[svn-commits] rmudgett: trunk r296168 - in /trunk: ./ channels/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Nov 24 16:52:12 CST 2010
Author: rmudgett
Date: Wed Nov 24 16:52:07 2010
New Revision: 296168
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=296168
Log:
Merged revisions 296167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r296167 | rmudgett | 2010-11-24 16:49:48 -0600 (Wed, 24 Nov 2010) | 57 lines
Merged revisions 296166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines
Merged revisions 296165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
The FXS connected phone has to have CW/CID support to fail, as it will
send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal
phone with no CID never fails. Also the SIP phone does not hear MOH when
the CW call is answered.
The DTMF end frame is suppressed when the phone acknowledges the CW signal
for CID. The problem is the DTMF begin frame needs to be suppressed as
well. The DTMF begin frame is causing SIP to start sending the DTMF RTP
frames. Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets.
* Suppress the DTMF begin and end frames when the channel driver is
looking for DTMF digits.
* Fixed a couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill.
* Fixed not sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is possible.)
* Suppress received audio when sending CW/CID spills. The other parties
involved do not need to hear the CW/CID spills and may be confused if the
CW call is for them.
(closes issue #18129)
Reported by: alecdavis
Patches:
issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
NOTE:
* v1.4 does not have the main problem fixed by suppressing the DTMF start
frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog ports, you
need to disable CW/CID either by configuring chan_dahdi.conf
callwaitingcallerid=no or dialing *70 before dialing the number to
temporarily disable CW.
........
................
................
Modified:
trunk/ (props changed)
trunk/channels/chan_dahdi.c
trunk/channels/sig_analog.c
trunk/channels/sig_analog.h
trunk/channels/sig_pri.h
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: trunk/channels/chan_dahdi.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_dahdi.c?view=diff&rev=296168&r1=296167&r2=296168
==============================================================================
--- trunk/channels/chan_dahdi.c (original)
+++ trunk/channels/chan_dahdi.c Wed Nov 24 16:52:07 2010
@@ -492,11 +492,12 @@
#define MASK_AVAIL (1 << 0) /*!< Channel available for PRI use */
#define MASK_INUSE (1 << 1) /*!< Channel currently in use */
-#define CALLWAITING_SILENT_SAMPLES ( (300 * 8) / READ_SIZE) /*!< 300 ms */
-#define CALLWAITING_REPEAT_SAMPLES ( (10000 * 8) / READ_SIZE) /*!< 10,000 ms */
-#define CIDCW_EXPIRE_SAMPLES ( (500 * 8) / READ_SIZE) /*!< 500 ms */
-#define MIN_MS_SINCE_FLASH ( (2000) ) /*!< 2000 ms */
-#define DEFAULT_RINGT ( (8000 * 8) / READ_SIZE) /*!< 8,000 ms */
+#define CALLWAITING_SILENT_SAMPLES ((300 * 8) / READ_SIZE) /*!< 300 ms */
+#define CALLWAITING_REPEAT_SAMPLES ((10000 * 8) / READ_SIZE) /*!< 10,000 ms */
+#define CALLWAITING_SUPPRESS_SAMPLES ((100 * 8) / READ_SIZE) /*!< 100 ms */
+#define CIDCW_EXPIRE_SAMPLES ((500 * 8) / READ_SIZE) /*!< 500 ms */
+#define MIN_MS_SINCE_FLASH ((2000) ) /*!< 2000 ms */
+#define DEFAULT_RINGT ((8000 * 8) / READ_SIZE) /*!< 8,000 ms */
struct dahdi_pvt;
@@ -1053,7 +1054,8 @@
struct timeval dtmfcid_delay; /*!< Time value used for allow line to settle */
int callingpres; /*!< The value of calling presentation that we're going to use when placing a PRI call */
int callwaitingrepeat; /*!< How many samples to wait before repeating call waiting */
- int cidcwexpire; /*!< When to expire our muting for CID/CW */
+ int cidcwexpire; /*!< When to stop waiting for CID/CW CAS response (In samples) */
+ int cid_suppress_expire; /*!< How many samples to suppress after a CID spill. */
/*! \brief Analog caller ID waveform sample buffer */
unsigned char *cidspill;
/*! \brief Position in the cidspill buffer to send out next. */
@@ -1074,7 +1076,12 @@
* characters are processed.
*/
int stripmsd;
- /*! \brief BOOLEAN. XXX Meaning what?? */
+ /*!
+ * \brief TRUE if Call Waiting (CW) CPE Alert Signal (CAS) is being sent.
+ * \note
+ * After CAS is sent, the call waiting caller id will be sent if the phone
+ * gives a positive reply.
+ */
int callwaitcas;
/*! \brief Number of call waiting rings. */
int callwaitrings;
@@ -1839,18 +1846,19 @@
return 0;
}
-static int send_callerid(struct dahdi_pvt *p);
-
static int my_stop_callwait(void *pvt)
{
struct dahdi_pvt *p = pvt;
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
return 0;
}
+static int send_callerid(struct dahdi_pvt *p);
static int save_conference(struct dahdi_pvt *p);
+static int restore_conference(struct dahdi_pvt *p);
static int my_callwait(void *pvt)
{
@@ -1860,6 +1868,11 @@
ast_log(LOG_WARNING, "Spill already exists?!?\n");
ast_free(p->cidspill);
}
+
+ /*
+ * SAS: Subscriber Alert Signal, 440Hz for 300ms
+ * CAS: CPE Alert Signal, 2130Hz * 2750Hz sine waves
+ */
if (!(p->cidspill = ast_malloc(2400 /* SAS */ + 680 /* CAS */ + READ_SIZE * 4)))
return -1;
save_conference(p);
@@ -1898,6 +1911,8 @@
caller->id.number.str,
AST_LAW(p));
} else {
+ ast_verb(3, "CPE supports Call Waiting Caller*ID. Sending '%s/%s'\n",
+ caller->id.name.str, caller->id.number.str);
p->callwaitcas = 0;
p->cidcwexpire = 0;
p->cidlen = ast_callerid_callwaiting_generate(p->cidspill,
@@ -1907,6 +1922,7 @@
p->cidlen += READ_SIZE * 4;
}
p->cidpos = 0;
+ p->cid_suppress_expire = 0;
send_callerid(p);
}
return 0;
@@ -1978,68 +1994,72 @@
static inline int dahdi_confmute(struct dahdi_pvt *p, int muted);
-static void my_handle_dtmfup(void *pvt, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest)
+static void my_handle_dtmf(void *pvt, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest)
{
struct ast_frame *f = *dest;
struct dahdi_pvt *p = pvt;
int idx = analogsub_to_dahdisub(analog_index);
- ast_debug(1, "DTMF digit: %c on %s\n", f->subclass.integer, ast->name);
+ ast_debug(1, "%s DTMF digit: 0x%02X '%c' on %s\n",
+ f->frametype == AST_FRAME_DTMF_BEGIN ? "Begin" : "End",
+ f->subclass.integer, f->subclass.integer, ast->name);
if (f->subclass.integer == 'f') {
- /* Fax tone -- Handle and return NULL */
- if ((p->callprogress & CALLPROGRESS_FAX) && !p->faxhandled) {
- /* If faxbuffers are configured, use them for the fax transmission */
- if (p->usefaxbuffers && !p->bufferoverrideinuse) {
- struct dahdi_bufferinfo bi = {
- .txbufpolicy = p->faxbuf_policy,
- .bufsize = p->bufsize,
- .numbufs = p->faxbuf_no
- };
- int res;
-
- if ((res = ioctl(p->subs[idx].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
- ast_log(LOG_WARNING, "Channel '%s' unable to set buffer policy, reason: %s\n", ast->name, strerror(errno));
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ /* Fax tone -- Handle and return NULL */
+ if ((p->callprogress & CALLPROGRESS_FAX) && !p->faxhandled) {
+ /* If faxbuffers are configured, use them for the fax transmission */
+ if (p->usefaxbuffers && !p->bufferoverrideinuse) {
+ struct dahdi_bufferinfo bi = {
+ .txbufpolicy = p->faxbuf_policy,
+ .bufsize = p->bufsize,
+ .numbufs = p->faxbuf_no
+ };
+ int res;
+
+ if ((res = ioctl(p->subs[idx].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
+ ast_log(LOG_WARNING, "Channel '%s' unable to set buffer policy, reason: %s\n", ast->name, strerror(errno));
+ } else {
+ p->bufferoverrideinuse = 1;
+ }
+ }
+ p->faxhandled = 1;
+ if (p->dsp) {
+ p->dsp_features &= ~DSP_FEATURE_FAX_DETECT;
+ ast_dsp_set_features(p->dsp, p->dsp_features);
+ ast_debug(1, "Disabling FAX tone detection on %s after tone received\n", ast->name);
+ }
+ if (strcmp(ast->exten, "fax")) {
+ const char *target_context = S_OR(ast->macrocontext, ast->context);
+
+ /* We need to unlock 'ast' here because ast_exists_extension has the
+ * potential to start autoservice on the channel. Such action is prone
+ * to deadlock.
+ */
+ ast_mutex_unlock(&p->lock);
+ ast_channel_unlock(ast);
+ if (ast_exists_extension(ast, target_context, "fax", 1,
+ S_COR(ast->caller.id.number.valid, ast->caller.id.number.str, NULL))) {
+ ast_channel_lock(ast);
+ ast_mutex_lock(&p->lock);
+ ast_verb(3, "Redirecting %s to fax extension\n", ast->name);
+ /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
+ pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
+ if (ast_async_goto(ast, target_context, "fax", 1))
+ ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
+ } else {
+ ast_channel_lock(ast);
+ ast_mutex_lock(&p->lock);
+ ast_log(LOG_NOTICE, "Fax detected, but no fax extension\n");
+ }
} else {
- p->bufferoverrideinuse = 1;
- }
- }
- p->faxhandled = 1;
- if (p->dsp) {
- p->dsp_features &= ~DSP_FEATURE_FAX_DETECT;
- ast_dsp_set_features(p->dsp, p->dsp_features);
- ast_debug(1, "Disabling FAX tone detection on %s after tone received\n", ast->name);
- }
- if (strcmp(ast->exten, "fax")) {
- const char *target_context = S_OR(ast->macrocontext, ast->context);
-
- /* We need to unlock 'ast' here because ast_exists_extension has the
- * potential to start autoservice on the channel. Such action is prone
- * to deadlock.
- */
- ast_mutex_unlock(&p->lock);
- ast_channel_unlock(ast);
- if (ast_exists_extension(ast, target_context, "fax", 1,
- S_COR(ast->caller.id.number.valid, ast->caller.id.number.str, NULL))) {
- ast_channel_lock(ast);
- ast_mutex_lock(&p->lock);
- ast_verb(3, "Redirecting %s to fax extension\n", ast->name);
- /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
- pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
- if (ast_async_goto(ast, target_context, "fax", 1))
- ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
- } else {
- ast_channel_lock(ast);
- ast_mutex_lock(&p->lock);
- ast_log(LOG_NOTICE, "Fax detected, but no fax extension\n");
+ ast_debug(1, "Already in a fax extension, not redirecting\n");
}
} else {
- ast_debug(1, "Already in a fax extension, not redirecting\n");
- }
- } else {
- ast_debug(1, "Fax already handled\n");
- }
- dahdi_confmute(p, 0);
+ ast_debug(1, "Fax already handled\n");
+ }
+ dahdi_confmute(p, 0);
+ }
p->subs[idx].f.frametype = AST_FRAME_NULL;
p->subs[idx].f.subclass.integer = 0;
*dest = &p->subs[idx].f;
@@ -2225,13 +2245,20 @@
return 0;
}
+static void my_set_callwaiting(void *pvt, int callwaiting_enable)
+{
+ struct dahdi_pvt *p = pvt;
+
+ p->callwaiting = callwaiting_enable;
+}
+
static void my_cancel_cidspill(void *pvt)
{
struct dahdi_pvt *p = pvt;
- if (p->cidspill) {
- ast_free(p->cidspill);
- p->cidspill = NULL;
- }
+
+ ast_free(p->cidspill);
+ p->cidspill = NULL;
+ restore_conference(p);
}
static int my_confmute(void *pvt, int mute)
@@ -3504,7 +3531,7 @@
.lock_private = my_lock_private,
.unlock_private = my_unlock_private,
.deadlock_avoidance_private = my_deadlock_avoidance_private,
- .handle_dtmfup = my_handle_dtmfup,
+ .handle_dtmf = my_handle_dtmf,
.wink = my_wink,
.new_ast_channel = my_new_analog_ast_channel,
.dsp_set_digitmode = my_dsp_set_digitmode,
@@ -3532,6 +3559,7 @@
.check_waitingfordt = my_check_waitingfordt,
.set_confirmanswer = my_set_confirmanswer,
.check_confirmanswer = my_check_confirmanswer,
+ .set_callwaiting = my_set_callwaiting,
.cancel_cidspill = my_cancel_cidspill,
.confmute = my_confmute,
.set_pulsedial = my_set_pulsedial,
@@ -5049,17 +5077,16 @@
ast_log(LOG_WARNING, "Unable to restore conference info: %s\n", strerror(errno));
return -1;
}
- }
- ast_debug(1, "Restored conferencing\n");
+ ast_debug(1, "Restored conferencing\n");
+ }
return 0;
}
-
-static int send_callerid(struct dahdi_pvt *p);
static int send_cwcidspill(struct dahdi_pvt *p)
{
p->callwaitcas = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
if (!(p->cidspill = ast_malloc(MAX_CALLERID_SIZE)))
return -1;
p->cidlen = ast_callerid_callwaiting_generate(p->cidspill, p->callwait_name, p->callwait_num, AST_LAW(p));
@@ -5122,11 +5149,13 @@
return 0;
p->cidpos += res;
}
+ p->cid_suppress_expire = CALLWAITING_SUPPRESS_SAMPLES;
ast_free(p->cidspill);
p->cidspill = NULL;
if (p->callwaitcas) {
/* Wait for CID/CW to expire */
p->cidcwexpire = CIDCW_EXPIRE_SAMPLES;
+ p->cid_suppress_expire = p->cidcwexpire;
} else
restore_conference(p);
return 0;
@@ -5140,6 +5169,11 @@
ast_log(LOG_WARNING, "Spill already exists?!?\n");
ast_free(p->cidspill);
}
+
+ /*
+ * SAS: Subscriber Alert Signal, 440Hz for 300ms
+ * CAS: CPE Alert Signal, 2130Hz * 2750Hz sine waves
+ */
if (!(p->cidspill = ast_malloc(2400 /* SAS */ + 680 /* CAS */ + READ_SIZE * 4)))
return -1;
save_conference(p);
@@ -5615,9 +5649,7 @@
break;
}
}
- if (p->cidspill) {
- ast_free(p->cidspill);
- }
+ ast_free(p->cidspill);
if (p->use_smdi)
ast_smdi_interface_unref(p->smdi_iface);
if (p->mwi_event_sub)
@@ -6339,11 +6371,11 @@
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
p->oprmode = 0;
ast->tech_pvt = NULL;
hangup_out:
- if (p->cidspill)
- ast_free(p->cidspill);
+ ast_free(p->cidspill);
p->cidspill = NULL;
ast_mutex_unlock(&p->lock);
@@ -7037,6 +7069,24 @@
return AST_BRIDGE_RETRY;
}
+ if ((p0->callwaiting && p0->callwaitingcallerid)
+ || (p1->callwaiting && p1->callwaitingcallerid)) {
+ /*
+ * Call Waiting Caller ID requires DTMF detection to know if it
+ * can send the CID spill.
+ *
+ * For now, don't attempt to native bridge if either channel
+ * needs DTMF detection. There is code below to handle it
+ * properly until DTMF is actually seen, but due to currently
+ * unresolved issues it's ignored...
+ */
+ ast_mutex_unlock(&p0->lock);
+ ast_mutex_unlock(&p1->lock);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+ return AST_BRIDGE_FAILED_NOWARN;
+ }
+
#if defined(HAVE_PRI)
if ((dahdi_sig_pri_lib_handles(p0->sig)
&& ((struct sig_pri_chan *) p0->sig_pvt)->no_b_channel)
@@ -7474,87 +7524,98 @@
return DAHDI_ALARM_NONE;
}
-static void dahdi_handle_dtmfup(struct ast_channel *ast, int idx, struct ast_frame **dest)
+static void dahdi_handle_dtmf(struct ast_channel *ast, int idx, struct ast_frame **dest)
{
struct dahdi_pvt *p = ast->tech_pvt;
struct ast_frame *f = *dest;
- ast_debug(1, "DTMF digit: %c on %s\n", (int) f->subclass.integer, ast->name);
+ ast_debug(1, "%s DTMF digit: 0x%02X '%c' on %s\n",
+ f->frametype == AST_FRAME_DTMF_BEGIN ? "Begin" : "End",
+ f->subclass.integer, f->subclass.integer, ast->name);
if (p->confirmanswer) {
- ast_debug(1, "Confirm answer on %s!\n", ast->name);
- /* Upon receiving a DTMF digit, consider this an answer confirmation instead
- of a DTMF digit */
- p->subs[idx].f.frametype = AST_FRAME_CONTROL;
- p->subs[idx].f.subclass.integer = AST_CONTROL_ANSWER;
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ ast_debug(1, "Confirm answer on %s!\n", ast->name);
+ /* Upon receiving a DTMF digit, consider this an answer confirmation instead
+ of a DTMF digit */
+ p->subs[idx].f.frametype = AST_FRAME_CONTROL;
+ p->subs[idx].f.subclass.integer = AST_CONTROL_ANSWER;
+ /* Reset confirmanswer so DTMF's will behave properly for the duration of the call */
+ p->confirmanswer = 0;
+ } else {
+ p->subs[idx].f.frametype = AST_FRAME_NULL;
+ p->subs[idx].f.subclass.integer = 0;
+ }
*dest = &p->subs[idx].f;
- /* Reset confirmanswer so DTMF's will behave properly for the duration of the call */
- p->confirmanswer = 0;
} else if (p->callwaitcas) {
- if ((f->subclass.integer == 'A') || (f->subclass.integer == 'D')) {
- ast_debug(1, "Got some DTMF, but it's for the CAS\n");
- if (p->cidspill)
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ if ((f->subclass.integer == 'A') || (f->subclass.integer == 'D')) {
+ ast_debug(1, "Got some DTMF, but it's for the CAS\n");
ast_free(p->cidspill);
- send_cwcidspill(p);
- }
- p->callwaitcas = 0;
+ p->cidspill = NULL;
+ send_cwcidspill(p);
+ }
+ p->callwaitcas = 0;
+ }
p->subs[idx].f.frametype = AST_FRAME_NULL;
p->subs[idx].f.subclass.integer = 0;
*dest = &p->subs[idx].f;
} else if (f->subclass.integer == 'f') {
- /* Fax tone -- Handle and return NULL */
- if ((p->callprogress & CALLPROGRESS_FAX) && !p->faxhandled) {
- /* If faxbuffers are configured, use them for the fax transmission */
- if (p->usefaxbuffers && !p->bufferoverrideinuse) {
- struct dahdi_bufferinfo bi = {
- .txbufpolicy = p->faxbuf_policy,
- .bufsize = p->bufsize,
- .numbufs = p->faxbuf_no
- };
- int res;
-
- if ((res = ioctl(p->subs[idx].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
- ast_log(LOG_WARNING, "Channel '%s' unable to set buffer policy, reason: %s\n", ast->name, strerror(errno));
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ /* Fax tone -- Handle and return NULL */
+ if ((p->callprogress & CALLPROGRESS_FAX) && !p->faxhandled) {
+ /* If faxbuffers are configured, use them for the fax transmission */
+ if (p->usefaxbuffers && !p->bufferoverrideinuse) {
+ struct dahdi_bufferinfo bi = {
+ .txbufpolicy = p->faxbuf_policy,
+ .bufsize = p->bufsize,
+ .numbufs = p->faxbuf_no
+ };
+ int res;
+
+ if ((res = ioctl(p->subs[idx].dfd, DAHDI_SET_BUFINFO, &bi)) < 0) {
+ ast_log(LOG_WARNING, "Channel '%s' unable to set buffer policy, reason: %s\n", ast->name, strerror(errno));
+ } else {
+ p->bufferoverrideinuse = 1;
+ }
+ }
+ p->faxhandled = 1;
+ if (p->dsp) {
+ p->dsp_features &= ~DSP_FEATURE_FAX_DETECT;
+ ast_dsp_set_features(p->dsp, p->dsp_features);
+ ast_debug(1, "Disabling FAX tone detection on %s after tone received\n", ast->name);
+ }
+ if (strcmp(ast->exten, "fax")) {
+ const char *target_context = S_OR(ast->macrocontext, ast->context);
+
+ /* We need to unlock 'ast' here because ast_exists_extension has the
+ * potential to start autoservice on the channel. Such action is prone
+ * to deadlock.
+ */
+ ast_mutex_unlock(&p->lock);
+ ast_channel_unlock(ast);
+ if (ast_exists_extension(ast, target_context, "fax", 1,
+ S_COR(ast->caller.id.number.valid, ast->caller.id.number.str, NULL))) {
+ ast_channel_lock(ast);
+ ast_mutex_lock(&p->lock);
+ ast_verb(3, "Redirecting %s to fax extension\n", ast->name);
+ /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
+ pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
+ if (ast_async_goto(ast, target_context, "fax", 1))
+ ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
+ } else {
+ ast_channel_lock(ast);
+ ast_mutex_lock(&p->lock);
+ ast_log(LOG_NOTICE, "Fax detected, but no fax extension\n");
+ }
} else {
- p->bufferoverrideinuse = 1;
- }
- }
- p->faxhandled = 1;
- if (p->dsp) {
- p->dsp_features &= ~DSP_FEATURE_FAX_DETECT;
- ast_dsp_set_features(p->dsp, p->dsp_features);
- ast_debug(1, "Disabling FAX tone detection on %s after tone received\n", ast->name);
- }
- if (strcmp(ast->exten, "fax")) {
- const char *target_context = S_OR(ast->macrocontext, ast->context);
-
- /* We need to unlock 'ast' here because ast_exists_extension has the
- * potential to start autoservice on the channel. Such action is prone
- * to deadlock.
- */
- ast_mutex_unlock(&p->lock);
- ast_channel_unlock(ast);
- if (ast_exists_extension(ast, target_context, "fax", 1,
- S_COR(ast->caller.id.number.valid, ast->caller.id.number.str, NULL))) {
- ast_channel_lock(ast);
- ast_mutex_lock(&p->lock);
- ast_verb(3, "Redirecting %s to fax extension\n", ast->name);
- /* Save the DID/DNIS when we transfer the fax call to a "fax" extension */
- pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast->exten);
- if (ast_async_goto(ast, target_context, "fax", 1))
- ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", ast->name, target_context);
- } else {
- ast_channel_lock(ast);
- ast_mutex_lock(&p->lock);
- ast_log(LOG_NOTICE, "Fax detected, but no fax extension\n");
+ ast_debug(1, "Already in a fax extension, not redirecting\n");
}
} else {
- ast_debug(1, "Already in a fax extension, not redirecting\n");
- }
- } else {
- ast_debug(1, "Fax already handled\n");
- }
- dahdi_confmute(p, 0);
+ ast_debug(1, "Fax already handled\n");
+ }
+ dahdi_confmute(p, 0);
+ }
p->subs[idx].f.frametype = AST_FRAME_NULL;
p->subs[idx].f.subclass.integer = 0;
*dest = &p->subs[idx].f;
@@ -7628,20 +7689,32 @@
} else
#endif /* defined(HAVE_PRI) */
{
+ /* Unmute conference */
dahdi_confmute(p, 0);
p->subs[idx].f.frametype = AST_FRAME_DTMF_END;
p->subs[idx].f.subclass.integer = res & 0xff;
- }
- dahdi_handle_dtmfup(ast, idx, &f);
+ dahdi_handle_dtmf(ast, idx, &f);
+ }
return f;
}
if (res & DAHDI_EVENT_DTMFDOWN) {
ast_debug(1, "DTMF Down '%c'\n", res & 0xff);
- /* Mute conference */
- dahdi_confmute(p, 1);
- p->subs[idx].f.frametype = AST_FRAME_DTMF_BEGIN;
- p->subs[idx].f.subclass.integer = res & 0xff;
+#if defined(HAVE_PRI)
+ if (dahdi_sig_pri_lib_handles(p->sig)
+ && !((struct sig_pri_chan *) p->sig_pvt)->proceeding
+ && p->pri
+ && (p->pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING)) {
+ /* absorb event */
+ } else
+#endif /* defined(HAVE_PRI) */
+ {
+ /* Mute conference */
+ dahdi_confmute(p, 1);
+ p->subs[idx].f.frametype = AST_FRAME_DTMF_BEGIN;
+ p->subs[idx].f.subclass.integer = res & 0xff;
+ dahdi_handle_dtmf(ast, idx, &f);
+ }
return &p->subs[idx].f;
}
@@ -7813,6 +7886,7 @@
#endif
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
p->owner = NULL;
/* Don't start streaming audio yet if the incoming call isn't up yet */
if (p->subs[SUB_REAL].owner->_state != AST_STATE_UP)
@@ -7959,11 +8033,12 @@
p->subs[SUB_REAL].needringing = 0;
dahdi_set_hook(p->subs[idx].dfd, DAHDI_OFFHOOK);
ast_debug(1, "channel %d answered\n", p->channel);
- if (p->cidspill) {
- /* Cancel any running CallerID spill */
- ast_free(p->cidspill);
- p->cidspill = NULL;
- }
+
+ /* Cancel any running CallerID spill */
+ ast_free(p->cidspill);
+ p->cidspill = NULL;
+ restore_conference(p);
+
p->dialing = 0;
p->callwaitcas = 0;
if (p->confirmanswer) {
@@ -8128,6 +8203,11 @@
case SIG_FXOKS:
ast_debug(1, "Winkflash, index: %d, normal: %d, callwait: %d, thirdcall: %d\n",
idx, p->subs[SUB_REAL].dfd, p->subs[SUB_CALLWAIT].dfd, p->subs[SUB_THREEWAY].dfd);
+
+ /* Cancel any running CallerID spill */
+ ast_free(p->cidspill);
+ p->cidspill = NULL;
+ restore_conference(p);
p->callwaitcas = 0;
if (idx != SUB_REAL) {
@@ -8147,6 +8227,7 @@
}
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
/* Start music on hold if appropriate */
if (!p->subs[SUB_CALLWAIT].inthreeway && ast_bridged_channel(p->subs[SUB_CALLWAIT].owner)) {
ast_queue_control_data(p->subs[SUB_CALLWAIT].owner, AST_CONTROL_HOLD,
@@ -8479,6 +8560,7 @@
dahdi_ring_phone(p);
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
} else
ast_log(LOG_WARNING, "Absorbed on hook, but nobody is left!?!?\n");
update_conf(p);
@@ -8510,6 +8592,7 @@
}
p->callwaitingrepeat = 0;
p->cidcwexpire = 0;
+ p->cid_suppress_expire = 0;
if (ast_bridged_channel(p->owner))
ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
p->subs[SUB_REAL].needunhold = 1;
@@ -8781,21 +8864,25 @@
return &p->subs[idx].f;
}
}
- /* Ensure the CW timer decrements only on a single subchannel */
- if (p->callwaitingrepeat && dahdi_get_index(ast, p, 1) == SUB_REAL) {
- p->callwaitingrepeat--;
- }
- if (p->cidcwexpire)
- p->cidcwexpire--;
- /* Repeat callwaiting */
- if (p->callwaitingrepeat == 1) {
- p->callwaitrings++;
- dahdi_callwait(ast);
- }
- /* Expire CID/CW */
- if (p->cidcwexpire == 1) {
- ast_verb(3, "CPE does not support Call Waiting Caller*ID.\n");
- restore_conference(p);
+ if (idx == SUB_REAL) {
+ /* Ensure the CW timers decrement only on a single subchannel */
+ if (p->cidcwexpire) {
+ if (!--p->cidcwexpire) {
+ /* Expired CID/CW */
+ ast_verb(3, "CPE does not support Call Waiting Caller*ID.\n");
+ restore_conference(p);
+ }
+ }
+ if (p->cid_suppress_expire) {
+ --p->cid_suppress_expire;
+ }
+ if (p->callwaitingrepeat) {
+ if (!--p->callwaitingrepeat) {
+ /* Expired, Repeat callwaiting tone */
+ ++p->callwaitrings;
+ dahdi_callwait(ast);
+ }
+ }
}
if (p->subs[idx].linear) {
p->subs[idx].f.datalen = READ_SIZE * 2;
@@ -8899,11 +8986,31 @@
} else
f = &p->subs[idx].f;
- if (f && (f->frametype == AST_FRAME_DTMF)) {
- if (analog_lib_handles(p->sig, p->radio, p->oprmode)) {
- analog_handle_dtmfup(p->sig_pvt, ast, idx, &f);
- } else
- dahdi_handle_dtmfup(ast, idx, &f);
+ if (f) {
+ switch (f->frametype) {
+ case AST_FRAME_DTMF_BEGIN:
+ case AST_FRAME_DTMF_END:
+ if (analog_lib_handles(p->sig, p->radio, p->oprmode)) {
+ analog_handle_dtmf(p->sig_pvt, ast, idx, &f);
+ } else {
+ dahdi_handle_dtmf(ast, idx, &f);
+ }
+ break;
+ case AST_FRAME_VOICE:
+ if (p->cidspill || p->cid_suppress_expire) {
+ /* We are/were sending a caller id spill. Suppress any echo. */
+ p->subs[idx].f.frametype = AST_FRAME_NULL;
+ p->subs[idx].f.subclass.integer = 0;
+ p->subs[idx].f.samples = 0;
+ p->subs[idx].f.mallocd = 0;
+ p->subs[idx].f.offset = 0;
+ p->subs[idx].f.data.ptr = NULL;
+ p->subs[idx].f.datalen= 0;
+ }
+ break;
+ default:
+ break;
+ }
}
/* If we have a fake_event, trigger exception to handle it */
@@ -8968,7 +9075,8 @@
return 0;
}
if (p->cidspill) {
- ast_debug(1, "Dropping frame since I've still got a callerid spill\n");
+ ast_debug(1, "Dropping frame since I've still got a callerid spill on %s...\n",
+ ast->name);
return 0;
}
/* Return if it's not valid data */
@@ -10880,11 +10988,9 @@
handled = 1;
if (dahdi_set_hook(pvt->subs[SUB_REAL].dfd, DAHDI_RINGOFF) ) {
- ast_log(LOG_WARNING, "Unable to finsh RP-AS: %s mwi send aborted\n", strerror(errno));
- if(pvt->cidspill) {
- ast_free(pvt->cidspill);
- pvt->cidspill = NULL;
- }
+ ast_log(LOG_WARNING, "Unable to finish RP-AS: %s mwi send aborted\n", strerror(errno));
+ ast_free(pvt->cidspill);
+ pvt->cidspill = NULL;
pvt->mwisend_data.mwisend_current = MWI_SEND_DONE;
pvt->mwisendactive = 0;
} else {
@@ -10961,11 +11067,12 @@
res = dahdi_set_hook(i->subs[SUB_REAL].dfd, DAHDI_OFFHOOK);
if (res && (errno == EBUSY))
break;
- if (i->cidspill) {
- /* Cancel VMWI spill */
- ast_free(i->cidspill);
- i->cidspill = NULL;
- }
+
+ /* Cancel VMWI spill */
+ ast_free(i->cidspill);
+ i->cidspill = NULL;
+ restore_conference(i);
+
if (i->immediate) {
dahdi_enable_ec(i);
/* The channel is immediately up. Start right away */
Modified: trunk/channels/sig_analog.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sig_analog.c?view=diff&rev=296168&r1=296167&r2=296168
==============================================================================
--- trunk/channels/sig_analog.c (original)
+++ trunk/channels/sig_analog.c Wed Nov 24 16:52:07 2010
@@ -624,10 +624,10 @@
return -1;
}
-static void analog_cb_handle_dtmfup(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest)
-{
- if (p->calls->handle_dtmfup) {
- p->calls->handle_dtmfup(p->chan_pvt, ast, analog_index, dest);
+static void analog_cb_handle_dtmf(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest)
+{
+ if (p->calls->handle_dtmf) {
+ p->calls->handle_dtmf(p->chan_pvt, ast, analog_index, dest);
}
}
@@ -859,10 +859,7 @@
static int analog_stop_callwait(struct analog_pvt *p)
{
- if (p->callwaitingcallerid) {
- p->callwaitcas = 0;
- }
-
+ p->callwaitcas = 0;
if (p->calls->stop_callwait) {
return p->calls->stop_callwait(p->chan_pvt);
}
@@ -871,13 +868,19 @@
static int analog_callwait(struct analog_pvt *p)
{
- if (p->callwaitingcallerid) {
- p->callwaitcas = 1;
- }
+ p->callwaitcas = p->callwaitingcallerid;
if (p->calls->callwait) {
return p->calls->callwait(p->chan_pvt);
}
return 0;
+}
+
+static void analog_set_callwaiting(struct analog_pvt *p, int callwaiting_enable)
+{
+ p->callwaiting = callwaiting_enable;
+ if (p->calls->set_callwaiting) {
+ p->calls->set_callwaiting(p->chan_pvt, callwaiting_enable);
+ }
}
static void analog_set_cadence(struct analog_pvt *p, struct ast_channel *chan)
@@ -1456,7 +1459,7 @@
ast_channel_setoption(ast,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
ast_channel_setoption(ast,AST_OPTION_TDD,&x,sizeof(char),0);
p->callwaitcas = 0;
- p->callwaiting = p->permcallwaiting;
+ analog_set_callwaiting(p, p->permcallwaiting);
p->hidecallerid = p->permhidecallerid;
analog_set_dialing(p, 0);
analog_update_conf(p);
@@ -1574,35 +1577,45 @@
}
}
-void analog_handle_dtmfup(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub idx, struct ast_frame **dest)
+void analog_handle_dtmf(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub idx, struct ast_frame **dest)
{
struct ast_frame *f = *dest;
- ast_debug(1, "DTMF digit: %c on %s\n", f->subclass.integer, ast->name);
+ ast_debug(1, "%s DTMF digit: 0x%02X '%c' on %s\n",
+ f->frametype == AST_FRAME_DTMF_BEGIN ? "Begin" : "End",
+ f->subclass.integer, f->subclass.integer, ast->name);
if (analog_check_confirmanswer(p)) {
- ast_debug(1, "Confirm answer on %s!\n", ast->name);
- /* Upon receiving a DTMF digit, consider this an answer confirmation instead
- of a DTMF digit */
- p->subs[idx].f.frametype = AST_FRAME_CONTROL;
- p->subs[idx].f.subclass.integer = AST_CONTROL_ANSWER;
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ ast_debug(1, "Confirm answer on %s!\n", ast->name);
+ /* Upon receiving a DTMF digit, consider this an answer confirmation instead
+ of a DTMF digit */
+ p->subs[idx].f.frametype = AST_FRAME_CONTROL;
+ p->subs[idx].f.subclass.integer = AST_CONTROL_ANSWER;
+ /* Reset confirmanswer so DTMF's will behave properly for the duration of the call */
+ analog_set_confirmanswer(p, 0);
+ } else {
+ p->subs[idx].f.frametype = AST_FRAME_NULL;
+ p->subs[idx].f.subclass.integer = 0;
+ }
*dest = &p->subs[idx].f;
- /* Reset confirmanswer so DTMF's will behave properly for the duration of the call */
- analog_set_confirmanswer(p, 0);
} else if (p->callwaitcas) {
- if ((f->subclass.integer == 'A') || (f->subclass.integer == 'D')) {
- ast_debug(1, "Got some DTMF, but it's for the CAS\n");
- p->caller.id.name.str = p->callwait_name;
- p->caller.id.number.str = p->callwait_num;
- analog_send_callerid(p, 1, &p->caller);
- }
- if (analog_handles_digit(f))
- p->callwaitcas = 0;
+ if (f->frametype == AST_FRAME_DTMF_END) {
+ if ((f->subclass.integer == 'A') || (f->subclass.integer == 'D')) {
+ ast_debug(1, "Got some DTMF, but it's for the CAS\n");
+ p->caller.id.name.str = p->callwait_name;
+ p->caller.id.number.str = p->callwait_num;
+ analog_send_callerid(p, 1, &p->caller);
+ }
+ if (analog_handles_digit(f)) {
+ p->callwaitcas = 0;
+ }
+ }
p->subs[idx].f.frametype = AST_FRAME_NULL;
p->subs[idx].f.subclass.integer = 0;
*dest = &p->subs[idx].f;
} else {
- analog_cb_handle_dtmfup(p, ast, idx, dest);
+ analog_cb_handle_dtmf(p, ast, idx, dest);
}
}
@@ -2122,7 +2135,7 @@
} else if (p->callwaiting && !strcmp(exten, "*70")) {
ast_verb(3, "Disabling call waiting on %s\n", chan->name);
/* Disable call waiting if enabled */
- p->callwaiting = 0;
+ analog_set_callwaiting(p, 0);
res = analog_play_tone(p, idx, ANALOG_TONE_DIALRECALL);
if (res) {
ast_log(LOG_WARNING, "Unable to do dial recall on channel %s: %s\n",
@@ -2631,7 +2644,7 @@
analog_confmute(p, 0);
p->subs[idx].f.frametype = AST_FRAME_DTMF_END;
p->subs[idx].f.subclass.integer = res & 0xff;
- analog_handle_dtmfup(p, ast, idx, &f);
+ analog_handle_dtmf(p, ast, idx, &f);
return f;
}
@@ -2641,6 +2654,7 @@
analog_confmute(p, 1);
p->subs[idx].f.frametype = AST_FRAME_DTMF_BEGIN;
p->subs[idx].f.subclass.integer = res & 0xff;
+ analog_handle_dtmf(p, ast, idx, &f);
return f;
}
@@ -2890,7 +2904,10 @@
analog_set_needringing(p, 0);
analog_off_hook(p);
ast_debug(1, "channel %d answered\n", p->channel);
+
+ /* Cancel any running CallerID spill */
analog_cancel_cidspill(p);
+
analog_set_dialing(p, 0);
p->callwaitcas = 0;
if (analog_check_confirmanswer(p)) {
@@ -3050,6 +3067,8 @@
ast_debug(1, "Winkflash, index: %d, normal: %d, callwait: %d, thirdcall: %d\n",
idx, analog_get_sub_fd(p, ANALOG_SUB_REAL), analog_get_sub_fd(p, ANALOG_SUB_CALLWAIT), analog_get_sub_fd(p, ANALOG_SUB_THREEWAY));
+ /* Cancel any running CallerID spill */
+ analog_cancel_cidspill(p);
p->callwaitcas = 0;
if (idx != ANALOG_SUB_REAL) {
@@ -3568,7 +3587,10 @@
if (res && (errno == EBUSY)) {
break;
}
+
+ /* Cancel VMWI spill */
analog_cancel_cidspill(i);
+
if (i->immediate) {
analog_set_echocanceller(i, 1);
/* The channel is immediately up. Start right away */
@@ -3826,7 +3848,7 @@
p->permcallwaiting = 0;
}
- p->callwaiting = p->permcallwaiting;
+ analog_set_callwaiting(p, p->permcallwaiting);
return 0;
}
Modified: trunk/channels/sig_analog.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sig_analog.h?view=diff&rev=296168&r1=296167&r2=296168
==============================================================================
--- trunk/channels/sig_analog.h (original)
+++ trunk/channels/sig_analog.h Wed Nov 24 16:52:07 2010
@@ -134,10 +134,10 @@
/* Do deadlock avoidance for the private signaling structure lock. */
void (* const deadlock_avoidance_private)(void *pvt);
- /* Function which is called back to handle any other DTMF up events that are received. Called by analog_handle_event. Why is this
+ /* Function which is called back to handle any other DTMF events that are received. Called by analog_handle_event. Why is this
* important to use, instead of just directly using events received before they are passed into the library? Because sometimes,
* (CWCID) the library absorbs DTMF events received. */
- void (* const handle_dtmfup)(void *pvt, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest);
+ void (* const handle_dtmf)(void *pvt, struct ast_channel *ast, enum analog_sub analog_index, struct ast_frame **dest);
int (* const get_event)(void *pvt);
int (* const wait_event)(void *pvt);
@@ -228,6 +228,7 @@
int (* const check_waitingfordt)(void *pvt);
void (* const set_confirmanswer)(void *pvt, int flag);
int (* const check_confirmanswer)(void *pvt);
+ void (* const set_callwaiting)(void *pvt, int callwaiting_enable);
void (* const cancel_cidspill)(void *pvt);
int (* const confmute)(void *pvt, int mute);
void (* const set_pulsedial)(void *pvt, int flag);
@@ -271,14 +272,14 @@
unsigned int dahditrcallerid:1; /*!< should we use the callerid from incoming call on dahdi transfer or not */
unsigned int hanguponpolarityswitch:1;
unsigned int immediate:1;
- unsigned int permcallwaiting:1;
+ unsigned int permcallwaiting:1; /*!< TRUE if call waiting is enabled. (Configured option) */
unsigned int permhidecallerid:1; /*!< Whether to hide our outgoing caller ID or not */
unsigned int pulse:1;
unsigned int threewaycalling:1;
unsigned int transfer:1;
unsigned int transfertobusy:1; /*!< allow flash-transfers to busy channels */
unsigned int use_callerid:1; /*!< Whether or not to use caller id on this channel */
- unsigned int callwaitingcallerid:1;
+ unsigned int callwaitingcallerid:1; /*!< TRUE if send caller ID for Call Waiting */
/*!
* \brief TRUE if SMDI (Simplified Message Desk Interface) is enabled
*/
@@ -289,6 +290,7 @@
/* Not used for anything but log messages. Could be just the TCID */
int channel; /*!< Channel Number */
+
enum analog_sigtype outsigmod;
int echotraining;
int cid_signalling; /*!< Asterisk callerid type we're using */
@@ -301,13 +303,21 @@
/* XXX: All variables after this are internal */
- unsigned int callwaiting:1;
+ unsigned int callwaiting:1; /*!< TRUE if call waiting is enabled. (Active option) */
unsigned int dialednone:1;
unsigned int dialing:1; /*!< TRUE if in the process of dialing digits or sending something */
unsigned int dnd:1; /*!< TRUE if Do-Not-Disturb is enabled. */
unsigned int echobreak:1;
unsigned int hidecallerid:1;
unsigned int outgoing:1;
+ unsigned int inalarm:1;
+ /*!
+ * \brief TRUE if Call Waiting (CW) CPE Alert Signal (CAS) is being sent.
+ * \note
+ * After CAS is sent, the call waiting caller id will be sent if the phone
+ * gives a positive reply.
+ */
+ unsigned int callwaitcas:1;
char callwait_num[AST_MAX_EXTENSION];
char callwait_name[AST_MAX_EXTENSION];
@@ -331,10 +341,6 @@
struct ast_channel *ss_astchan;
/* All variables after this are definitely going to be audited */
- unsigned int inalarm:1;
-
- int callwaitcas;
-
int ringt;
int ringt_base;
};
@@ -360,7 +366,7 @@
int analog_config_complete(struct analog_pvt *p);
-void analog_handle_dtmfup(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub index, struct ast_frame **dest);
+void analog_handle_dtmf(struct analog_pvt *p, struct ast_channel *ast, enum analog_sub index, struct ast_frame **dest);
enum analog_cid_start analog_str_to_cidstart(const char *value);
Modified: trunk/channels/sig_pri.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sig_pri.h?view=diff&rev=296168&r1=296167&r2=296168
==============================================================================
--- trunk/channels/sig_pri.h (original)
+++ trunk/channels/sig_pri.h Wed Nov 24 16:52:07 2010
@@ -90,10 +90,10 @@
void (* const unlock_private)(void *pvt);
[... 12 lines stripped ...]
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