[asterisk-commits] mjordan: branch mjordan/terrys_parting_shot r379597 - in /team/mjordan/terrys...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sat Jan 19 23:03:33 CST 2013
Author: mjordan
Date: Sat Jan 19 23:03:29 2013
New Revision: 379597
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=379597
Log:
Update: collapse CLI commands down to a single command, update docs
The four CLI commands are now just one: 'config show help'.
Updated app_confbridge, udptl, part of chan_motif.
Modified:
team/mjordan/terrys_parting_shot/apps/confbridge/conf_config_parser.c
team/mjordan/terrys_parting_shot/channels/chan_motif.c
team/mjordan/terrys_parting_shot/doc/appdocsxml.dtd
team/mjordan/terrys_parting_shot/include/asterisk/xmldoc.h
team/mjordan/terrys_parting_shot/main/config_options.c
team/mjordan/terrys_parting_shot/main/udptl.c
team/mjordan/terrys_parting_shot/main/xmldoc.c
Modified: team/mjordan/terrys_parting_shot/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/mjordan/terrys_parting_shot/apps/confbridge/conf_config_parser.c?view=diff&rev=379597&r1=379596&r2=379597
==============================================================================
--- team/mjordan/terrys_parting_shot/apps/confbridge/conf_config_parser.c (original)
+++ team/mjordan/terrys_parting_shot/apps/confbridge/conf_config_parser.c Sat Jan 19 23:03:29 2013
@@ -45,12 +45,25 @@
<configInfo name="app_confbridge" language="en_US">
<configFile name="confbridge.conf">
<configObject name="global">
- <synopsis>Unused, but reserved</synopsis>
+ <synopsis>Unused, but reserved.</synopsis>
</configObject>
<configObject name="user_profile">
- <synopsis>A named profile to apply to specific callers</synopsis>
+ <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 <replaceable>type</replaceable> parameter has a value
+ of <literal>user</literal>.
+ </para></description>
<configOption name="type">
- <synopsis>Define this configuration category as a user profile</synopsis>
+ <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>
@@ -73,9 +86,9 @@
<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.
+ 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">
@@ -108,55 +121,57 @@
<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.
+ 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.
+ 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 ofmilliseconds 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.
-
- What this value affects internally:
-
+ 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.
+ 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.
-
+ 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.
-
- By default this value is 2500ms. Valid values are 1 through 2^31
- </para></description>
+ </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>
@@ -165,37 +180,42 @@
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.
-
+ 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.
+ 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.
+ 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.
-
- By default this value is 160 ms. Valid values are 1 through 2^31
- </para></description>
+ 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 JITTERBUFFER
+ 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 JITTERBUFFER dialplan function
+ jitterbuffer, disable this option and use the <literal>JITTERBUFFER</literal> dialplan function
on the user before entering the ConfBridge application.
</para></description>
</configOption>
@@ -204,8 +224,22 @@
</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 <replaceable>type</replaceable> 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>
@@ -239,7 +273,7 @@
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
- 'confbridge-${name of conference bridge}-${start time}.wav
+ <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>
@@ -251,7 +285,7 @@
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 CONFBRIDGE() dialplan function. 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.
@@ -265,25 +299,30 @@
_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.
-
- --- MODES ---
- none: 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.
-
- follow_talker: The video feed will follow whoever is talking and providing video.
-
- last_marked: 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.
-
- first_marked: 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></description>
+ 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>
@@ -301,42 +340,43 @@
<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: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
+ 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.
-
- sound_join : The sound played to everyone when someone enters the conference.
- sound_leave : The sound played to everyone when someone leaves the conference.
- sound_has_joined : The sound played before announcing someone's name has
- joined the conference. This is used for user intros.
- Example "_____ has joined the conference"
- sound_has_left : The sound played when announcing someone's name has
- left the conference. This is used for user intros.
- Example "_____ has left the conference"
- sound_kicked : The sound played to a user who has been kicked from the conference.
- sound_muted : The sound played when the mute option it toggled on.
- sound_unmuted : The sound played when the mute option it toggled off.
- sound_only_person: The sound played when the user is the only person in the conference.
- sound_only_one : The sound played to a user when there is only one other
- person is in the conference.
- sound_there_are : The sound played when announcing how many users there
- are in a conference.
- sound_other_in_party : This file is used in conjunction with 'sound_there_are"
- when announcing how many users there are in the conference.
- The sounds are stringed together like this.
- "sound_there_are" ${number of participants} "sound_other_in_party"
- sound_place_into_conference : The sound played when someone is placed into the conference
- after waiting for a marked user.
- sound_wait_for_leader : The sound played when a user is placed into a conference that
- can not start until a marked user enters.
- sound_leader_has_left : The sound played when the last marked user leaves the conference.
- sound_get_pin : The sound played when prompting for a conference pin number.
- sound_invalid_pin : The sound played when an invalid pin is entered too many times.
- sound_locked : The sound played to a user trying to join a locked conference.
- sound_locked_now : The sound played to an admin after toggling the conference to locked mode.
- sound_unlocked_now: The sound played to an admin after toggling the conference to unlocked mode.
- sound_error_menu : The sound played when an invalid menu option is entered.
- </para></description>
+ 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>
@@ -345,92 +385,106 @@
<configObject name="menu">
<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 ---
-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.
-
-Below is a list of menu actions that can be assigned to a DTMF sequence.
-
-A single DTMF sequence can have multiple actions associated with it. This is
-accomplished by stringing the actions together and using a ',' as the
-delimiter. Example: Both listening and talking volume is reset when '5' is
-pressed. 5=reset_talking_volume, reset_listening_volume
-
-playback(filename&filename2&...)
- ; Playback will play back an audio file to a channel
- ; and then immediately return to the conference.
- ; This file can not be interupted by DTMF.
- ; Mutliple files can be chained together using the
- ; '&' character.
-playback_and_continue(filename&filename2&...)
- ; playback_and_continue 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 '&' character as a delimiter.
-toggle_mute ; 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.
-no_op ; 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.
-decrease_listening_volume ; Decreases the channel's listening volume.
-increase_listening_volume ; Increases the channel's listening volume.
-reset_listening_volume ; Reset channel's listening volume to default level.
-
-decrease_talking_volume ; Decreases the channel's talking volume.
-increase_talking_volume ; Icreases the channel's talking volume.
-reset_talking_volume ; Reset channel's talking volume to default level.
-
-dialplan_exec(context,exten,priority) ; The dialplan_exec 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!
-leave_conference ; This action allows a user to exit the conference and continue
- ; execution in the dialplan.
-
-admin_kick_last ; 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.
-
-admin_toggle_conference_lock ; 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.
-
-set_as_single_video_src ; 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 video_mode is set to.
-
-release_as_single_video_src ; This action allows a user to release themselves as
- ; the video source. If video_mode is not set to "none"
- ; this action will result in the conference returning to
- ; whatever video mode the bridge profile is using.
- ;
- ; 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 video_mode option is set to
- ; upon release of the video src.
-
-admin_toggle_mute_participants ; 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.
-
-participant_count ; This action plays back the number of participants currently
- ; in a conference
- </para></description>
+ <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>
Modified: team/mjordan/terrys_parting_shot/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/team/mjordan/terrys_parting_shot/channels/chan_motif.c?view=diff&rev=379597&r1=379596&r2=379597
==============================================================================
--- team/mjordan/terrys_parting_shot/channels/chan_motif.c (original)
+++ team/mjordan/terrys_parting_shot/channels/chan_motif.c Sat Jan 19 23:03:29 2013
@@ -81,23 +81,26 @@
<configInfo name="chan_motif" language="en_US">
<configFile name="motif.conf">
<configObject name="endpoint">
+ <synopsis>The configuration for an endpoint.</synopsis>
+ <description><para>
+ </para></description>
<configOption name="context">
<synopsis>Default dialplan context that incoming sessions will be routed to</synopsis>
</configOption>
<configOption name="callgroup">
- <synopsis></synopsis>
+ <synopsis>A callgroup to assign to this endpoint.</synopsis>
</configOption>
<configOption name="pickupgroup">
- <synopsis></synopsis>
+ <synopsis>A pickup group to assign to this endpoint.</synopsis>
</configOption>
<configOption name="language">
- <synopsis></synopsis>
+ <synopsis>The default language for this endpoint.</synopsis>
</configOption>
<configOption name="musicclass">
- <synopsis></synopsis>
+ <synopsis>Default music on hold class for this endpoint.</synopsis>
</configOption>
<configOption name="parkinglot">
- <synopsis></synopsis>
+ <synopsis>Default parking lot for this endpoint.</synopsis>
</configOption>
<configOption name="accountcode">
<synopsis>Accout code for CDR purposes</synopsis>
@@ -112,7 +115,45 @@
<synopsis>Connection to accept traffic on and on which to send traffic out</synopsis>
</configOption>
<configOption name="transport">
- <synopsis>The transport to use (ice-udp, google, or google-v1)</synopsis>
+ <synopsis>The transport to use for the endpoint.</synopsis>
+ <description>
+ <para>There are three different transports and protocol derivatives supported
+ by <literal>chan_motif</literal>. They are in order of preference:</para>
+ <para>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>Choose the transport for this endpoint. Allowed transports are
+ <literal>ice-udp</literal>, <literal>google</literal>, or <literal>google-v1</literal>.</para>
+ <enumlist>
+ <enum name="ice-udp"></enum>
+ <enum name="google"></enum>
+ <enum name="google-v1"></enum>
+ </enumlist>
+ </description>
</configOption>
<configOption name="maxicecandidates">
<synopsis>Maximum number of ICE candidates to offer</synopsis>
Modified: team/mjordan/terrys_parting_shot/doc/appdocsxml.dtd
URL: http://svnview.digium.com/svn/asterisk/team/mjordan/terrys_parting_shot/doc/appdocsxml.dtd?view=diff&rev=379597&r1=379596&r2=379597
==============================================================================
--- team/mjordan/terrys_parting_shot/doc/appdocsxml.dtd (original)
+++ team/mjordan/terrys_parting_shot/doc/appdocsxml.dtd Sat Jan 19 23:03:29 2013
@@ -46,13 +46,14 @@
<!ELEMENT configFile (configObject+)>
<!ATTLIST configFile name CDATA #REQUIRED>
- <!ELEMENT configObject (synopsis?|description?|syntax?|configOption)*>
[... 799 lines stripped ...]
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