[svn-commits] mjordan: trunk r381527 - in /trunk: ./ apps/ apps/confbridge/ channels/ confi...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Feb 15 07:38:21 CST 2013
Author: mjordan
Date: Fri Feb 15 07:38:12 2013
New Revision: 381527
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381527
Log:
Add CLI configuration documentation
This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:
1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
options that use the configuration framework
This patch include configuration documentation for the following modules:
* chan_motif
* res_xmpp
* app_confbridge
* app_skel
* udptl
Two new CLI commands have been added:
* config show help - show configuration help by module, category, and item
* xmldoc dump - dump the in-memory representation of the XML documentation to
a new XML file.
Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058
patches:
on review 2058 uploaded by twilson
Modified:
trunk/ (props changed)
trunk/Makefile
trunk/apps/app_skel.c
trunk/apps/confbridge/conf_config_parser.c
trunk/channels/chan_motif.c
trunk/configs/motif.conf.sample
trunk/configs/xmpp.conf.sample
trunk/doc/appdocsxml.dtd
trunk/include/asterisk/_private.h
trunk/include/asterisk/config_options.h
trunk/include/asterisk/xml.h
trunk/include/asterisk/xmldoc.h
trunk/main/asterisk.c
trunk/main/config_options.c
trunk/main/named_acl.c
trunk/main/udptl.c
trunk/main/xml.c
trunk/main/xmldoc.c
trunk/res/res_xmpp.c
Propchange: trunk/
------------------------------------------------------------------------------
automerge = *
Propchange: trunk/
------------------------------------------------------------------------------
svnmerge-integrated = /trunk:1-379815
Modified: trunk/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/Makefile?view=diff&rev=381527&r1=381526&r2=381527
==============================================================================
--- trunk/Makefile (original)
+++ trunk/Makefile Fri Feb 15 07:38:12 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: trunk/apps/app_skel.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_skel.c?view=diff&rev=381527&r1=381526&r2=381527
==============================================================================
--- trunk/apps/app_skel.c (original)
+++ trunk/apps/app_skel.c Fri Feb 15 07:38:12 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: trunk/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/confbridge/conf_config_parser.c?view=diff&rev=381527&r1=381526&r2=381527
==============================================================================
--- trunk/apps/confbridge/conf_config_parser.c (original)
+++ trunk/apps/confbridge/conf_config_parser.c Fri Feb 15 07:38:12 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>
+ <enum name="sound_invalid_pin"><para>The sound played when an invalid pin is entered too many times.</para></enum>
+ <enum name="sound_locked"><para>The sound played to a user trying to join a locked conference.</para></enum>
+ <enum name="sound_locked_now"><para>The sound played to an admin after toggling the conference to locked mode.</para></enum>
+ <enum name="sound_unlocked_now"><para>The sound played to an admin after toggling the conference to unlocked mode.</para></enum>
+ <enum name="sound_error_menu"><para>The sound played when an invalid menu option is entered.</para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ <configOption name="template">
+ <synopsis>When using the CONFBRIDGE dialplan function, use a bridge profile as a template for creating a new temporary profile</synopsis>
+ </configOption>
+ </configObject>
+ <configObject name="menu">
+ <synopsis>A conference user menu</synopsis>
+ <description>
+ <para>Conference users, as defined by a <literal>conf_user</literal>,
+ can have a DTMF menu assigned to their profile when they enter the
+ <literal>ConfBridge</literal> application.</para>
+ </description>
+ <configOption name="type">
+ <synopsis>Define this configuration category as a menu</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="^[0-9A-D*#]+$">
+ <synopsis>DTMF sequences to assign various confbridge actions to</synopsis>
+ <description><para>--- ConfBridge Menu Options ---</para>
+ <para>The ConfBridge application also has the ability to apply custom DTMF menus to
+ each channel using the application. Like the User and Bridge profiles a menu
+ is passed in to ConfBridge as an argument in the dialplan.</para>
+ <para>Below is a list of menu actions that can be assigned to a DTMF sequence.</para>
+ <note><para>
+ A single DTMF sequence can have multiple actions associated with it. This is
+ accomplished by stringing the actions together and using a <literal>,</literal> as the
+ delimiter. Example: Both listening and talking volume is reset when <literal>5</literal> is
+ pressed. <literal>5=reset_talking_volume, reset_listening_volume</literal></para></note>
+ <enumlist>
+ <enum name="playback(filename&filename2&...)"><para>
+ <literal>playback</literal> will play back an audio file to a channel
+ and then immediately return to the conference.
+ This file can not be interupted by DTMF.
+ Multiple files can be chained together using the
+ <literal>&</literal> character.</para></enum>
+ <enum name="playback_and_continue(filename&filename2&...)"><para>
+ <literal>playback_and_continue</literal> will
+ play back a prompt while continuing to
+ collect the dtmf sequence. This is useful
+ when using a menu prompt that describes all
+ the menu options. Note however that any DTMF
+ during this action will terminate the prompts
+ playback. Prompt files can be chained together
+ using the <literal>&</literal> character as a delimiter.</para></enum>
+ <enum name="toggle_mute"><para>
+ Toggle turning on and off mute. Mute will make the user silent
+ to everyone else, but the user will still be able to listen in.
+ continue to collect the dtmf sequence.</para></enum>
+ <enum name="no_op"><para>
+ This action does nothing (No Operation). Its only real purpose exists for
+ being able to reserve a sequence in the config as a menu exit sequence.</para></enum>
+ <enum name="decrease_listening_volume"><para>
+ Decreases the channel's listening volume.</para></enum>
+ <enum name="increase_listening_volume"><para>
+ Increases the channel's listening volume.</para></enum>
+ <enum name="reset_listening_volume"><para>
+ Reset channel's listening volume to default level.</para></enum>
+ <enum name="decrease_talking_volume"><para>
+ Decreases the channel's talking volume.</para></enum>
+ <enum name="increase_talking_volume"><para>
+ Increases the channel's talking volume.</para></enum>
+ <enum name="reset_talking_volume"><para>
+ Reset channel's talking volume to default level.</para></enum>
+ <enum name="dialplan_exec(context,exten,priority)"><para>
+ The <literal>dialplan_exec</literal> action allows a user
+ to escape from the conference and execute
+ commands in the dialplan. Once the dialplan
+ exits the user will be put back into the
+ conference. The possibilities are endless!</para></enum>
+ <enum name="leave_conference"><para>
+ This action allows a user to exit the conference and continue
+ execution in the dialplan.</para></enum>
+ <enum name="admin_kick_last"><para>
+ This action allows an Admin to kick the last participant from the
+ conference. This action will only work for admins which allows
+ a single menu to be used for both users and admins.</para></enum>
+ <enum name="admin_toggle_conference_lock"><para>
+ This action allows an Admin to toggle locking and
+ unlocking the conference. Non admins can not use
+ this action even if it is in their menu.</para></enum>
+ <enum name="set_as_single_video_src"><para>
+ This action allows any user to set themselves as the
+ single video source distributed to all participants.
+ This will make the video feed stick to them regardless
+ of what the <literal>video_mode</literal> is set to.</para></enum>
+ <enum name="release_as_single_video_src"><para>
+ This action allows a user to release themselves as
+ the video source. If <literal>video_mode</literal> is not set to <literal>none</literal>
+ this action will result in the conference returning to
+ whatever video mode the bridge profile is using.</para>
+ <para>Note that this action will have no effect if the user
+ is not currently the video source. Also, the user is
+ not guaranteed by using this action that they will not
+ become the video source again. The bridge will return
+ to whatever operation the <literal>video_mode</literal> option is set to
+ upon release of the video src.</para></enum>
+ <enum name="admin_toggle_mute_participants"><para>
+ This action allows an administrator to toggle the mute
+ state for all non-admins within a conference. All
+ admin users are unaffected by this option. Note that all
+ users, regardless of their admin status, are notified
+ that the conference is muted.</para></enum>
+ <enum name="participant_count"><para>
+ This action plays back the number of participants currently
+ in a conference</para></enum>
+ </enumlist>
+ </description>
+ </configOption>
+ </configObject>
+ </configFile>
+ </configInfo>
+***/
+
struct confbridge_cfg {
struct ao2_container *bridge_profiles;
struct ao2_container *user_profiles;
@@ -81,6 +539,7 @@
static struct aco_type bridge_type = {
.type = ACO_ITEM,
+ .name = "bridge_profile",
.category_match = ACO_BLACKLIST,
.category = "^general$",
.matchfield = "type",
@@ -117,6 +576,7 @@
static struct aco_type user_type = {
.type = ACO_ITEM,
+ .name = "user_profile",
.category_match = ACO_BLACKLIST,
.category = "^general$",
.matchfield = "type",
@@ -147,6 +607,7 @@
static struct aco_type menu_type = {
.type = ACO_ITEM,
+ .name = "menu",
.category_match = ACO_BLACKLIST,
.category = "^general$",
.matchfield = "type",
@@ -164,6 +625,7 @@
/* The general category is reserved, but unused */
static struct aco_type general_type = {
.type = ACO_GLOBAL,
+ .name = "global",
.category_match = ACO_WHITELIST,
.category = "^general$",
};
@@ -1293,8 +1755,6 @@
/* User options */
aco_option_register(&cfg_info, "type", ACO_EXACT, user_types, NULL, OPT_NOOP_T, 0, 0);
- aco_option_register(&cfg_info, "type", ACO_EXACT, bridge_types, NULL, OPT_NOOP_T, 0, 0);
- aco_option_register(&cfg_info, "type", ACO_EXACT, menu_types, NULL, OPT_NOOP_T, 0, 0);
aco_option_register(&cfg_info, "admin", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_ADMIN);
aco_option_register(&cfg_info, "marked", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_MARKEDUSER);
aco_option_register(&cfg_info, "startmuted", ACO_EXACT, user_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct user_profile, flags), USER_OPT_STARTMUTED);
@@ -1321,6 +1781,7 @@
aco_option_register_custom(&cfg_info, "template", ACO_EXACT, user_types, NULL, user_template_handler, 0);
/* Bridge options */
+ aco_option_register(&cfg_info, "type", ACO_EXACT, bridge_types, NULL, OPT_NOOP_T, 0, 0);
aco_option_register(&cfg_info, "jitterbuffer", ACO_EXACT, bridge_types, "no", OPT_BOOLFLAG_T, 1, FLDSET(struct bridge_profile, flags), USER_OPT_JITTERBUFFER);
/* "auto" will fail to parse as a uint, but we use PARSE_DEFAULT to set the value to 0 in that case, which is the value that auto resolves to */
aco_option_register(&cfg_info, "internal_sample_rate", ACO_EXACT, bridge_types, "0", OPT_UINT_T, PARSE_DEFAULT, FLDSET(struct bridge_profile, internal_sample_rate), 0);
@@ -1334,6 +1795,7 @@
aco_option_register_custom(&cfg_info, "template", ACO_EXACT, bridge_types, NULL, bridge_template_handler, 0);
/* Menu options */
+ aco_option_register(&cfg_info, "type", ACO_EXACT, menu_types, NULL, OPT_NOOP_T, 0, 0);
aco_option_register_custom(&cfg_info, "^[0-9A-D*#]+$", ACO_REGEX, menu_types, NULL, menu_option_handler, 0);
if (aco_process_config(&cfg_info, reload) == ACO_PROCESS_ERROR) {
Modified: trunk/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_motif.c?view=diff&rev=381527&r1=381526&r2=381527
==============================================================================
--- trunk/channels/chan_motif.c (original)
+++ trunk/channels/chan_motif.c Fri Feb 15 07:38:12 2013
@@ -76,6 +76,145 @@
#include "asterisk/astobj.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/xmpp.h"
+
+/*** DOCUMENTATION
+ <configInfo name="chan_motif" language="en_US">
+ <synopsis>Jingle Channel Driver</synopsis>
+ <description>
+ <para><emphasis>Transports</emphasis></para>
+ <para>There are three different transports and protocol derivatives
+ supported by <literal>chan_motif</literal>. They are in order of
+ preference: Jingle using ICE-UDP, Google Jingle, and Google-V1.</para>
+ <para>Jingle as defined in XEP-0166 supports the widest range of
+ features. It is referred to as <literal>ice-udp</literal>. This is
+ the specification that Jingle clients implement.</para>
+ <para>Google Jingle follows the Jingle specification for signaling
+ but uses a custom transport for media. It is supported by the
+ Google Talk Plug-in in Gmail and by some other Jingle clients. It
+ is referred to as <literal>google</literal> in this file.</para>
+ <para>Google-V1 is the original Google Talk signaling protocol
+ which uses an initial preliminary version of Jingle. It also uses
+ the same custom transport as Google Jingle for media. It is
+ supported by Google Voice, some other Jingle clients, and the
+ Windows Google Talk client. It is referred to as <literal>google-v1</literal>
+ in this file.</para>
+ <para>Incoming sessions will automatically switch to the correct
+ transport once it has been determined.</para>
+ <para>Outgoing sessions are capable of determining if the target
+ is capable of Jingle or a Google transport if the target is in the
+ roster. Unfortunately it is not possible to differentiate between
+ a Google Jingle or Google-V1 capable resource until a session
+ initiate attempt occurs. If a resource is determined to use a
+ Google transport it will initially use Google Jingle but will fall
+ back to Google-V1 if required.</para>
+ <para>If an outgoing session attempt fails due to failure to
+ support the given transport <literal>chan_motif</literal> will
+ fall back in preference order listed previously until all
+ transports have been exhausted.</para>
+ <para><emphasis>Dialing and Resource Selection Strategy</emphasis></para>
+ <para>Placing a call through an endpoint can be accomplished using the
+ following dial string:</para>
+ <para><literal>Motif/[endpoint name]/[target]</literal></para>
+ <para>When placing an outgoing call through an endpoint the requested
+ target is searched for in the roster list. If present the first Jingle
+ or Google Jingle capable resource is specifically targeted. Since the
+ capabilities of the resource are known the outgoing session initiation
+ will disregard the configured transport and use the determined one.</para>
+ <para>If the target is not found in the roster the target will be used
+ as-is and a session will be initiated using the transport specified
+ in this configuration file. If no transport has been specified the
+ endpoint defaults to <literal>ice-udp</literal>.</para>
+ <para><emphasis>Video Support</emphasis></para>
+ <para>Support for video does not need to be explicitly enabled.
+ Configuring any video codec on your endpoint will automatically enable
+ it.</para>
+ <para><emphasis>DTMF</emphasis></para>
+ <para>The only supported method for DTMF is RFC2833. This is always
+ enabled on audio streams and negotiated if possible.</para>
+ <para><emphasis>Incoming Calls</emphasis></para>
+ <para>Incoming calls will first look for the extension matching the
+ name of the endpoint in the configured context. If no such extension
+ exists the call will automatically fall back to the <literal>s</literal> extension.</para>
+ <para><emphasis>CallerID</emphasis></para>
[... 1794 lines stripped ...]
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