Author: dlee
Date: Tue Aug 6 09:48:12 2013
New Revision: 396335
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=396335
Log:
Merged revisions 396175-396331 from http://svn.asterisk.org/svn/asterisk/trunk
Added:
team/dlee/ASTERISK-22243/res/parking/parking_tests.c
- copied unchanged from r396331, trunk/res/parking/parking_tests.c
Modified:
team/dlee/ASTERISK-22243/ (props changed)
team/dlee/ASTERISK-22243/UPGRADE-11.txt
team/dlee/ASTERISK-22243/apps/app_minivm.c
team/dlee/ASTERISK-22243/apps/app_playback.c
team/dlee/ASTERISK-22243/apps/app_stack.c
team/dlee/ASTERISK-22243/apps/app_voicemail.c
team/dlee/ASTERISK-22243/bridges/bridge_holding.c
team/dlee/ASTERISK-22243/channels/chan_sip.c
team/dlee/ASTERISK-22243/funcs/func_frame_trace.c
team/dlee/ASTERISK-22243/funcs/func_global.c
team/dlee/ASTERISK-22243/funcs/func_strings.c
team/dlee/ASTERISK-22243/include/asterisk/app.h
team/dlee/ASTERISK-22243/include/asterisk/bridge_basic.h
team/dlee/ASTERISK-22243/include/asterisk/bridge_roles.h
team/dlee/ASTERISK-22243/include/asterisk/frame.h
team/dlee/ASTERISK-22243/include/asterisk/stasis.h
team/dlee/ASTERISK-22243/include/asterisk/stasis_app.h
team/dlee/ASTERISK-22243/include/asterisk/stasis_app_recording.h
team/dlee/ASTERISK-22243/main/app.c
team/dlee/ASTERISK-22243/main/bridge_basic.c
team/dlee/ASTERISK-22243/main/bridge_roles.c
team/dlee/ASTERISK-22243/main/cdr.c
team/dlee/ASTERISK-22243/main/channel.c
team/dlee/ASTERISK-22243/main/features.c
team/dlee/ASTERISK-22243/main/pbx.c
team/dlee/ASTERISK-22243/main/stasis_cache.c
team/dlee/ASTERISK-22243/main/stasis_cache_pattern.c
team/dlee/ASTERISK-22243/pbx/pbx_dundi.c
team/dlee/ASTERISK-22243/pbx/pbx_loopback.c
team/dlee/ASTERISK-22243/res/ari/resource_bridges.c
team/dlee/ASTERISK-22243/res/ari/resource_bridges.h
team/dlee/ASTERISK-22243/res/ari/resource_recordings.c
team/dlee/ASTERISK-22243/res/ari/resource_recordings.h
team/dlee/ASTERISK-22243/res/parking/parking_bridge.c
team/dlee/ASTERISK-22243/res/parking/res_parking.h
team/dlee/ASTERISK-22243/res/res_ari_bridges.c
team/dlee/ASTERISK-22243/res/res_ari_recordings.c
team/dlee/ASTERISK-22243/res/res_parking.c
team/dlee/ASTERISK-22243/res/res_pjsip.c
team/dlee/ASTERISK-22243/res/res_pjsip/config_system.c
team/dlee/ASTERISK-22243/res/res_pjsip/include/res_pjsip_private.h
team/dlee/ASTERISK-22243/res/res_pjsip_exten_state.c
team/dlee/ASTERISK-22243/res/res_pjsip_messaging.c
team/dlee/ASTERISK-22243/res/res_pjsip_notify.c
team/dlee/ASTERISK-22243/res/res_pjsip_outbound_registration.c
team/dlee/ASTERISK-22243/res/res_stasis_recording.c
team/dlee/ASTERISK-22243/res/stasis/control.c
team/dlee/ASTERISK-22243/rest-api/api-docs/bridges.json
team/dlee/ASTERISK-22243/rest-api/api-docs/recordings.json
team/dlee/ASTERISK-22243/tests/test_stasis.c
team/dlee/ASTERISK-22243/utils/extconf.c
Propchange: team/dlee/ASTERISK-22243/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/dlee/ASTERISK-22243/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Aug 6 09:48:12 2013
@@ -1,1 +1,1 @@
-/trunk:1-396167
+/trunk:1-396334
Modified: team/dlee/ASTERISK-22243/UPGRADE-11.txt
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/UPGRADE-11.txt?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/UPGRADE-11.txt (original)
+++ team/dlee/ASTERISK-22243/UPGRADE-11.txt Tue Aug 6 09:48:12 2013
@@ -19,6 +19,11 @@
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
===
===========================================================
+From 11.5 to 11.6:
+* res_agi will now properly indicate if there was an error in streaming an
+ audio file. The result code will be -1 and the result returned from the
+ the function will be RESULT_FAILURE instead of the prior behavior of always
+ returning RESULT_SUCCESS even if there was an error.
From 11.4 to 11.5:
* The default settings for chan_sip are now overriden properly by the general
Modified: team/dlee/ASTERISK-22243/apps/app_minivm.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/apps/app_minivm.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/apps/app_minivm.c (original)
+++ team/dlee/ASTERISK-22243/apps/app_minivm.c Tue Aug 6 09:48:12 2013
@@ -1674,7 +1674,7 @@
ast_channel_setoption(chan, AST_OPTION_RXGAIN,
&record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, MVM_OPERATOR))
canceldtmf = "0";
- cmd = ast_play_and_record_full(chan, playfile,
recordfile, maxtime, fmt, duration, sound_duration, global_silencethreshold,
global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0,
AST_RECORD_IF_EXISTS_OVERWRITE);
+ cmd = ast_play_and_record_full(chan, playfile,
recordfile, maxtime, fmt, duration, sound_duration, 0, global_silencethreshold,
global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0,
AST_RECORD_IF_EXISTS_OVERWRITE);
if (record_gain)
ast_channel_setoption(chan, AST_OPTION_RXGAIN,
&zero_gain, sizeof(zero_gain), 0);
if (cmd == -1) /* User has hung up, no options to give
*/
Modified: team/dlee/ASTERISK-22243/apps/app_playback.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/apps/app_playback.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/apps/app_playback.c (original)
+++ team/dlee/ASTERISK-22243/apps/app_playback.c Tue Aug 6 09:48:12 2013
@@ -220,6 +220,10 @@
s = x + 1;
ast_debug(2, "value is <%s>\n", s);
n = ast_var_assign("SAY", s);
+ if (!n) {
+ ast_log(LOG_ERROR, "Memory allocation error in do_say\n");
+ return -1;
+ }
AST_LIST_INSERT_HEAD(&head, n, entries);
/* scan the body, one piece at a time */
Modified: team/dlee/ASTERISK-22243/apps/app_stack.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/apps/app_stack.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/apps/app_stack.c (original)
+++ team/dlee/ASTERISK-22243/apps/app_stack.c Tue Aug 6 09:48:12 2013
@@ -273,8 +273,9 @@
}
if (!found) {
- variables = ast_var_assign(var, "");
- AST_LIST_INSERT_HEAD(&frame->varshead, variables, entries);
+ if ((variables = ast_var_assign(var, ""))) {
+ AST_LIST_INSERT_HEAD(&frame->varshead, variables,
entries);
+ }
pbx_builtin_pushvar_helper(chan, var, value);
} else {
pbx_builtin_setvar_helper(chan, var, value);
Modified: team/dlee/ASTERISK-22243/apps/app_voicemail.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/apps/app_voicemail.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/apps/app_voicemail.c (original)
+++ team/dlee/ASTERISK-22243/apps/app_voicemail.c Tue Aug 6 09:48:12 2013
@@ -14682,7 +14682,7 @@
ast_channel_setoption(chan, AST_OPTION_RXGAIN,
&record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, VM_OPERATOR))
canceldtmf = "0";
- cmd = ast_play_and_record_full(chan, playfile,
tempfile, maxtime, fmt, duration, sound_duration, silencethreshold, maxsilence,
unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
+ cmd = ast_play_and_record_full(chan, playfile,
tempfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold,
maxsilence, unlockdir, acceptdtmf, canceldtmf, 0,
AST_RECORD_IF_EXISTS_OVERWRITE);
if (strchr(canceldtmf, cmd)) {
/* need this flag here to distinguish between pressing
'0' during message recording or after */
canceleddtmf = 1;
Modified: team/dlee/ASTERISK-22243/bridges/bridge_holding.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/bridges/bridge_holding.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/bridges/bridge_holding.c (original)
+++ team/dlee/ASTERISK-22243/bridges/bridge_holding.c Tue Aug 6 09:48:12 2013
@@ -81,7 +81,7 @@
ast_moh_stop(bridge_channel->chan);
break;
case IDLE_MODE_RINGING:
- ast_indicate(bridge_channel->chan, -1);
+ ast_bridge_channel_queue_control_data(bridge_channel, -1, NULL,
0);
break;
case IDLE_MODE_NONE:
break;
@@ -124,7 +124,7 @@
ast_moh_start(bridge_channel->chan, ast_strlen_zero(moh_class)
? NULL : moh_class, NULL);
break;
case IDLE_MODE_RINGING:
- ast_indicate(bridge_channel->chan, AST_CONTROL_RINGING);
+ ast_bridge_channel_queue_control_data(bridge_channel,
AST_CONTROL_RINGING, NULL, 0);
break;
case IDLE_MODE_NONE:
break;
@@ -302,6 +302,44 @@
return 0;
}
+static void holding_bridge_suspend(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel)
+{
+ struct holding_channel *hc = bridge_channel ? bridge_channel->tech_pvt
: NULL;
+
+ if (!hc) {
+ return;
+ }
+
+ if (!ast_test_flag(&hc->holding_roles, HOLDING_ROLE_PARTICIPANT)) {
+ return;
+ }
+
+ participant_stop_hold_audio(bridge_channel);
+}
+
+static void holding_bridge_unsuspend(struct ast_bridge *bridge, struct
ast_bridge_channel *bridge_channel)
+{
+ struct holding_channel *hc = bridge_channel ? bridge_channel->tech_pvt
: NULL;
+ struct ast_bridge_channel *announcer_channel = bridge->tech_pvt;
+
+ if (!hc) {
+ return;
+ }
+
+ /* If the bridge channel being unsuspended is not a participant, no
need to do anything. */
+ if (!ast_test_flag(&hc->holding_roles, HOLDING_ROLE_PARTICIPANT)) {
+ return;
+ }
+
+ /* If there is currently an announcer channel in the bridge, there is
also no need to do anything. */
+ if (announcer_channel) {
+ return;
+ }
+
+ /* Otherwise we need to resume the entertainment. */
+ participant_start_hold_audio(bridge_channel);
+}
+
static struct ast_bridge_technology holding_bridge = {
.name = "holding_bridge",
.capabilities = AST_BRIDGE_CAPABILITY_HOLDING,
@@ -309,6 +347,8 @@
.write = holding_bridge_write,
.join = holding_bridge_join,
.leave = holding_bridge_leave,
+ .suspend = holding_bridge_suspend,
+ .unsuspend = holding_bridge_unsuspend,
};
static int unload_module(void)
Modified: team/dlee/ASTERISK-22243/channels/chan_sip.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/channels/chan_sip.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/channels/chan_sip.c (original)
+++ team/dlee/ASTERISK-22243/channels/chan_sip.c Tue Aug 6 09:48:12 2013
@@ -15272,6 +15272,8 @@
append_history(p, "RegistryInit", "Account: %s@%s",
r->username, r->hostname);
}
+ p->socket.type = r->transport;
+
/* Use port number specified if no SRV record was found */
if (!ast_sockaddr_isnull(&r->us)) {
if (!ast_sockaddr_port(&r->us) && r->portno) {
Modified: team/dlee/ASTERISK-22243/funcs/func_frame_trace.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/funcs/func_frame_trace.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/funcs/func_frame_trace.c (original)
+++ team/dlee/ASTERISK-22243/funcs/func_frame_trace.c Tue Aug 6 09:48:12 2013
@@ -343,6 +343,18 @@
case AST_CONTROL_STREAM_FORWARD:
ast_verbose("SubClass: STREAM_FORWARD\n");
break;
+ case AST_CONTROL_RECORD_CANCEL:
+ ast_verbose("SubClass: RECORD_CANCEL\n");
+ break;
+ case AST_CONTROL_RECORD_STOP:
+ ast_verbose("SubClass: RECORD_STOP\n");
+ break;
+ case AST_CONTROL_RECORD_SUSPEND:
+ ast_verbose("SubClass: RECORD_SUSPEND\n");
+ break;
+ case AST_CONTROL_RECORD_MUTE:
+ ast_verbose("SubClass: RECORD_MUTE\n");
+ break;
}
if (frame->subclass.integer == -1) {
Modified: team/dlee/ASTERISK-22243/funcs/func_global.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/funcs/func_global.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/funcs/func_global.c (original)
+++ team/dlee/ASTERISK-22243/funcs/func_global.c Tue Aug 6 09:48:12 2013
@@ -282,11 +282,12 @@
}
AST_LIST_TRAVERSE_SAFE_END;
- var = ast_var_assign(args.var, S_OR(value, ""));
- AST_LIST_INSERT_HEAD(varshead, var, entries);
-
- sprintf(shared_buffer, "SHARED(%s)", args.var);
- ast_channel_publish_varset(chan, shared_buffer, value);
+ if ((var = ast_var_assign(args.var, S_OR(value, "")))) {
+ AST_LIST_INSERT_HEAD(varshead, var, entries);
+
+ sprintf(shared_buffer, "SHARED(%s)", args.var);
+ ast_channel_publish_varset(chan, shared_buffer, value);
+ }
ast_channel_unlock(chan);
Modified: team/dlee/ASTERISK-22243/funcs/func_strings.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/funcs/func_strings.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/funcs/func_strings.c (original)
+++ team/dlee/ASTERISK-22243/funcs/func_strings.c Tue Aug 6 09:48:12 2013
@@ -1698,6 +1698,12 @@
for (i = 0; i < ARRAY_LEN(test_args); i++) {
struct ast_var_t *var = ast_var_assign("FIELDS",
test_args[i].fields);
+ if (!var) {
+ ast_test_status_update(test, "Out of memory\n");
+ res = AST_TEST_FAIL;
+ break;
+ }
+
AST_LIST_INSERT_HEAD(ast_channel_varshead(chan), var, entries);
snprintf(expression, sizeof(expression),
"${FIELDNUM(%s,%s,%s)}", var->name, test_args[i].delim, test_args[i].field);
@@ -1800,6 +1806,13 @@
char tmp[512], tmp2[512] = "";
struct ast_var_t *var = ast_var_assign("test_string",
test_strings[i][0]);
+ if (!var) {
+ ast_test_status_update(test, "Unable to allocate
variable\n");
+ ast_free(str);
+ ast_channel_release(chan);
+ return AST_TEST_FAIL;
+ }
+
AST_LIST_INSERT_HEAD(ast_channel_varshead(chan), var, entries);
if (test_strings[i][3]) {
Modified: team/dlee/ASTERISK-22243/include/asterisk/app.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/app.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/app.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/app.h Tue Aug 6 09:48:12 2013
@@ -709,11 +709,12 @@
* skip_confirmation_sound is false.
*
* \param chan the channel being recorded
- * \param playfile Filename of sound to play before recording begins
+ * \param playfile Filename of sound to play before recording begins. A beep
is also played when playfile completes, before the recording begins.
* \param recordfile Filename to save the recording
* \param maxtime_sec Longest possible message length in seconds
* \param fmt string containing all formats to be recorded delimited by '|'
* \param duration pointer to integer for storing length of the recording
+ * \param beep If true, play a beep before recording begins (and doesn't play
\a playfile)
* \param sound_duration pointer to integer for storing length of the
recording minus all silence
* \param silencethreshold tolerance of noise levels that can be considered
silence for the purpose of silence timeout, -1 for default
* \param maxsilence_ms Length of time in milliseconds which will trigger a
timeout from silence, -1 for default
@@ -728,7 +729,7 @@
* \retval 't' Recording ended from the message exceeding the maximum duration
* \retval dtmfchar Recording ended via the return value's DTMF character for
either cancel or accept.
*/
-int ast_play_and_record_full(struct ast_channel *chan, const char *playfile,
const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int
*sound_duration, int silencethreshold, int maxsilence_ms, const char *path,
const char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound,
enum ast_record_if_exists if_exists);
+int ast_play_and_record_full(struct ast_channel *chan, const char *playfile,
const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int
*sound_duration, int beep, int silencethreshold, int maxsilence_ms, const char
*path, const char *acceptdtmf, const char *canceldtmf, int
skip_confirmation_sound, enum ast_record_if_exists if_exists);
/*!
* \brief Record a file based on input from a channel. Use default accept and
cancel DTMF.
Modified: team/dlee/ASTERISK-22243/include/asterisk/bridge_basic.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/bridge_basic.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/bridge_basic.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/bridge_basic.h Tue Aug 6
09:48:12 2013
@@ -89,6 +89,22 @@
int ast_bridge_features_ds_set(struct ast_channel *chan, struct ast_flags
*flags);
/*!
+ * \brief Append basic bridge DTMF feature flags on the channel.
+ * \since 12.0.0
+ *
+ * \param chan Channel to append DTMF features datastore.
+ * \param flags Builtin DTMF feature flags. (ast_bridge_config flags)
+ *
+ * \note The channel must be locked before calling this function.
+ * \note This function differs from ast_bridge_features_ds_set only in that it
won't
+ * remove features already set on the channel.
+ *
+ * \retval 0 on success.
+ * \retval -1 on error.
+ */
+int ast_bridge_features_ds_append(struct ast_channel *chan, struct ast_flags
*flags);
+
+/*!
* \brief Setup DTMF feature hooks using the channel features datastore
property.
* \since 12.0.0
*
Modified: team/dlee/ASTERISK-22243/include/asterisk/bridge_roles.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/bridge_roles.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/bridge_roles.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/bridge_roles.h Tue Aug 6
09:48:12 2013
@@ -50,6 +50,13 @@
* \param role_name Name of the role being removed
*/
void ast_channel_remove_bridge_role(struct ast_channel *chan, const char
*role_name);
+
+/*!
+ * \brief Removes all bridge roles currently on a channel
+ *
+ * \param chan Channel the roles are being removed from
+ */
+void ast_channel_clear_bridge_roles(struct ast_channel *chan);
/*!
* \brief Set a role option on a channel
Modified: team/dlee/ASTERISK-22243/include/asterisk/frame.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/frame.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/frame.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/frame.h Tue Aug 6 09:48:12 2013
@@ -278,7 +278,11 @@
AST_CONTROL_STREAM_RESTART = 1002, /*!< Indicate to a channel in
playback to restart the stream */
AST_CONTROL_STREAM_REVERSE = 1003, /*!< Indicate to a channel in
playback to rewind */
AST_CONTROL_STREAM_FORWARD = 1004, /*!< Indicate to a channel in
playback to fast forward */
-
+ /* Control frames to manipulate recording on a channel. */
+ AST_CONTROL_RECORD_CANCEL = 1100, /*!< Indicated to a channel in
record to stop recording and discard the file */
+ AST_CONTROL_RECORD_STOP = 1101, /*!< Indicated to a channel in record
to stop recording */
+ AST_CONTROL_RECORD_SUSPEND = 1102, /*!< Indicated to a channel in
record to suspend/unsuspend recording */
+ AST_CONTROL_RECORD_MUTE = 1103, /*!< Indicated to a channel in record
to mute/unmute (i.e. write silence) recording */
};
enum ast_frame_read_action {
Modified: team/dlee/ASTERISK-22243/include/asterisk/stasis.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/stasis.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/stasis.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/stasis.h Tue Aug 6 09:48:12 2013
@@ -110,12 +110,12 @@
* It's a thread safe container, so freely use the stasis_cache_get() and
* stasis_cache_dump() to query the cache.
*
- * The \ref stasis_caching_topic provides a topic that forwards non-cacheable
- * messages unchanged. A cacheable message is wrapped in a \ref
- * stasis_cache_update message which provides the old snapshot (or \c NULL if
- * this is a new cache entry), and the new snapshot (or \c NULL if the entry
was
- * removed from the cache). A stasis_cache_clear_create() message must be sent
- * to the topic in order to remove entries from the cache.
+ * The \ref stasis_caching_topic discards non-cacheable messages. A cacheable
+ * message is wrapped in a \ref stasis_cache_update message which provides the
+ * old snapshot (or \c NULL if this is a new cache entry), and the new snapshot
+ * (or \c NULL if the entry was removed from the cache). A
+ * stasis_cache_clear_create() message must be sent to the topic in order to
+ * remove entries from the cache.
*
* In order to unsubscribe a \ref stasis_caching_topic from the upstream topic,
* call stasis_caching_unsubscribe(). Due to cyclic references, the \ref
Modified: team/dlee/ASTERISK-22243/include/asterisk/stasis_app.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/stasis_app.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/stasis_app.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/stasis_app.h Tue Aug 6 09:48:12
2013
@@ -174,6 +174,24 @@
int stasis_app_control_dial(struct stasis_app_control *control, const char
*endpoint, int timeout);
/*!
+ * \brief Apply a bridge role to a channel controlled by a stasis app control
+ *
+ * \param control Control for \c res_stasis
+ * \param role Role to apply
+ *
+ * \return 0 for success
+ * \return -1 for error.
+ */
+int stasis_app_control_add_role(struct stasis_app_control *control, const char
*role);
+
+/*!
+ * \brief Clear bridge roles currently applied to a channel controlled by a
stasis app control
+ *
+ * \param control Control for \c res_stasis
+ */
+void stasis_app_control_clear_roles(struct stasis_app_control *control);
+
+/*!
* \brief Exit \c res_stasis and continue execution in the dialplan.
*
* If the channel is no longer in \c res_stasis, this function does nothing.
Modified: team/dlee/ASTERISK-22243/include/asterisk/stasis_app_recording.h
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/include/asterisk/stasis_app_recording.h?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/include/asterisk/stasis_app_recording.h (original)
+++ team/dlee/ASTERISK-22243/include/asterisk/stasis_app_recording.h Tue Aug 6
09:48:12 2013
@@ -44,14 +44,30 @@
STASIS_APP_RECORDING_STATE_PAUSED,
/*! The media has stopped recording */
STASIS_APP_RECORDING_STATE_COMPLETE,
- /*! The media has stopped playing */
+ /*! The media has stopped recording, with error */
STASIS_APP_RECORDING_STATE_FAILED,
+ /*! The media has stopped recording, discard the recording file */
+ STASIS_APP_RECORDING_STATE_CANCELED,
+ /*! Sentinel */
+ STASIS_APP_RECORDING_STATE_MAX,
};
/*! Valid operation for controlling a recording. */
enum stasis_app_recording_media_operation {
- /*! Stop the recording operation. */
+ /*! Stop the recording, deleting the media file(s) */
+ STASIS_APP_RECORDING_CANCEL,
+ /*! Stop the recording. */
STASIS_APP_RECORDING_STOP,
+ /*! Pause the recording */
+ STASIS_APP_RECORDING_PAUSE,
+ /*! Unpause the recording */
+ STASIS_APP_RECORDING_UNPAUSE,
+ /*! Mute the recording (record silence) */
+ STASIS_APP_RECORDING_MUTE,
+ /*! Unmute the recording */
+ STASIS_APP_RECORDING_UNMUTE,
+ /*! Sentinel */
+ STASIS_APP_RECORDING_OPER_MAX,
};
#define STASIS_APP_RECORDING_TERMINATE_INVALID 0
Modified: team/dlee/ASTERISK-22243/main/app.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/main/app.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/main/app.c (original)
+++ team/dlee/ASTERISK-22243/main/app.c Tue Aug 6 09:48:12 2013
@@ -1145,6 +1145,78 @@
return d;
}
+/*!
+ * \brief Construct a silence frame of the same duration as \a orig.
+ *
+ * The \a orig frame must be \ref AST_FORMAT_SLINEAR.
+ *
+ * \param orig Frame as basis for silence to generate.
+ * \return New frame of silence; free with ast_frfree().
+ * \return \c NULL on error.
+ */
+static struct ast_frame *make_silence(const struct ast_frame *orig)
+{
+ struct ast_frame *silence;
+ size_t size;
+ size_t datalen;
+ size_t samples = 0;
+ struct ast_frame *next;
+
+ if (!orig) {
+ return NULL;
+ }
+
+ if (orig->subclass.format.id != AST_FORMAT_SLINEAR) {
+ ast_log(LOG_WARNING, "Attempting to silence non-slin frame\n");
+ return NULL;
+ }
+
+ for (next = AST_LIST_NEXT(orig, frame_list);
+ orig;
+ orig = next, next = orig ? AST_LIST_NEXT(orig, frame_list) :
NULL) {
+ samples += orig->samples;
+ }
+
+ ast_verb(4, "Silencing %zd samples\n", samples);
+
+
+ datalen = sizeof(short) * samples;
+ size = sizeof(*silence) + datalen;
+ silence = ast_calloc(1, size);
+ if (!silence) {
+ return NULL;
+ }
+
+ silence->mallocd = AST_MALLOCD_HDR;
+ silence->frametype = AST_FRAME_VOICE;
+ silence->data.ptr = (void *)(silence + 1);
+ silence->samples = samples;
+ silence->datalen = datalen;
+
+ ast_format_set(&silence->subclass.format, AST_FORMAT_SLINEAR, 0);
+
+ return silence;
+}
+
+/*!
+ * \brief Sets a channel's read format to \ref AST_FORMAT_SLINEAR, recording
+ * its original format.
+ *
+ * \param chan Channel to modify.
+ * \param[out] orig_format Output variable to store channel's original read
+ * format.
+ * \return 0 on success.
+ * \return -1 on error.
+ */
+static int set_read_to_slin(struct ast_channel *chan, struct ast_format
*orig_format)
+{
+ if (!chan || !orig_format) {
+ return -1;
+ }
+ ast_format_copy(orig_format, ast_channel_readformat(chan));
+ return ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+}
+
static int global_silence_threshold = 128;
static int global_maxsilence = 0;
@@ -1274,8 +1346,7 @@
return -1;
}
ast_dsp_set_threshold(sildet, silencethreshold);
- ast_format_copy(&rfmt, ast_channel_readformat(chan));
- res = ast_set_read_format_by_id(chan, AST_FORMAT_SLINEAR);
+ res = set_read_to_slin(chan, &rfmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode,
giving up\n");
ast_dsp_free(sildet);
@@ -1293,9 +1364,15 @@
}
if (x == fmtcnt) {
- /* Loop forever, writing the packets we read to the writer(s),
until
- we read a digit or get a hangup */
+ /* Loop, writing the packets we read to the writer(s), until
+ * we have reason to stop. */
struct ast_frame *f;
+ int paused = 0;
+ int muted = 0;
+ time_t pause_start = 0;
+ int paused_secs = 0;
+ int pausedsilence = 0;
+
for (;;) {
if (!(res = ast_waitfor(chan, 2000))) {
ast_debug(1, "One waitfor failed, trying
another\n");
@@ -1315,11 +1392,29 @@
}
if (f->frametype == AST_FRAME_VOICE) {
/* write each format */
- for (x = 0; x < fmtcnt; x++) {
- if (prepend && !others[x]) {
- break;
+ if (paused) {
+ /* It's all good */
+ res = 0;
+ } else {
+ RAII_VAR(struct ast_frame *, silence,
NULL, ast_frame_dtor);
+ struct ast_frame *orig = f;
+
+ if (muted) {
+ silence = make_silence(orig);
+ if (!silence) {
+ ast_log(LOG_WARNING,
+ "Error creating
silence\n");
+ break;
+ }
+ f = silence;
}
- res = ast_writestream(others[x], f);
+ for (x = 0; x < fmtcnt; x++) {
+ if (prepend && !others[x]) {
+ break;
+ }
+ res =
ast_writestream(others[x], f);
+ }
+ f = orig;
}
/* Silence Detection */
@@ -1330,6 +1425,17 @@
totalsilence += olddspsilence;
}
olddspsilence = dspsilence;
+
+ if (paused) {
+ /* record how much silence
there was while we are paused */
+ pausedsilence = dspsilence;
+ } else if (dspsilence > pausedsilence) {
+ /* ignore the paused silence */
+ dspsilence -= pausedsilence;
+ } else {
+ /* dspsilence has reset, reset
pausedsilence */
+ pausedsilence = 0;
+ }
if (dspsilence > maxsilence) {
/* Ended happily with silence */
@@ -1362,15 +1468,51 @@
break;
}
if (strchr(canceldtmf, f->subclass.integer)) {
- ast_verb(3, "User cancelled message by
pressing %c\n", f->subclass.integer);
+ ast_verb(3, "User canceled message by
pressing %c\n", f->subclass.integer);
res = f->subclass.integer;
outmsg = 0;
break;
}
- }
- if (maxtime) {
+ } else if (f->frametype == AST_FRAME_CONTROL) {
+ if (f->subclass.integer ==
AST_CONTROL_RECORD_CANCEL) {
+ ast_verb(3, "Message canceled by
control\n");
+ outmsg = 0; /* cancels the recording */
+ res = 0;
+ break;
+ } else if (f->subclass.integer ==
AST_CONTROL_RECORD_STOP) {
+ ast_verb(3, "Message ended by
control\n");
+ res = 0;
+ break;
+ } else if (f->subclass.integer ==
AST_CONTROL_RECORD_SUSPEND) {
+ paused = !paused;
+ ast_verb(3, "Message %spaused by
control\n",
+ paused ? "" : "un");
+ if (paused) {
+ pause_start = time(NULL);
+ } else {
+ paused_secs += time(NULL) -
pause_start;
+ }
+ } else if (f->subclass.integer ==
AST_CONTROL_RECORD_MUTE) {
+ muted = !muted;
+ ast_verb(3, "Message %smuted by
control\n",
+ muted ? "" : "un");
+ /* We can only silence slin frames, so
+ * set the mode, if we haven't already
+ * for sildet
+ */
+ if (muted && !rfmt.id) {
+ ast_verb(3, "Setting read
format to linear mode\n");
+ res = set_read_to_slin(chan,
&rfmt);
+ if (res < 0) {
+ ast_log(LOG_WARNING,
"Unable to set to linear mode, giving up\n");
+ break;
+ }
+ }
+ }
+ }
+ if (maxtime && !paused) {
end = time(NULL);
- if (maxtime < (end - start)) {
+ if (maxtime < (end - start - paused_secs)) {
ast_verb(3, "Took too long, cutting it
short...\n");
res = 't';
outmsg = 2;
@@ -1493,9 +1635,9 @@
static const char default_acceptdtmf[] = "#";
static const char default_canceldtmf[] = "";
-int ast_play_and_record_full(struct ast_channel *chan, const char *playfile,
const char *recordfile, int maxtime, const char *fmt, int *duration, int
*sound_duration, int silencethreshold, int maxsilence, const char *path, const
char *acceptdtmf, const char *canceldtmf, int skip_confirmation_sound, enum
ast_record_if_exists if_exists)
-{
- return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt,
duration, sound_duration, 0, silencethreshold, maxsilence, path, 0,
S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf),
skip_confirmation_sound, if_exists);
+int ast_play_and_record_full(struct ast_channel *chan, const char *playfile,
const char *recordfile, int maxtime, const char *fmt, int *duration, int
*sound_duration, int beep, int silencethreshold, int maxsilence, const char
*path, const char *acceptdtmf, const char *canceldtmf, int
skip_confirmation_sound, enum ast_record_if_exists if_exists)
+{
+ return __ast_play_and_record(chan, playfile, recordfile, maxtime, fmt,
duration, sound_duration, beep, silencethreshold, maxsilence, path, 0,
S_OR(acceptdtmf, default_acceptdtmf), S_OR(canceldtmf, default_canceldtmf),
skip_confirmation_sound, if_exists);
}
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const
char *recordfile, int maxtime, const char *fmt, int *duration, int
*sound_duration, int silencethreshold, int maxsilence, const char *path)
Modified: team/dlee/ASTERISK-22243/main/bridge_basic.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/main/bridge_basic.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/main/bridge_basic.c (original)
+++ team/dlee/ASTERISK-22243/main/bridge_basic.c Tue Aug 6 09:48:12 2013
@@ -220,7 +220,7 @@
return dtmf_features_flags_to_string(&held_copy, buffer, buf_size);
}
-int ast_bridge_features_ds_set(struct ast_channel *chan, struct ast_flags
*flags)
+static int bridge_features_ds_set_full(struct ast_channel *chan, struct
ast_flags *flags, int replace)
{
struct ast_datastore *datastore;
struct ast_flags *ds_flags;
@@ -228,7 +228,12 @@
datastore = ast_channel_datastore_find(chan, &dtmf_features_info, NULL);
if (datastore) {
ds_flags = datastore->data;
- *ds_flags = *flags;
+ if (replace) {
+ *ds_flags = *flags;
+ } else {
+ flags->flags = flags->flags | ds_flags->flags;
+ *ds_flags = *flags;
+ }
return 0;
}
@@ -247,6 +252,16 @@
datastore->data = ds_flags;
ast_channel_datastore_add(chan, datastore);
return 0;
+}
+
+int ast_bridge_features_ds_set(struct ast_channel *chan, struct ast_flags
*flags)
+{
+ return bridge_features_ds_set_full(chan, flags, 1);
+}
+
+int ast_bridge_features_ds_append(struct ast_channel *chan, struct ast_flags
*flags)
+{
+ return bridge_features_ds_set_full(chan, flags, 0);
}
struct ast_flags *ast_bridge_features_ds_get(struct ast_channel *chan)
Modified: team/dlee/ASTERISK-22243/main/bridge_roles.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/main/bridge_roles.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/main/bridge_roles.c (original)
+++ team/dlee/ASTERISK-22243/main/bridge_roles.c Tue Aug 6 09:48:12 2013
@@ -355,6 +355,25 @@
ast_debug(2, "Role %s did not exist on channel %s\n", role_name,
ast_channel_name(chan));
}
+void ast_channel_clear_bridge_roles(struct ast_channel *chan)
+{
+ struct bridge_roles_datastore *roles_datastore =
fetch_bridge_roles_datastore(chan);
+ struct bridge_role *role;
+
+ if (!roles_datastore) {
+ /* The roles datastore didn't already exist, so there is no
need to remove any roles */
+ ast_debug(2, "Roles did not exist on channel %s\n",
ast_channel_name(chan));
+ return;
+ }
+
+ AST_LIST_TRAVERSE_SAFE_BEGIN(&roles_datastore->role_list, role, list) {
+ ast_debug(2, "Removing bridge role %s from channel %s\n",
role->role, ast_channel_name(chan));
+ AST_LIST_REMOVE_CURRENT(list);
+ bridge_role_destroy(role);
+ }
+ AST_LIST_TRAVERSE_SAFE_END;
+}
+
int ast_channel_set_bridge_role_option(struct ast_channel *channel, const char
*role_name, const char *option, const char *value)
{
struct bridge_role *role = get_role_from_channel(channel, role_name);
Modified: team/dlee/ASTERISK-22243/main/cdr.c
URL:
http://svnview.digium.com/svn/asterisk/team/dlee/ASTERISK-22243/main/cdr.c?view=diff&rev=396335&r1=396334&r2=396335
==============================================================================
--- team/dlee/ASTERISK-22243/main/cdr.c (original)
+++ team/dlee/ASTERISK-22243/main/cdr.c Tue Aug 6 09:48:12 2013
@@ -699,8 +699,8 @@
AST_LIST_TRAVERSE(from_list, variables, entries) {
if (variables &&
(var = ast_var_name(variables)) && (val =
ast_var_value(variables)) &&
- !ast_strlen_zero(var) && !ast_strlen_zero(val)) {
- newvariable = ast_var_assign(var, val);
+ !ast_strlen_zero(var) && !ast_strlen_zero(val) &&
+ (newvariable = ast_var_assign(var, val))) {
AST_LIST_INSERT_HEAD(to_list, newvariable, entries);
x++;
}
@@ -1033,8 +1033,7 @@
}
AST_LIST_TRAVERSE_SAFE_END;
- if (value) {
- newvariable = ast_var_assign(name, value);
+ if (value && (newvariable = ast_var_assign(name, value))) {
AST_LIST_INSERT_HEAD(headp, newvariable, entries);
}
}
@@ -1113,15 +1112,15 @@
copy_variables(&cdr_copy->varshead, &it_cdr->party_a.variables);
AST_LIST_TRAVERSE(&it_cdr->party_b.variables, it_var, entries) {
int found = 0;
+ struct ast_var_t *newvariable;
AST_LIST_TRAVERSE(&cdr_copy->varshead, it_copy_var,
entries) {
if (!strcmp(ast_var_name(it_var),
ast_var_name(it_copy_var))) {
found = 1;
break;
}
}
- if (!found) {
- AST_LIST_INSERT_TAIL(&cdr_copy->varshead,
ast_var_assign(ast_var_name(it_var),
- ast_var_value(it_var)),
entries);
+ if (!found && (newvariable =
ast_var_assign(ast_var_name(it_var), ast_var_value(it_var)))) {
+ AST_LIST_INSERT_TAIL(&cdr_copy->varshead,
newvariable, entries);
}
}
@@ -1482,8 +1481,9 @@
while ((cand_cdr_master = ao2_iterator_next(it_cdrs))) {
struct cdr_object *cand_cdr;
-
- ao2_lock(cand_cdr_master);
+ RAII_VAR(struct cdr_object *, cdr_cleanup, cand_cdr_master,
ao2_cleanup);
+ SCOPED_AO2LOCK(lock, cand_cdr_master);
+
for (cand_cdr = cand_cdr_master; cand_cdr; cand_cdr =
cand_cdr->next) {
/* Skip any records that are not in a bridge or in this
bridge.
* I'm not sure how that would happen, but it pays to
be careful. */
@@ -1499,8 +1499,6 @@
success = 1;
break;
}
- ao2_unlock(cand_cdr_master);
- ao2_t_ref(cand_cdr_master, -1, "Drop iterator reference");
}
ao2_iterator_destroy(it_cdrs);
@@ -1625,8 +1623,9 @@
while ((cand_cdr_master = ao2_iterator_next(it_cdrs))) {
struct cdr_object *cand_cdr;
-
- ao2_lock(cand_cdr_master);
+ RAII_VAR(struct cdr_object *, cdr_cleanup, cand_cdr_master,
ao2_cleanup);
+ SCOPED_AO2LOCK(lock, cand_cdr_master);
+
for (cand_cdr = cand_cdr_master; cand_cdr; cand_cdr =
cand_cdr->next) {
/* Skip any records that are not in a bridge or in this
bridge.
* I'm not sure how that would happen, but it pays to
be careful. */
@@ -1655,8 +1654,6 @@
success = 1;
break;
}
- ao2_unlock(cand_cdr_master);
- ao2_t_ref(cand_cdr_master, -1, "Drop iterator reference");
}
ao2_iterator_destroy(it_cdrs);
@@ -2277,7 +2274,7 @@
ao2_cleanup(candidates);
return NULL;
}
- while ((cand_cdr_master = ao2_iterator_next(it_cdrs))) {
+ for (; (cand_cdr_master = ao2_iterator_next(it_cdrs));
ao2_cleanup(cand_cdr_master)) {
SCOPED_AO2LOCK(lock, cand_cdr_master);
add_candidate_for_bridge(bridge->uniqueid, candidates,
cand_cdr_master, 1);
}
@@ -2291,7 +2288,7 @@
ao2_cleanup(candidates);
return NULL;
}
- while ((cand_cdr_master = ao2_iterator_next(it_cdrs))) {
+ for (; (cand_cdr_master = ao2_iterator_next(it_cdrs));
ao2_cleanup(cand_cdr_master)) {
SCOPED_AO2LOCK(lock, cand_cdr_master);
add_candidate_for_bridge(bridge->uniqueid, candidates,
cand_cdr_master, 0);
}
@@ -2861,7 +2858,7 @@
return -1;
}
- while ((cdr = ao2_iterator_next(it_cdrs))) {
+ for (; (cdr = ao2_iterator_next(it_cdrs)); ao2_unlock(cdr),
ao2_cleanup(cdr)) {
ao2_lock(cdr);
for (it_cdr = cdr; it_cdr; it_cdr = it_cdr->next) {
struct varshead *headp = NULL;
@@ -2877,8 +2874,6 @@
set_variable(headp, name, value);
}
}
- ao2_unlock(cdr);
- ao2_ref(cdr, -1);
}
ao2_iterator_destroy(it_cdrs);
@@ -3598,7 +3593,7 @@
ast_cli(a->fd, TITLE_STRING, "Channel", "Dst. Channel", "LastApp",
"Start", "Answer", "End", "Billsec", "Duration");
[... 1408 lines stripped ...]
--
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