[svn-commits] mjordan: branch oej/oolong-path-support-trunk r381913 - in /team/oej/oolong-p...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Feb 22 21:47:26 CST 2013
Author: mjordan
Date: Fri Feb 22 21:47:21 2013
New Revision: 381913
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381913
Log:
Re-enable automerge
Modified:
team/oej/oolong-path-support-trunk/ (props changed)
team/oej/oolong-path-support-trunk/CHANGES
team/oej/oolong-path-support-trunk/Makefile
team/oej/oolong-path-support-trunk/apps/app_confbridge.c
team/oej/oolong-path-support-trunk/apps/app_dial.c
team/oej/oolong-path-support-trunk/apps/app_skel.c
team/oej/oolong-path-support-trunk/apps/confbridge/conf_config_parser.c
team/oej/oolong-path-support-trunk/apps/confbridge/include/confbridge.h
team/oej/oolong-path-support-trunk/channels/chan_motif.c
team/oej/oolong-path-support-trunk/channels/chan_sip.c
team/oej/oolong-path-support-trunk/channels/chan_skinny.c
team/oej/oolong-path-support-trunk/configs/motif.conf.sample
team/oej/oolong-path-support-trunk/configs/sip.conf.sample
team/oej/oolong-path-support-trunk/configs/skinny.conf.sample
team/oej/oolong-path-support-trunk/configs/xmpp.conf.sample
team/oej/oolong-path-support-trunk/configure
team/oej/oolong-path-support-trunk/configure.ac
team/oej/oolong-path-support-trunk/doc/appdocsxml.dtd
team/oej/oolong-path-support-trunk/funcs/func_presencestate.c
team/oej/oolong-path-support-trunk/include/asterisk/_private.h
team/oej/oolong-path-support-trunk/include/asterisk/config_options.h
team/oej/oolong-path-support-trunk/include/asterisk/logger.h
team/oej/oolong-path-support-trunk/include/asterisk/module.h
team/oej/oolong-path-support-trunk/include/asterisk/sorcery.h
team/oej/oolong-path-support-trunk/include/asterisk/xml.h
team/oej/oolong-path-support-trunk/include/asterisk/xmldoc.h
team/oej/oolong-path-support-trunk/main/asterisk.c
team/oej/oolong-path-support-trunk/main/autoservice.c
team/oej/oolong-path-support-trunk/main/cli.c
team/oej/oolong-path-support-trunk/main/config_options.c
team/oej/oolong-path-support-trunk/main/features.c
team/oej/oolong-path-support-trunk/main/loader.c
team/oej/oolong-path-support-trunk/main/logger.c
team/oej/oolong-path-support-trunk/main/manager.c
team/oej/oolong-path-support-trunk/main/named_acl.c
team/oej/oolong-path-support-trunk/main/presencestate.c
team/oej/oolong-path-support-trunk/main/sorcery.c
team/oej/oolong-path-support-trunk/main/udptl.c
team/oej/oolong-path-support-trunk/main/xml.c
team/oej/oolong-path-support-trunk/main/xmldoc.c
team/oej/oolong-path-support-trunk/res/res_agi.c
team/oej/oolong-path-support-trunk/res/res_sorcery_config.c
team/oej/oolong-path-support-trunk/res/res_sorcery_memory.c
team/oej/oolong-path-support-trunk/res/res_xmpp.c
team/oej/oolong-path-support-trunk/res/snmp/agent.c
team/oej/oolong-path-support-trunk/tests/test_sorcery.c
Propchange: team/oej/oolong-path-support-trunk/
------------------------------------------------------------------------------
automerge = *
Propchange: team/oej/oolong-path-support-trunk/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/oej/oolong-path-support-trunk/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Feb 22 21:47:21 2013
@@ -1,1 +1,1 @@
-/trunk:1-381492
+/trunk:1-381912
Modified: team/oej/oolong-path-support-trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/CHANGES?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/CHANGES (original)
+++ team/oej/oolong-path-support-trunk/CHANGES Fri Feb 22 21:47:21 2013
@@ -161,6 +161,9 @@
* Added announcement configuration option to user profile. If set the sound
file will be played to the user, and only the user, upon joining the
conference bridge.
+
+ * Added record_file_append option that defaults to "yes", but if set to no
+ will create a new file between each start/stop recording.
Dial
Modified: team/oej/oolong-path-support-trunk/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/Makefile?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/Makefile (original)
+++ team/oej/oolong-path-support-trunk/Makefile Fri Feb 22 21:47:21 2013
@@ -456,7 +456,7 @@
@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
@for x in $(MOD_SUBDIRS); do \
printf "$$x " ; \
- for i in $$x/*.c; do \
+ for i in `find $$x -name *.c`; do \
$(AWK) -f build_tools/get_documentation $$i >> $@ ; \
done ; \
done
Modified: team/oej/oolong-path-support-trunk/apps/app_confbridge.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/apps/app_confbridge.c?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/apps/app_confbridge.c (original)
+++ team/oej/oolong-path-support-trunk/apps/app_confbridge.c Fri Feb 22 21:47:21 2013
@@ -596,6 +596,37 @@
return tmp;
}
+static void set_rec_filename(struct conference_bridge *bridge, struct ast_str **filename)
+{
+ char *rec_file = bridge->b_profile.rec_file;
+ time_t now;
+ char *ext;
+
+ if (ast_str_strlen(*filename) && ast_test_flag(&bridge->b_profile, BRIDGE_OPT_RECORD_FILE_APPEND)) {
+ return;
+ }
+
+ time(&now);
+
+ ast_str_reset(*filename);
+ if (ast_strlen_zero(rec_file)) {
+ ast_str_set(filename, 0, "confbridge-%s-%u.wav", bridge->name, (unsigned int)now);
+ } else {
+ /* insert time before file extension */
+ ext = strrchr(rec_file, '.');
+ if (ext) {
+ ast_str_set_substr(filename, 0, rec_file, ext - rec_file);
+ ast_str_append(filename, 0, "-%u%s", (unsigned int)now, ext);
+ } else {
+ ast_str_set(filename, 0, "%s-%u", rec_file, (unsigned int)now);
+ }
+ }
+
+ if (ast_test_flag(&bridge->b_profile, BRIDGE_OPT_RECORD_FILE_APPEND)) {
+ ast_str_append(filename, 0, ",a");
+ }
+}
+
static void *record_thread(void *obj)
{
struct conference_bridge *conference_bridge = obj;
@@ -614,16 +645,7 @@
/* XXX If we get an EXIT right here, START will essentially be a no-op */
while (conference_bridge->record_state != CONF_RECORD_EXIT) {
- if (!(ast_strlen_zero(conference_bridge->b_profile.rec_file))) {
- ast_str_append(&filename, 0, "%s", conference_bridge->b_profile.rec_file);
- } else {
- time_t now;
- time(&now);
- ast_str_append(&filename, 0, "confbridge-%s-%u.wav",
- conference_bridge->name,
- (unsigned int) now);
- }
-
+ set_rec_filename(conference_bridge, &filename);
chan = ast_channel_ref(conference_bridge->record_chan);
ast_answer(chan);
pbx_exec(chan, mixmonapp, ast_str_buffer(filename));
@@ -753,9 +775,16 @@
*/
static int start_conf_record_thread(struct conference_bridge *conference_bridge)
{
+ conf_start_record(conference_bridge);
+
+ /*
+ * if the thread has already been started, don't start another
+ */
+ if (conference_bridge->record_thread != AST_PTHREADT_NULL) {
+ return 0;
+ }
+
ao2_ref(conference_bridge, +1); /* give the record thread a ref */
-
- conf_start_record(conference_bridge);
if (ast_pthread_create_background(&conference_bridge->record_thread, NULL, record_thread, conference_bridge)) {
ast_log(LOG_WARNING, "Failed to create recording channel for conference %s\n", conference_bridge->name);
@@ -2157,10 +2186,10 @@
struct conference_bridge *bridge = NULL;
char *res = NULL;
int wordlen = strlen(word);
- struct ao2_iterator i;
-
- i = ao2_iterator_init(conference_bridges, 0);
- while ((bridge = ao2_iterator_next(&i))) {
+ struct ao2_iterator iter;
+
+ iter = ao2_iterator_init(conference_bridges, 0);
+ while ((bridge = ao2_iterator_next(&iter))) {
if (!strncasecmp(bridge->name, word, wordlen) && ++which > state) {
res = ast_strdup(bridge->name);
ao2_ref(bridge, -1);
@@ -2168,7 +2197,7 @@
}
ao2_ref(bridge, -1);
}
- ao2_iterator_destroy(&i);
+ ao2_iterator_destroy(&iter);
return res;
}
@@ -2224,9 +2253,31 @@
return CLI_SUCCESS;
}
-static void handle_cli_confbridge_list_item(struct ast_cli_args *a, struct conference_bridge_user *participant)
-{
+static void handle_cli_confbridge_list_item(struct ast_cli_args *a, struct conference_bridge_user *participant, int waiting)
+{
+ char flag_str[5 + 1];/* Max flags + terminator */
+ int pos = 0;
+
+ /* Build flags column string. */
+ if (ast_test_flag(&participant->u_profile, USER_OPT_ADMIN)) {
+ flag_str[pos++] = 'A';
+ }
+ if (ast_test_flag(&participant->u_profile, USER_OPT_MARKEDUSER)) {
+ flag_str[pos++] = 'M';
+ }
+ if (ast_test_flag(&participant->u_profile, USER_OPT_WAITMARKED)) {
+ flag_str[pos++] = 'W';
+ }
+ if (ast_test_flag(&participant->u_profile, USER_OPT_ENDMARKED)) {
+ flag_str[pos++] = 'E';
+ }
+ if (waiting) {
+ flag_str[pos++] = 'w';
+ }
+ flag_str[pos] = '\0';
+
ast_cli(a->fd, "%-29s ", ast_channel_name(participant->chan));
+ ast_cli(a->fd, "%-5s ", flag_str);
ast_cli(a->fd, "%-17s", participant->u_profile.name);
ast_cli(a->fd, "%-17s", participant->b_profile.name);
ast_cli(a->fd, "%-17s", participant->menu_name);
@@ -2236,17 +2287,24 @@
static char *handle_cli_confbridge_list(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
- struct ao2_iterator i;
- struct conference_bridge *bridge = NULL;
- struct conference_bridge tmp;
- struct conference_bridge_user *participant = NULL;
+ struct conference_bridge *bridge;
switch (cmd) {
case CLI_INIT:
e->command = "confbridge list";
e->usage =
"Usage: confbridge list [<name>]\n"
- " Lists all currently active conference bridges.\n";
+ " Lists all currently active conference bridges or a specific conference bridge.\n"
+ "\n"
+ " When a conference bridge name is provided, flags may be shown for users. Below\n"
+ " are the flags and what they represent.\n"
+ "\n"
+ " Flags:\n"
+ " A - The user is an admin\n"
+ " M - The user is a marked user\n"
+ " W - The user must wait for a marked user to join\n"
+ " E - The user will be kicked after the last marked user leaves the conference\n"
+ " w - The user is waiting for a marked user to join\n";
return NULL;
case CLI_GENERATE:
if (a->pos == 2) {
@@ -2256,32 +2314,37 @@
}
if (a->argc == 2) {
+ struct ao2_iterator iter;
+
ast_cli(a->fd, "Conference Bridge Name Users Marked Locked?\n");
ast_cli(a->fd, "================================ ====== ====== ========\n");
- i = ao2_iterator_init(conference_bridges, 0);
- while ((bridge = ao2_iterator_next(&i))) {
+ iter = ao2_iterator_init(conference_bridges, 0);
+ while ((bridge = ao2_iterator_next(&iter))) {
ast_cli(a->fd, "%-32s %6i %6i %s\n", bridge->name, bridge->activeusers + bridge->waitingusers, bridge->markedusers, (bridge->locked ? "locked" : "unlocked"));
ao2_ref(bridge, -1);
}
- ao2_iterator_destroy(&i);
+ ao2_iterator_destroy(&iter);
return CLI_SUCCESS;
}
if (a->argc == 3) {
+ struct conference_bridge_user *participant;
+ struct conference_bridge tmp;
+
ast_copy_string(tmp.name, a->argv[2], sizeof(tmp.name));
bridge = ao2_find(conference_bridges, &tmp, OBJ_POINTER);
if (!bridge) {
ast_cli(a->fd, "No conference bridge named '%s' found!\n", a->argv[2]);
return CLI_SUCCESS;
}
- ast_cli(a->fd, "Channel User Profile Bridge Profile Menu CallerID\n");
- ast_cli(a->fd, "============================= ================ ================ ================ ================\n");
+ ast_cli(a->fd, "Channel Flags User Profile Bridge Profile Menu CallerID\n");
+ ast_cli(a->fd, "============================= ===== ================ ================ ================ ================\n");
ao2_lock(bridge);
AST_LIST_TRAVERSE(&bridge->active_list, participant, list) {
- handle_cli_confbridge_list_item(a, participant);
+ handle_cli_confbridge_list_item(a, participant, 0);
}
AST_LIST_TRAVERSE(&bridge->waiting_list, participant, list) {
- handle_cli_confbridge_list_item(a, participant);
+ handle_cli_confbridge_list_item(a, participant, 1);
}
ao2_unlock(bridge);
ao2_ref(bridge, -1);
@@ -2580,7 +2643,7 @@
.read = func_confbridge_info,
};
-static void action_confbridgelist_item(struct mansession *s, const char *id_text, struct conference_bridge *bridge, struct conference_bridge_user *participant)
+static void action_confbridgelist_item(struct mansession *s, const char *id_text, struct conference_bridge *bridge, struct conference_bridge_user *participant, int waiting)
{
astman_append(s,
"Event: ConfbridgeList\r\n"
@@ -2591,6 +2654,9 @@
"Channel: %s\r\n"
"Admin: %s\r\n"
"MarkedUser: %s\r\n"
+ "WaitMarked: %s\r\n"
+ "EndMarked: %s\r\n"
+ "Waiting: %s\r\n"
"\r\n",
id_text,
bridge->name,
@@ -2598,19 +2664,23 @@
S_COR(ast_channel_caller(participant->chan)->id.name.valid, ast_channel_caller(participant->chan)->id.name.str, "<no name>"),
ast_channel_name(participant->chan),
ast_test_flag(&participant->u_profile, USER_OPT_ADMIN) ? "Yes" : "No",
- ast_test_flag(&participant->u_profile, USER_OPT_MARKEDUSER) ? "Yes" : "No");
+ ast_test_flag(&participant->u_profile, USER_OPT_MARKEDUSER) ? "Yes" : "No",
+ ast_test_flag(&participant->u_profile, USER_OPT_WAITMARKED) ? "Yes" : "No",
+ ast_test_flag(&participant->u_profile, USER_OPT_ENDMARKED) ? "Yes" : "No",
+ waiting ? "Yes" : "No");
}
static int action_confbridgelist(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
const char *conference = astman_get_header(m, "Conference");
- struct conference_bridge_user *participant = NULL;
- struct conference_bridge *bridge = NULL;
+ struct conference_bridge_user *participant;
+ struct conference_bridge *bridge;
struct conference_bridge tmp;
- char id_text[80] = "";
+ char id_text[80];
int total = 0;
+ id_text[0] = '\0';
if (!ast_strlen_zero(actionid)) {
snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", actionid);
}
@@ -2634,11 +2704,11 @@
ao2_lock(bridge);
AST_LIST_TRAVERSE(&bridge->active_list, participant, list) {
total++;
- action_confbridgelist_item(s, id_text, bridge, participant);
+ action_confbridgelist_item(s, id_text, bridge, participant, 0);
}
AST_LIST_TRAVERSE(&bridge->waiting_list, participant, list) {
total++;
- action_confbridgelist_item(s, id_text, bridge, participant);
+ action_confbridgelist_item(s, id_text, bridge, participant, 1);
}
ao2_unlock(bridge);
ao2_ref(bridge, -1);
@@ -2657,7 +2727,7 @@
{
const char *actionid = astman_get_header(m, "ActionID");
struct conference_bridge *bridge = NULL;
- struct ao2_iterator i;
+ struct ao2_iterator iter;
char id_text[512] = "";
int totalitems = 0;
@@ -2673,8 +2743,8 @@
astman_send_listack(s, m, "Confbridge conferences will follow", "start");
/* Traverse the conference list */
- i = ao2_iterator_init(conference_bridges, 0);
- while ((bridge = ao2_iterator_next(&i))) {
+ iter = ao2_iterator_init(conference_bridges, 0);
+ while ((bridge = ao2_iterator_next(&iter))) {
totalitems++;
ao2_lock(bridge);
@@ -2695,7 +2765,7 @@
ao2_ref(bridge, -1);
}
- ao2_iterator_destroy(&i);
+ ao2_iterator_destroy(&iter);
/* Send final confirmation */
astman_append(s,
Modified: team/oej/oolong-path-support-trunk/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/apps/app_dial.c?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/apps/app_dial.c (original)
+++ team/oej/oolong-path-support-trunk/apps/app_dial.c Fri Feb 22 21:47:21 2013
@@ -3096,7 +3096,7 @@
}
ast_channel_early_bridge(chan, NULL);
- hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
+ hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
senddialendevent(chan, pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
Modified: team/oej/oolong-path-support-trunk/apps/app_skel.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/apps/app_skel.c?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/apps/app_skel.c (original)
+++ team/oej/oolong-path-support-trunk/apps/app_skel.c Fri Feb 22 21:47:21 2013
@@ -86,6 +86,51 @@
from. It shows you the basic structure to create your own Asterisk applications.</para>
</description>
</application>
+
+ <configInfo name="app_skel" language="en_US">
+ <configFile name="app_skel.conf">
+ <configObject name="globals">
+ <synopsis>Options that apply globally to app_skel</synopsis>
+ <configOption name="games">
+ <synopsis>The number of games a single execution of SkelGuessNumber will play</synopsis>
+ </configOption>
+ <configOption name="cheat">
+ <synopsis>Should the computer cheat?</synopsis>
+ <description><para>If enabled, the computer will ignore winning guesses.</para></description>
+ </configOption>
+ </configObject>
+ <configObject name="sounds">
+ <synopsis>Prompts for SkelGuessNumber to play</synopsis>
+ <configOption name="prompt" default="please-enter-your&number&queue-less-than">
+ <synopsis>A prompt directing the user to enter a number less than the max number</synopsis>
+ </configOption>
+ <configOption name="wrong_guess" default="vm-pls-try-again">
+ <synopsis>The sound file to play when a wrong guess is made</synopsis>
+ </configOption>
+ <configOption name="right_guess" default="auth-thankyou">
+ <synopsis>The sound file to play when a correct guess is made</synopsis>
+ </configOption>
+ <configOption name="too_low">
+ <synopsis>The sound file to play when a guess is too low</synopsis>
+ </configOption>
+ <configOption name="too_high">
+ <synopsis>The sound file to play when a guess is too high</synopsis>
+ </configOption>
+ <configOption name="lose" default="vm-goodbye">
+ <synopsis>The sound file to play when a player loses</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="level">
+ <synopsis>Defined levels for the SkelGuessNumber game</synopsis>
+ <configOption name="max_number">
+ <synopsis>The maximum in the range of numbers to guess (1 is the implied minimum)</synopsis>
+ </configOption>
+ <configOption name="max_guesses">
+ <synopsis>The maximum number of guesses before a game is considered lost</synopsis>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
***/
static char *app = "SkelGuessNumber";
@@ -197,6 +242,7 @@
/*! \brief An aco_type structure to link the "general" category to the skel_global_config type */
static struct aco_type global_option = {
.type = ACO_GLOBAL,
+ .name = "globals",
.item_offset = offsetof(struct skel_config, global),
.category_match = ACO_WHITELIST,
.category = "^general$",
@@ -207,6 +253,7 @@
/*! \brief An aco_type structure to link the "sounds" category to the skel_global_config type */
static struct aco_type sound_option = {
.type = ACO_GLOBAL,
+ .name = "sounds",
.item_offset = offsetof(struct skel_config, global),
.category_match = ACO_WHITELIST,
.category = "^sounds$",
@@ -217,6 +264,7 @@
/*! \brief An aco_type structure to link the everything but the "general" and "sounds" categories to the skel_level type */
static struct aco_type level_option = {
.type = ACO_ITEM,
+ .name = "level",
.category_match = ACO_BLACKLIST,
.category = "^(general|sounds)$",
.item_alloc = skel_level_alloc,
Modified: team/oej/oolong-path-support-trunk/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/apps/confbridge/conf_config_parser.c?view=diff&rev=381913&r1=381912&r2=381913
==============================================================================
--- team/oej/oolong-path-support-trunk/apps/confbridge/conf_config_parser.c (original)
+++ team/oej/oolong-path-support-trunk/apps/confbridge/conf_config_parser.c Fri Feb 22 21:47:21 2013
@@ -40,6 +40,464 @@
#include "asterisk/stringfields.h"
#include "asterisk/pbx.h"
+
+/*** DOCUMENTATION
+ <configInfo name="app_confbridge" language="en_US">
+ <synopsis>Conference Bridge Application</synopsis>
+ <configFile name="confbridge.conf">
+ <configObject name="global">
+ <synopsis>Unused, but reserved.</synopsis>
+ </configObject>
+ <configObject name="user_profile">
+ <synopsis>A named profile to apply to specific callers.</synopsis>
+ <description><para>Callers in a ConfBridge have a profile associated with them
+ that determine their options. A configuration section is determined to be a
+ user_profile when the <literal>type</literal> parameter has a value
+ of <literal>user</literal>.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Define this configuration category as a user profile.</synopsis>
+ <description><para>The type parameter determines how a context in the
+ configuration file is interpreted.</para>
+ <enumlist>
+ <enum name="user"><para>Configure the context as a <replaceable>user_profile</replaceable></para></enum>
+ <enum name="bridge"><para>Configure the context as a <replaceable>bridge_profile</replaceable></para></enum>
+ <enum name="menu"><para>Configure the context as a <replaceable>menu</replaceable></para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="admin">
+ <synopsis>Sets if the user is an admin or not</synopsis>
+ </configOption>
+ <configOption name="marked">
+ <synopsis>Sets if this is a marked user or not</synopsis>
+ </configOption>
+ <configOption name="startmuted">
+ <synopsis>Sets if all users should start out muted</synopsis>
+ </configOption>
+ <configOption name="music_on_hold_when_empty">
+ <synopsis>Play MOH when user is alone or waiting on a marked user</synopsis>
+ </configOption>
+ <configOption name="quiet">
+ <synopsis>Silence enter/leave prompts and user intros for this user</synopsis>
+ </configOption>
+ <configOption name="announce_user_count">
+ <synopsis>Sets if the number of users should be announced to the user</synopsis>
+ </configOption>
+ <configOption name="announce_user_count_all">
+ <synopsis>Announce user count to all the other users when this user joins</synopsis>
+ <description><para>Sets if the number of users should be announced to all the other users
+ in the conference when this user joins. This option can be either set to 'yes' or
+ a number. When set to a number, the announcement will only occur once the user
+ count is above the specified number.
+ </para></description>
+ </configOption>
+ <configOption name="announce_only_user">
+ <synopsis>Announce to a user when they join an empty conference</synopsis>
+ </configOption>
+ <configOption name="wait_marked">
+ <synopsis>Sets if the user must wait for a marked user to enter before joining a conference</synopsis>
+ </configOption>
+ <configOption name="end_marked">
+ <synopsis>Kick the user from the conference when the last marked user leaves</synopsis>
+ </configOption>
+ <configOption name="talk_detection_events">
+ <synopsis>Set whether or not notifications of when a user begins and ends talking should be sent out as events over AMI</synopsis>
+ </configOption>
+ <configOption name="dtmf_passthrough">
+ <synopsis>Sets whether or not DTMF should pass through the conference</synopsis>
+ </configOption>
+ <configOption name="announce_join_leave">
+ <synopsis>Prompt user for their name when joining a conference and play it to the conference when they enter</synopsis>
+ </configOption>
+ <configOption name="pin">
+ <synopsis>Sets a PIN the user must enter before joining the conference</synopsis>
+ </configOption>
+ <configOption name="music_on_hold_class">
+ <synopsis>The MOH class to use for this user</synopsis>
+ </configOption>
+ <configOption name="announcement">
+ <synopsis>Sound file to play to the user when they join a conference</synopsis>
+ </configOption>
+ <configOption name="denoise">
+ <synopsis>Apply a denoise filter to the audio before mixing</synopsis>
+ <description><para>Sets whether or not a denoise filter should be applied
+ to the audio before mixing or not. Off by default. Requires
+ codec_speex to be built and installed. Do not confuse this option
+ with drop_silence. Denoise is useful if there is a lot of background
+ noise for a user as it attempts to remove the noise while preserving
+ the speech. This option does NOT remove silence from being mixed into
+ the conference and does come at the cost of a slight performance hit.
+ </para></description>
+ </configOption>
+ <configOption name="dsp_drop_silence">
+ <synopsis>Drop what Asterisk detects as silence from audio sent to the bridge</synopsis>
+ <description><para>
+ This option drops what Asterisk detects as silence from
+ entering into the bridge. Enabling this option will drastically
+ improve performance and help remove the buildup of background
+ noise from the conference. Highly recommended for large conferences
+ due to its performance enhancements.
+ </para></description>
+ </configOption>
+ <configOption name="dsp_silence_threshold">
+ <synopsis>The number of milliseconds of detected silence necessary to trigger silence detection</synopsis>
+ <description><para>
+ The time in milliseconds of sound falling within the what
+ the dsp has established as baseline silence before a user
+ is considered be silent. This value affects several
+ operations and should not be changed unless the impact
+ on call quality is fully understood.</para>
+ <para>What this value affects internally:</para>
+ <para>
+ 1. When talk detection AMI events are enabled, this value
+ determines when the user has stopped talking after a
+ period of talking. If this value is set too low
+ AMI events indicating the user has stopped talking
+ may get falsely sent out when the user briefly pauses
+ during mid sentence.
+ </para>
+ <para>
+ 2. The drop_silence option depends on this value to
+ determine when the user's audio should begin to be
+ dropped from the conference bridge after the user
+ stops talking. If this value is set too low the user's
+ audio stream may sound choppy to the other participants.
+ This is caused by the user transitioning constantly from
+ silence to talking during mid sentence.
+ </para>
+ <para>
+ The best way to approach this option is to set it slightly above
+ the maximum amount of ms of silence a user may generate during
+ natural speech.
+ </para>
+ <para>By default this value is 2500ms. Valid values are 1 through 2^31.</para>
+ </description>
+ </configOption>
+ <configOption name="dsp_talking_threshold">
+ <synopsis>The number of milliseconds of detected non-silence necessary to triger talk detection</synopsis>
+ <description><para>
+ The time in milliseconds of sound above what the dsp has
+ established as base line silence for a user before a user
+ is considered to be talking. This value affects several
+ operations and should not be changed unless the impact on
+ call quality is fully understood.</para>
+ <para>
+ What this value affects internally:
+ </para>
+ <para>
+ 1. Audio is only mixed out of a user's incoming audio stream
+ if talking is detected. If this value is set too
+ loose the user will hear themselves briefly each
+ time they begin talking until the dsp has time to
+ establish that they are in fact talking.
+ </para>
+ <para>
+ 2. When talk detection AMI events are enabled, this value
+ determines when talking has begun which results in
+ an AMI event to fire. If this value is set too tight
+ AMI events may be falsely triggered by variants in
+ room noise.
+ </para>
+ <para>
+ 3. The drop_silence option depends on this value to determine
+ when the user's audio should be mixed into the bridge
+ after periods of silence. If this value is too loose
+ the beginning of a user's speech will get cut off as they
+ transition from silence to talking.
+ </para>
+ <para>By default this value is 160 ms. Valid values are 1 through 2^31</para>
+ </description>
+ </configOption>
+ <configOption name="jitterbuffer">
+ <synopsis>Place a jitter buffer on the user's audio stream before audio mixing is performed</synopsis>
+ <description><para>
+ Enabling this option places a jitterbuffer on the user's audio stream
+ before audio mixing is performed. This is highly recommended but will
+ add a slight delay to the audio. This option is using the <literal>JITTERBUFFER</literal>
+ dialplan function's default adaptive jitterbuffer. For a more fine tuned
+ jitterbuffer, disable this option and use the <literal>JITTERBUFFER</literal> dialplan function
+ on the user before entering the ConfBridge application.
+ </para></description>
+ </configOption>
+ <configOption name="template">
+ <synopsis>When using the CONFBRIDGE dialplan function, use a user profile as a template for creating a new temporary profile</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="bridge_profile">
+ <synopsis>A named profile to apply to specific bridges.</synopsis>
+ <description><para>ConfBridge bridges have a profile associated with them
+ that determine their options. A configuration section is determined to be a
+ <literal>bridge_profile</literal> when the <literal>type</literal> parameter has a value
+ of <literal>bridge</literal>.
+ </para></description>
+ <configOption name="type">
+ <synopsis>Define this configuration category as a bridge profile</synopsis>
+ <description><para>The type parameter determines how a context in the
+ configuration file is interpreted.</para>
+ <enumlist>
+ <enum name="user"><para>Configure the context as a <replaceable>user_profile</replaceable></para></enum>
+ <enum name="bridge"><para>Configure the context as a <replaceable>bridge_profile</replaceable></para></enum>
+ <enum name="menu"><para>Configure the context as a <replaceable>menu</replaceable></para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="jitterbuffer">
+ <synopsis>Place a jitter buffer on the conference's audio stream</synopsis>
+ </configOption>
+ <configOption name="internal_sample_rate">
+ <synopsis>Set the internal native sample rate for mixing the conference</synopsis>
+ <description><para>
+ Sets the internal native sample rate the
+ conference is mixed at. This is set to automatically
+ adjust the sample rate to the best quality by default.
+ Other values can be anything from 8000-192000. If a
+ sample rate is set that Asterisk does not support, the
+ closest sample rate Asterisk does support to the one requested
+ will be used.
+ </para></description>
+ </configOption>
+ <configOption name="mixing_interval">
+ <synopsis>Sets the internal mixing interval in milliseconds for the bridge</synopsis>
+ <description><para>
+ Sets the internal mixing interval in milliseconds for the bridge. This
+ number reflects how tight or loose the mixing will be for the conference.
+ In order to improve performance a larger mixing interval such as 40ms may
+ be chosen. Using a larger mixing interval comes at the cost of introducing
+ larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
+ or 80.
+ </para></description>
+ </configOption>
+ <configOption name="record_conference">
+ <synopsis>Record the conference starting with the first active user's entrance and ending with the last active user's exit</synopsis>
+ <description><para>
+ Records the conference call starting when the first user
+ enters the room, and ending when the last user exits the room.
+ The default recorded filename is
+ <filename>'confbridge-${name of conference bridge}-${start time}.wav</filename>
+ and the default format is 8khz slinear. This file will be
+ located in the configured monitoring directory in asterisk.conf.
+ </para></description>
+ </configOption>
+ <configOption name="record_file" default="confbridge-${name of conference bridge}-${start time}.wav">
+ <synopsis>The filename of the conference recording</synopsis>
+ <description><para>
+ When record_conference is set to yes, the specific name of the
+ record file can be set using this option. Note that since multiple
+ conferences may use the same bridge profile, this may cause issues
+ depending on the configuration. It is recommended to only use this
+ option dynamically with the <literal>CONFBRIDGE()</literal> dialplan function. This
+ allows the record name to be specified and a unique name to be chosen.
+ By default, the record_file is stored in Asterisk's spool/monitor directory
+ with a unique filename starting with the 'confbridge' prefix.
+ </para></description>
+ </configOption>
+ <configOption name="video_mode">
+ <synopsis>Sets how confbridge handles video distribution to the conference participants</synopsis>
+ <description><para>
+ Sets how confbridge handles video distribution to the conference participants.
+ Note that participants wanting to view and be the source of a video feed
+ _MUST_ be sharing the same video codec. Also, using video in conjunction with
+ with the jitterbuffer currently results in the audio being slightly out of sync
+ with the video. This is a result of the jitterbuffer only working on the audio
+ stream. It is recommended to disable the jitterbuffer when video is used.</para>
+ <enumlist>
+ <enum name="none">
+ <para>No video sources are set by default in the conference. It is still
+ possible for a user to be set as a video source via AMI or DTMF action
+ at any time.</para>
+ </enum>
+ <enum name="follow_talker">
+ <para>The video feed will follow whoever is talking and providing video.</para>
+ </enum>
+ <enum name="last_marked">
+ <para>The last marked user to join the conference with video capabilities
+ will be the single source of video distributed to all participants.
+ If multiple marked users are capable of video, the last one to join
+ is always the source, when that user leaves it goes to the one who
+ joined before them.</para>
+ </enum>
+ <enum name="first_marked">
+ <para>The first marked user to join the conference with video capabilities
+ is the single source of video distribution among all participants. If
+ that user leaves, the marked user to join after them becomes the source.</para>
+ </enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="max_members">
+ <synopsis>Limit the maximum number of participants for a single conference</synopsis>
+ <description><para>
+ This option limits the number of participants for a single
+ conference to a specific number. By default conferences
+ have no participant limit. After the limit is reached, the
+ conference will be locked until someone leaves. Note however
+ that an Admin user will always be alowed to join the conference
+ regardless if this limit is reached or not.
+ </para></description>
+ </configOption>
+ <configOption name="^sound_">
+ <synopsis>Override the various conference bridge sound files</synopsis>
+ <description><para>
+ All sounds in the conference are customizable using the bridge profile options below.
+ Simply state the option followed by the filename or full path of the filename after
+ the option. Example: <literal>sound_had_joined=conf-hasjoin</literal> This will play the <literal>conf-hasjoin</literal>
+ sound file found in the sounds directory when announcing someone's name is joining the
+ conference.</para>
+ <enumlist>
+ <enum name="sound_join"><para>The sound played to everyone when someone enters the conference.</para></enum>
+ <enum name="sound_leave"><para>The sound played to everyone when someone leaves the conference.</para></enum>
+ <enum name="sound_has_joined"><para>The sound played before announcing someone's name has
+ joined the conference. This is used for user intros.
+ Example <literal>"_____ has joined the conference"</literal></para></enum>
+ <enum name="sound_has_left"><para>The sound played when announcing someone's name has
+ left the conference. This is used for user intros.
+ Example <literal>"_____ has left the conference"</literal></para></enum>
+ <enum name="sound_kicked"><para>The sound played to a user who has been kicked from the conference.</para></enum>
+ <enum name="sound_muted"><para>The sound played when the mute option it toggled on.</para></enum>
+ <enum name="sound_unmuted"><para>The sound played when the mute option it toggled off.</para></enum>
+ <enum name="sound_only_person"><para>The sound played when the user is the only person in the conference.</para></enum>
+ <enum name="sound_only_one"><para>The sound played to a user when there is only one other
+ person is in the conference.</para></enum>
+ <enum name="sound_there_are"><para>The sound played when announcing how many users there
+ are in a conference.</para></enum>
+ <enum name="sound_other_in_party"><para>This file is used in conjunction with <literal>sound_there_are</literal>
+ when announcing how many users there are in the conference.
+ The sounds are stringed together like this.
+ <literal>"sound_there_are" ${number of participants} "sound_other_in_party"</literal></para></enum>
+ <enum name="sound_place_into_conference"><para>The sound played when someone is placed into the conference
+ after waiting for a marked user.</para></enum>
+ <enum name="sound_wait_for_leader"><para>The sound played when a user is placed into a conference that
+ can not start until a marked user enters.</para></enum>
+ <enum name="sound_leader_has_left"><para>The sound played when the last marked user leaves the conference.</para></enum>
+ <enum name="sound_get_pin"><para>The sound played when prompting for a conference pin number.</para></enum>
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