_name);
+ owner->cid.cid_name=NULL;
+ owner->cid.cid_num = strdup(p->callerid);
+ */
+ cid_name[0] = '\0';
+ cid_num[0] = '\0';
+ ast_callerid_split(p->callerid, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
+ ast_set_callerid(owner, cid_num, cid_name, cid_num);
+
+ } else
+ ast_verb(3, " %s: DTMF CallerID: no owner to assign CID \n",p->dev);
+ } else if ( p->dtmf_caller_pos < AST_MAX_EXTENSION ) {
+ if ( p->dtmf_caller_pos >= 0 )
+ p->callerid[p->dtmf_caller_pos] = e->data;
+ p->dtmf_caller_pos++;
+ }
+ }
+ break;
+ }
+ if (p->wantdtmf == 1) {
+ stoptone(p->handle);
+ p->wantdtmf = 0;
+ }
+ p->state=VPB_STATE_GETDTMF;
+ s[0] = e->data;
+ strncat(p->ext, s, sizeof(p->ext) - strlen(p->ext) - 1);
+ #if 0
+ if (!strcmp(p->ext,ast_pickup_ext())) {
+ /* Call pickup has been dialled! */
+ if (ast_pickup_call(c)) {
+ /* Call pickup wasnt possible */
+ }
+ }
+ else
+ #endif
+ if (ast_exists_extension(NULL, p->context, p->ext, 1, p->callerid)){
+ if ( ast_canmatch_extension(NULL, p->context, p->ext, 1, p->callerid)){
+ ast_verb(4, "%s: handle_notowned: Multiple matches on [%s] in [%s]\n", p->dev,p->ext , p->context);
+ /* Start DTMF IDD timer */
+ vpb_timer_stop(p->dtmfidd_timer);
+ vpb_timer_start(p->dtmfidd_timer);
+ }
+ else {
+ ast_verb(4, "%s: handle_notowned: Matched on [%s] in [%s]\n", p->dev,p->ext , p->context);
+ vpb_new(p,AST_STATE_UP, p->context);
+ }
[... 1750 lines stripped ...]
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