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authorrmudgett <rmudgett@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-24 22:42:45 +0000
committerrmudgett <rmudgett@f38db490-d61c-443f-a65b-d21fe96a405b>2010-11-24 22:42:45 +0000
commit04bb1bcffe40311959e7f9fb2758a0390d130463 (patch)
tree4bf594ad2227e81bd8721b9e8c0eb6ee94a06ea4
parent2fb244fe00c57426ca0e0b5d7f446680f310c418 (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@296166 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_dahdi.c310
1 files changed, 197 insertions, 113 deletions
diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index 67836b48d..34102c493 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -399,11 +399,12 @@ 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;
@@ -1047,7 +1048,8 @@ static struct dahdi_pvt {
int cid_start; /*!< CID start indicator, polarity or ring */
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. */
@@ -1068,7 +1070,12 @@ static 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;
@@ -3007,8 +3014,8 @@ 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;
}
@@ -3018,6 +3025,7 @@ 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));
@@ -3077,11 +3085,13 @@ 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;
@@ -3095,6 +3105,11 @@ 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);
@@ -3901,6 +3916,8 @@ static void destroy_dahdi_pvt(struct dahdi_pvt **pvt)
p->prev->next = p->next;
if (p->next)
p->next->prev = p->prev;
+
+ ast_free(p->cidspill);
if (p->use_smdi)
ast_smdi_interface_unref(p->smdi_iface);
if (p->mwi_event_sub)
@@ -3979,15 +3996,11 @@ static void destroy_all_channels(void)
/* Destroy all the interfaces and free their memory */
p = iflist;
while (p) {
- /* Free any callerid */
- if (p->cidspill)
- ast_free(p->cidspill);
pl = p;
p = p->next;
x = pl->channel;
/* Free associated memory */
- if (pl)
- destroy_dahdi_pvt(&pl);
+ destroy_dahdi_pvt(&pl);
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_2 "Unregistered channel %d\n", x);
}
@@ -4725,15 +4738,14 @@ static int dahdi_hangup(struct ast_channel *ast)
default:
tone_zone_play_tone(p->subs[SUB_REAL].dfd, -1);
}
- if (p->cidspill)
- ast_free(p->cidspill);
+ ast_free(p->cidspill);
+ p->cidspill = NULL;
if (p->sig)
dahdi_disable_ec(p);
x = 0;
ast_channel_setoption(ast,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
ast_channel_setoption(ast,AST_OPTION_TDD,&x,sizeof(char),0);
p->didtdd = 0;
- p->cidspill = NULL;
p->callwaitcas = 0;
p->callwaiting = p->permcallwaiting;
p->hidecallerid = p->permhidecallerid;
@@ -4767,6 +4779,7 @@ 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;
ast_mutex_unlock(&p->lock);
@@ -5371,6 +5384,24 @@ 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 ((oi0 == SUB_REAL) && (oi1 == SUB_REAL)) {
if (p0->owner && p1->owner) {
/* If we don't have a call-wait in a 3-way, and we aren't in a 3-way, we can be master */
@@ -5785,81 +5816,92 @@ 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", f->subclass, ast->name);
+ ast_debug(1, "%s DTMF digit: 0x%02X '%c' on %s\n",
+ f->frametype == AST_FRAME_DTMF_BEGIN ? "Begin" : "End",
+ f->subclass, f->subclass, 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 = 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 = 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 = 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 == 'A') || (f->subclass == '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 == 'A') || (f->subclass == '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 = 0;
*dest = &p->subs[idx].f;
} else if (f->subclass == '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 faxbuffer 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 faxbuffer policy, reason: %s\n", ast->name, strerror(errno));
+ } else {
+ p->bufferoverrideinuse = 1;
+ }
}
- }
- p->faxhandled = 1;
- if (strcmp(ast->exten, "fax")) {
- const char *target_context = S_OR(ast->macrocontext, ast->context);
+ p->faxhandled = 1;
+ 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, ast->cid.cid_num)) {
- 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, ast->cid.cid_num)) {
+ 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 = 0;
*dest = &p->subs[idx].f;
@@ -5920,27 +5962,35 @@ static struct ast_frame *dahdi_handle_event(struct ast_channel *ast)
if (res & (DAHDI_EVENT_PULSEDIGIT | DAHDI_EVENT_DTMFUP)) {
p->pulsedial = (res & DAHDI_EVENT_PULSEDIGIT) ? 1 : 0;
ast_debug(1, "Detected %sdigit '%c'\n", p->pulsedial ? "pulse ": "", res & 0xff);
-#ifdef HAVE_PRI
+#if defined(HAVE_PRI)
if (!p->proceeding && ((p->sig == SIG_PRI) || (p->sig == SIG_BRI) || (p->sig == SIG_BRI_PTMP)) && p->pri && (p->pri->overlapdial & DAHDI_OVERLAPDIAL_INCOMING)) {
/* absorb event */
- } else {
-#endif
+ } 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 = res & 0xff;
-#ifdef HAVE_PRI
+ dahdi_handle_dtmf(ast, idx, &f);
}
-#endif
- 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 = res & 0xff;
+#if defined(HAVE_PRI)
+ if (!p->proceeding && ((p->sig == SIG_PRI) || (p->sig == SIG_BRI) || (p->sig == SIG_BRI_PTMP)) && 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 = res & 0xff;
+ dahdi_handle_dtmf(ast, idx, &f);
+ }
return &p->subs[idx].f;
}
@@ -6122,6 +6172,7 @@ 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)
@@ -6266,11 +6317,12 @@ 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) {
@@ -6441,6 +6493,11 @@ 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) {
@@ -6471,6 +6528,7 @@ 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)) {
@@ -6866,6 +6924,7 @@ 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 %s, but nobody is left!?!?\n",
event2str(res));
@@ -6896,6 +6955,7 @@ 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;
@@ -7158,21 +7218,25 @@ 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;
@@ -7274,8 +7338,28 @@ static struct ast_frame *dahdi_read(struct ast_channel *ast)
} else
f = &p->subs[idx].f;
- if (f && (f->frametype == AST_FRAME_DTMF))
- dahdi_handle_dtmfup(ast, idx, &f);
+ if (f) {
+ switch (f->frametype) {
+ case AST_FRAME_DTMF_BEGIN:
+ case AST_FRAME_DTMF_END:
+ 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 = 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 */
if (p->fake_event)
@@ -7355,7 +7439,8 @@ 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 */
@@ -9457,11 +9542,9 @@ 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 {
@@ -9537,11 +9620,12 @@ static struct dahdi_pvt *handle_init_event(struct dahdi_pvt *i, int event)
i->fxsoffhookstate = 1;
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 */