[asterisk-commits] mjordan: trunk r407085 - in /trunk: ./ apps/ main/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Jan 31 17:40:54 CST 2014
Author: mjordan
Date: Fri Jan 31 17:40:51 2014
New Revision: 407085
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=407085
Log:
CDRs: fix a variety of dial status problems, h/hangup handler creating CDRs
This patch fixes a number of small-ish problems that were noticed when
witnessing the records that the FreePBX dialplan produces:
(1) Mid-call events (as well as privacy options) have the ability to change the
overall state of the Dial operation after the called party answers. This
means that publishing the DialEnd event when the called party is premature;
we have to wait for the execution of these subroutines to complete before
we can signal the overall status of the DialEnd. This patch moves that
publication and adds handlers for the mid-call events.
(2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto
datastore is detected. This flag was preventing CDRs from being recorded
for all outbound channels that had a 'continue' option enabled on them by
the Dial application.
(3) The CDR engine now locks the 'Dial' application as being the CDR
application if it detects that the current CDR has entered that app. This
is similar to the logic that is done for Parking. In general, if we entered
into Dial, then we want that CDR to record the application as such - this
prevents pre-dial handlers, mid-call handlers, and other shenaniganry
from changing the application value.
(4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places
to determine if the channel is in hangup logic or dead. In either case, we
don't want to record changes in the channel.
(5) The default option for "endbeforehexten" has been changed to "yes". In
general, you don't want to see CDRs in the 'h' exten or in hangup logic.
Since the semantics of that option changed in 12, it made sense to update
the default value as well.
(6) Finally, because we now have the ability to synchronize on the messages
published to the CDR topic, on shutdown the CDR engine will now synchronize
to the messages currently in flight. This helps to ensure that all
in-flight CDRs are written before shutting down.
(closes issue ASTERISK-23164)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3154
........
Merged revisions 407084 from http://svn.asterisk.org/svn/asterisk/branches/12
Modified:
trunk/ (props changed)
trunk/UPGRADE.txt
trunk/apps/app_dial.c
trunk/main/bridge_after.c
trunk/main/cdr.c
trunk/main/manager_channels.c
trunk/main/pbx.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-12-merged' - no diff available.
Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Fri Jan 31 17:40:51 2014
@@ -22,7 +22,8 @@
===========================================================
From 12 to 13:
-* The per console verbose level feature as previously implemented caused a
+
+- The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console verbose
@@ -30,6 +31,13 @@
console even though it may appear to only affect the current console. If
an older version of rasterisk connects to the new version then the
"core set verbose" command will have no effect.
+
+CDRs:
+ - The "endbeforehexten" setting now defaults to "yes", instead of "no".
+ When set to "no", yhis setting will cause a new CDR to be generated when a
+ channel enters into hangup logic (either the 'h' extension or a hangup
+ handler subroutine). In general, this is not the preferred default: this
+ causes extra CDRs to be generated for a channel in many common dialplans.
CLI commands:
- "core show settings" now lists the current console verbosity in addition
@@ -51,25 +59,25 @@
instead of a "basic" one.
Realtime Configuration:
- * New columns have been added to realtime tables for 'support_path' on
+ - New columns have been added to realtime tables for 'support_path' on
ps_registrations and ps_aors and for 'path' on ps_contacts for the new
SIP Path support in chan_pjsip.
- * The following new tables have been added for pjsip realtime: 'ps_systems',
+ - The following new tables have been added for pjsip realtime: 'ps_systems',
'ps_globals', 'ps_tranports', 'ps_registrations'.
- * The following columns were added to the 'ps_aors' realtime table:
+ - The following columns were added to the 'ps_aors' realtime table:
'maximum_expiration', 'outbound_proxy', and 'support_path'.
- * The following columns were added to the 'ps_contacts' realtime table:
+ - The following columns were added to the 'ps_contacts' realtime table:
'outbound_proxy' and 'path'.
- * New columns have been added to the ps_endpoints realtime table for the
+ - New columns have been added to the ps_endpoints realtime table for the
'media_address', 'redirect_method' and 'set_var' options. Also the
'mwi_fromuser' column was renamed to 'mwi_from_user'.
- * WARNING: The database migration script that adds the 'extensions' table for
+ - WARNING: The database migration script that adds the 'extensions' table for
realtime had to be modified due to an error when installing for MySQL. The
'extensions' table's 'id' column was changed to be a primary key. This could
potentially cause a migration problem. If so, it may be necessary to
manually alter the affected table/column to bring it back in line with the
migration scripts.
- * A new column was added to the 'ps_globals' realtime table for the 'debug'
+ - A new column was added to the 'ps_globals' realtime table for the 'debug'
option.
Modified: trunk/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_dial.c?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Fri Jan 31 17:40:51 2014
@@ -1285,7 +1285,10 @@
}
}
peer = c;
- ast_channel_publish_dial(in, peer, NULL, "ANSWER");
+ /* Inform everyone else that they've been canceled.
+ * The dial end event for the peer will be sent out after
+ * other Dial options have been handled.
+ */
publish_dial_end_event(in, out_chans, peer, "CANCEL");
ast_copy_flags64(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
@@ -2739,6 +2742,7 @@
}
if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
+ ast_channel_publish_dial(chan, peer, NULL, pa.status);
res = 0;
goto out;
}
@@ -2836,6 +2840,7 @@
if (continue_exec)
*continue_exec = 1;
res = 0;
+ ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
goto done;
}
@@ -2848,7 +2853,6 @@
ast_channel_exten_set(peer, ast_channel_exten(chan));
ast_channel_unlock(peer);
ast_channel_unlock(chan);
-
ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
res = ast_app_exec_macro(chan, peer, opt_args[OPT_ARG_CALLEE_MACRO]);
@@ -2889,6 +2893,7 @@
ast_set_flag64(peerflags, OPT_GO_ON);
}
}
+ ast_channel_publish_dial(chan, peer, NULL, macro_result);
} else {
ast_channel_unlock(peer);
}
@@ -2971,6 +2976,7 @@
ast_set_flag64(peerflags, OPT_GO_ON);
}
}
+ ast_channel_publish_dial(chan, peer, NULL, gosub_result);
} else {
ast_channel_unlock(peer);
ast_channel_unlock(chan);
@@ -2978,6 +2984,12 @@
}
if (!res) {
+
+ /* None of the Dial options changed our status; inform
+ * everyone that this channel answered
+ */
+ ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
+
if (!ast_tvzero(calldurationlimit)) {
struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
ast_channel_lock(peer);
Modified: trunk/main/bridge_after.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/bridge_after.c?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/main/bridge_after.c (original)
+++ trunk/main/bridge_after.c Fri Jan 31 17:40:51 2014
@@ -477,6 +477,9 @@
ast_pbx_h_exten_run(chan, after_bridge->context);
}
} else if (!ast_check_hangup(chan)) {
+ /* Clear the outgoing flag */
+ ast_clear_flag(ast_channel_flags(chan), AST_FLAG_OUTGOING);
+
if (after_bridge->specific) {
goto_failed = ast_explicit_goto(chan, after_bridge->context,
after_bridge->exten, after_bridge->priority);
Modified: trunk/main/cdr.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/cdr.c?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/main/cdr.c (original)
+++ trunk/main/cdr.c Fri Jan 31 17:40:51 2014
@@ -129,7 +129,7 @@
channel. Any statistics are gathered from this new CDR. By enabling
this option, no new CDR is created for the dialplan logic that is
executed in <literal>h</literal> extensions or attached hangup handler
- subroutines. The default value is <literal>no</literal>, indicating
+ subroutines. The default value is <literal>yes</literal>, indicating
that a CDR will be generated during hangup logic.</para>
</description>
</configOption>
@@ -203,7 +203,7 @@
#define DEFAULT_BATCHMODE "0"
#define DEFAULT_UNANSWERED "0"
#define DEFAULT_CONGESTION "0"
-#define DEFAULT_END_BEFORE_H_EXTEN "0"
+#define DEFAULT_END_BEFORE_H_EXTEN "1"
#define DEFAULT_INITIATED_SECONDS "0"
#define DEFAULT_BATCH_SIZE "100"
@@ -346,6 +346,9 @@
/*! \brief The parent topic for all topics we want to aggregate for CDRs */
static struct stasis_topic *cdr_topic;
+
+/*! \brief A message type used to synchronize with the CDR topic */
+STASIS_MESSAGE_TYPE_DEFN_LOCAL(cdr_sync_message_type);
struct cdr_object;
@@ -1089,7 +1092,7 @@
/* Don't create records for CDRs where the party A was a dialed channel */
if (snapshot_is_dialed(it_cdr->party_a.snapshot) && !it_cdr->party_b.snapshot) {
- CDR_DEBUG(mod_cfg, "%p - %s is dialed and has no Party B; discarding\n", it_cdr,
+ ast_debug(1, "CDR for %s is dialed and has no Party B; discarding\n",
it_cdr->party_a.snapshot->name);
continue;
}
@@ -1347,7 +1350,14 @@
RAII_VAR(struct module_config *, mod_cfg, ao2_global_obj_ref(module_configs), ao2_cleanup);
ast_assert(strcasecmp(snapshot->name, cdr->party_a.snapshot->name) == 0);
+
cdr_object_swap_snapshot(&cdr->party_a, snapshot);
+ /* Ignore any snapshots from a dead or dying channel */
+ if (ast_test_flag(&snapshot->softhangup_flags, AST_SOFTHANGUP_HANGUP_EXEC)
+ && ast_test_flag(&mod_cfg->general->settings, CDR_END_BEFORE_H_EXTEN)) {
+ cdr_object_check_party_a_hangup(cdr);
+ return 0;
+ }
/* When Party A is originated to an application and the application exits, the stack
* will attempt to clear the application and restore the dummy originate application
@@ -1359,6 +1369,15 @@
&& !ast_test_flag(&cdr->flags, AST_CDR_LOCK_APP)) {
ast_string_field_set(cdr, appl, snapshot->appl);
ast_string_field_set(cdr, data, snapshot->data);
+
+ /* Dial (app_dial) is a special case. Because pre-dial handlers, which
+ * execute before the dial begins, will alter the application/data to
+ * something people typically don't want to see, if we see a channel enter
+ * into Dial here, we set the appl/data accordingly and lock it.
+ */
+ if (!strcmp(snapshot->appl, "Dial")) {
+ ast_set_flag(&cdr->flags, AST_CDR_LOCK_APP);
+ }
}
ast_string_field_set(cdr, linkedid, snapshot->linkedid);
@@ -1434,6 +1453,12 @@
cdr_object_swap_snapshot(&cdr->party_b, peer);
CDR_DEBUG(mod_cfg, "%p - Updated Party B %s snapshot\n", cdr,
cdr->party_b.snapshot->name);
+
+ /* If we have two parties, lock the application that caused the
+ * two parties to be associated. This prevents mid-call event
+ * macros/gosubs from perturbing the CDR application/data
+ */
+ ast_set_flag(&cdr->flags, AST_CDR_LOCK_APP);
} else if (!strcasecmp(cdr->party_a.snapshot->name, peer->name)) {
/* We're the entity being dialed, i.e., outbound origination */
cdr_object_swap_snapshot(&cdr->party_a, peer);
@@ -1822,9 +1847,8 @@
RAII_VAR(struct module_config *, mod_cfg,
ao2_global_obj_ref(module_configs), ao2_cleanup);
- /* If we ignore hangup logic, indicate that we don't need a new CDR */
- if (ast_test_flag(&mod_cfg->general->settings, CDR_END_BEFORE_H_EXTEN)
- && ast_test_flag(&snapshot->softhangup_flags, AST_SOFTHANGUP_HANGUP_EXEC)) {
+ if (ast_test_flag(&snapshot->softhangup_flags, AST_SOFTHANGUP_HANGUP_EXEC)
+ && ast_test_flag(&mod_cfg->general->settings, CDR_END_BEFORE_H_EXTEN)) {
return 0;
}
@@ -1922,7 +1946,7 @@
continue;
}
CDR_DEBUG(mod_cfg, "%p - Processing Dial Begin message for channel %s, peer %s\n",
- cdr,
+ it_cdr,
caller ? caller->name : "(none)",
peer ? peer->name : "(none)");
res &= it_cdr->fn_table->process_dial_begin(it_cdr,
@@ -1933,7 +1957,7 @@
continue;
}
CDR_DEBUG(mod_cfg, "%p - Processing Dial End message for channel %s, peer %s\n",
- cdr,
+ it_cdr,
caller ? caller->name : "(none)",
peer ? peer->name : "(none)");
it_cdr->fn_table->process_dial_end(it_cdr,
@@ -1999,11 +2023,10 @@
RAII_VAR(struct module_config *, mod_cfg,
ao2_global_obj_ref(module_configs), ao2_cleanup);
- if (!new_snapshot) {
- return 0;
- }
-
- if (ast_test_flag(&new_snapshot->flags, AST_FLAG_DEAD)) {
+ /* If we're dead, we don't need a new CDR */
+ if (!new_snapshot
+ || (ast_test_flag(&new_snapshot->softhangup_flags, AST_SOFTHANGUP_HANGUP_EXEC)
+ && ast_test_flag(&mod_cfg->general->settings, CDR_END_BEFORE_H_EXTEN))) {
return 0;
}
@@ -2194,6 +2217,7 @@
/* Party A */
ao2_lock(cdr);
for (it_cdr = cdr; it_cdr; it_cdr = it_cdr->next) {
+
if (!it_cdr->fn_table->process_bridge_leave) {
continue;
}
@@ -2560,6 +2584,19 @@
ao2_unlock(cdr);
+}
+
+/*!
+ * \brief Handler for a synchronization message
+ * \param data Passed on
+ * \param sub The stasis subscription for this message callback
+ * \param topic The topic this message was published for
+ * \param message A blank ao2 object
+ * */
+static void handle_cdr_sync_message(void *data, struct stasis_subscription *sub,
+ struct stasis_message *message)
+{
+ return;
}
struct ast_cdr_config *ast_cdr_get_config(void)
@@ -3337,6 +3374,11 @@
new_cdr->fn_table = cdr_obj->fn_table;
ast_string_field_set(new_cdr, bridge, cdr->bridge);
new_cdr->flags = cdr->flags;
+ /* Explicitly clear the AST_CDR_LOCK_APP flag - we want
+ * the application to be changed on the new CDR if the
+ * dialplan demands it
+ */
+ ast_clear_flag(&new_cdr->flags, AST_CDR_LOCK_APP);
/* If there's a Party B, copy it over as well */
if (cdr_obj->party_b.snapshot) {
@@ -4017,6 +4059,8 @@
ao2_cleanup(cdr_topic);
cdr_topic = NULL;
+ STASIS_MESSAGE_TYPE_CLEANUP(cdr_sync_message_type);
+
ao2_callback(active_cdrs_by_channel, OBJ_NODATA, cdr_object_dispatch_all_cb,
NULL);
finalize_batch_mode();
@@ -4120,11 +4164,17 @@
if (!stasis_router) {
return -1;
}
+
+ if (STASIS_MESSAGE_TYPE_INIT(cdr_sync_message_type)) {
+ return -1;
+ }
+
stasis_message_router_add_cache_update(stasis_router, ast_channel_snapshot_type(), handle_channel_cache_message, NULL);
stasis_message_router_add(stasis_router, ast_channel_dial_type(), handle_dial_message, NULL);
stasis_message_router_add(stasis_router, ast_channel_entered_bridge_type(), handle_bridge_enter_message, NULL);
stasis_message_router_add(stasis_router, ast_channel_left_bridge_type(), handle_bridge_leave_message, NULL);
stasis_message_router_add(stasis_router, ast_parked_call_type(), handle_parked_call_message, NULL);
+ stasis_message_router_add(stasis_router, cdr_sync_message_type(), handle_cdr_sync_message, NULL);
active_cdrs_by_channel = ao2_container_alloc(NUM_CDR_BUCKETS,
cdr_object_channel_hash_fn, cdr_object_channel_cmp_fn);
@@ -4149,6 +4199,8 @@
void ast_cdr_engine_term(void)
{
RAII_VAR(struct module_config *, mod_cfg, ao2_global_obj_ref(module_configs), ao2_cleanup);
+ RAII_VAR(void *, payload, ao2_alloc(sizeof(*payload), NULL), ao2_cleanup);
+ RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
/* Since this is called explicitly during process shutdown, we might not have ever
* been initialized. If so, the config object will be NULL.
@@ -4156,10 +4208,22 @@
if (!mod_cfg) {
return;
}
- if (!ast_test_flag(&mod_cfg->general->settings, CDR_BATCHMODE)) {
+
+ /* Make sure we have the needed items */
+ if (!stasis_router || !payload) {
return;
}
- cdr_submit_batch(ast_test_flag(&mod_cfg->general->batch_settings.settings, BATCH_MODE_SAFE_SHUTDOWN));
+
+ ast_debug(1, "CDR Engine termination request received; waiting on messages...\n");
+
+ message = stasis_message_create(cdr_sync_message_type(), payload);
+ if (message) {
+ stasis_message_router_publish_sync(stasis_router, message);
+ }
+
+ if (ast_test_flag(&mod_cfg->general->settings, CDR_BATCHMODE)) {
+ cdr_submit_batch(ast_test_flag(&mod_cfg->general->batch_settings.settings, BATCH_MODE_SAFE_SHUTDOWN));
+ }
}
int ast_cdr_engine_reload(void)
Modified: trunk/main/manager_channels.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/manager_channels.c?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/main/manager_channels.c (original)
+++ trunk/main/manager_channels.c Fri Jan 31 17:40:51 2014
@@ -151,12 +151,38 @@
<parameter name="DialStatus">
<para>The result of the dial operation.</para>
<enumlist>
- <enum name="ANSWER" />
- <enum name="BUSY" />
- <enum name="CANCEL" />
- <enum name="CHANUNAVAIL" />
- <enum name="CONGESTION" />
- <enum name="NOANSWER" />
+ <enum name="ABORT">
+ <para>The call was aborted.</para>
+ </enum>
+ <enum name="ANSWER">
+ <para>The caller answered.</para>
+ </enum>
+ <enum name="BUSY">
+ <para>The caller was busy.</para>
+ </enum>
+ <enum name="CANCEL">
+ <para>The caller cancelled the call.</para>
+ </enum>
+ <enum name="CHANUNAVAIL">
+ <para>The requested channel is unavailable.</para>
+ </enum>
+ <enum name="CONGESTION">
+ <para>The called party is congested.</para>
+ </enum>
+ <enum name="CONTINUE">
+ <para>The dial completed, but the caller elected
+ to continue in the dialplan.</para>
+ </enum>
+ <enum name="GOTO">
+ <para>The dial completed, but the caller jumped to
+ a dialplan location.</para>
+ <para>If known, the location the caller is jumping
+ to will be appended to the result following a
+ ":".</para>
+ </enum>
+ <enum name="NOANSWER">
+ <para>The called party failed to answer.</para>
+ </enum>
</enumlist>
</parameter>
</syntax>
Modified: trunk/main/pbx.c
URL: http://svnview.digium.com/svn/asterisk/trunk/main/pbx.c?view=diff&rev=407085&r1=407084&r2=407085
==============================================================================
--- trunk/main/pbx.c (original)
+++ trunk/main/pbx.c Fri Jan 31 17:40:51 2014
@@ -5914,18 +5914,18 @@
ast_channel_lock(chan);
- /* Set h exten location */
- if (context != ast_channel_context(chan)) {
- ast_channel_context_set(chan, context);
- }
- ast_channel_exten_set(chan, "h");
- ast_channel_priority_set(chan, 1);
-
/*
* Make sure that the channel is marked as hungup since we are
* going to run the h exten on it.
*/
ast_softhangup_nolock(chan, AST_SOFTHANGUP_HANGUP_EXEC);
+
+ /* Set h exten location */
+ if (context != ast_channel_context(chan)) {
+ ast_channel_context_set(chan, context);
+ }
+ ast_channel_exten_set(chan, "h");
+ ast_channel_priority_set(chan, 1);
/* Save autoloop flag */
autoloopflag = ast_test_flag(ast_channel_flags(chan), AST_FLAG_IN_AUTOLOOP);
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