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Merged revisions 296167 via svnmerge from

Browse Source 
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.
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
Richard Mudgett 15 years ago
parent ddd0ae53d2
commit ccdc417ab5
4 changed files with 349 additions and 214 deletions
Show Stats Download Patch File Download Diff File

449
channels/chan_dahdi.c
Unescape View File

@ -492,11 +492,12 @@ static inline int dahdi_wait_event(int fd)
#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 dahdi_pvt {
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 @@ struct dahdi_pvt {
* 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 @@ static int my_distinctive_ring(struct ast_channel *chan, void *pvt, int idx, int
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 @@ static int my_callwait(void *pvt)
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 @@ static int my_send_callerid(void *pvt, int cwcid, struct ast_party_caller *calle
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 @@ static int my_send_callerid(void *pvt, int cwcid, struct ast_party_caller *calle
p->cidlen += READ_SIZE * 4;
}
p->cidpos = 0;
p->cid_suppress_expire = 0;
send_callerid(p);
}
return 0;
@ -1978,68 +1994,72 @@ static int reset_conf(struct dahdi_pvt *p);
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));
} else {
p->bufferoverrideinuse = 1;
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);
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);
/* 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 {
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");
ast_debug(1, "Fax already handled\n");
}
} else {
ast_debug(1, "Fax already handled\n");
dahdi_confmute(p, 0);
}
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 @@ static int my_check_confirmanswer(void *pvt)
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 @@ static struct analog_callback dahdi_analog_callbacks =
.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 @@ static struct analog_callback dahdi_analog_callbacks =
.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 @@ static int restore_conference(struct dahdi_pvt *p)
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 @@ static int send_callerid(struct dahdi_pvt *p)
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 @@ static int dahdi_callwait(struct ast_channel *ast)
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 @@ static void destroy_dahdi_pvt(struct dahdi_pvt *pvt)
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 @@ static int dahdi_hangup(struct ast_channel *ast)
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 @@ static enum ast_bridge_result dahdi_bridge(struct ast_channel *c0, struct ast_ch
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 @@ static int get_alarms(struct dahdi_pvt *p)
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->cidspill = NULL;
send_cwcidspill(p);
}
p->callwaitcas = 0;
}
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));
} else {
p->bufferoverrideinuse = 1;
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);
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);
/* 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 {
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");
ast_debug(1, "Fax already handled\n");
}
} else {
ast_debug(1, "Fax already handled\n");
dahdi_confmute(p, 0);
}
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 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
} 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_dtmf(ast, idx, &f);
}
dahdi_handle_dtmfup(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 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
#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 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
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 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
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 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
}
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 @@ static struct ast_frame *__dahdi_exception(struct ast_channel *ast)
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 @@ static struct ast_frame *__dahdi_exception(struct ast_channel *ast)
}
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 @@ static struct ast_frame *dahdi_read(struct ast_channel *ast)
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 @@ static struct ast_frame *dahdi_read(struct ast_channel *ast)
} 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 @@ static int dahdi_write(struct ast_channel *ast, struct ast_frame *frame)
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 @@ static int mwi_send_process_event(struct dahdi_pvt * pvt, int event)
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 @@ static struct dahdi_pvt *handle_init_event(struct dahdi_pvt *i, int event)
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 */

84
channels/sig_analog.c
Unescape View File

@ -624,10 +624,10 @@ static int analog_dsp_set_digitmode(struct analog_pvt *p, enum analog_dsp_digitm
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)
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_dtmfup) {
p->calls->handle_dtmfup(p->chan_pvt, ast, analog_index, dest);
if (p->calls->handle_dtmf) {
p->calls->handle_dtmf(p->chan_pvt, ast, analog_index, dest);
}
}
@ -859,10 +859,7 @@ int analog_available(struct analog_pvt *p)
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);
}
@ -871,15 +868,21 @@ static int analog_stop_callwait(struct analog_pvt *p)
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)
{
if (p->calls->set_cadence) {
@ -1456,7 +1459,7 @@ int analog_hangup(struct analog_pvt *p, struct ast_channel *ast)
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 @@ static int analog_handles_digit(struct ast_frame *f)
}
}
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 (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;
}
}
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 @@ static void *__analog_ss_thread(void *data)
} 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 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
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 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
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 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
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 @@ static struct ast_frame *__analog_handle_event(struct analog_pvt *p, struct ast_
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 @@ void *analog_handle_init_event(struct analog_pvt *i, int event)
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 @@ int analog_config_complete(struct analog_pvt *p)
p->permcallwaiting = 0;
}
p->callwaiting = p->permcallwaiting;
analog_set_callwaiting(p, p->permcallwaiting);
return 0;
}

26
channels/sig_analog.h
Unescape View File

@ -134,10 +134,10 @@ struct analog_callback {
/* 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 @@ struct analog_callback {
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 @@ struct analog_pvt {
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 @@ struct analog_pvt {
/* 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 @@ struct analog_pvt {
/* 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 analog_pvt {
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 @@ void *analog_handle_init_event(struct analog_pvt *i, int event);
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);

4
channels/sig_pri.h
Unescape View File

@ -90,10 +90,10 @@ struct sig_pri_callback {
void (* const unlock_private)(void *pvt);
/* Lock the private in the signalling private structure. ... */
void (* const lock_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 dial_digits)(void *pvt, enum analog_sub sub, struct analog_dialoperation *dop);
int (* const play_tone)(void *pvt, enum sig_pri_tone tone);

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