[asterisk-commits] file: branch file/pimp_sip_media r381580 - in /team/file/pimp_sip_media: ./ a...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Feb 15 14:22:53 CST 2013
Author: file
Date: Fri Feb 15 14:22:49 2013
New Revision: 381580
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381580
Log:
Multiple revisions 381524,381531,381544,381546,381579
........
r381524 | file | 2013-02-14 21:17:04 -0400 (Thu, 14 Feb 2013) | 1 line
Add a missing ao2_ref call for the media update task.
........
r381531 | root | 2013-02-15 10:19:28 -0400 (Fri, 15 Feb 2013) | 1 line
automerge cancel
........
r381544 | mmichelson | 2013-02-15 11:27:11 -0400 (Fri, 15 Feb 2013) | 3 lines
Resolve conflict and reset automerge.
........
r381546 | root | 2013-02-15 12:17:52 -0400 (Fri, 15 Feb 2013) | 1 line
automerge cancel
........
r381579 | mmichelson | 2013-02-15 15:49:32 -0400 (Fri, 15 Feb 2013) | 3 lines
Resolve conflict and reset automerge
........
Merged revisions 381524,381531,381544,381546,381579 from http://svn.asterisk.org/svn/asterisk/team/group/pimp_my_sip
Modified:
team/file/pimp_sip_media/ (props changed)
team/file/pimp_sip_media/Makefile
team/file/pimp_sip_media/apps/app_skel.c
team/file/pimp_sip_media/apps/confbridge/conf_config_parser.c
team/file/pimp_sip_media/channels/chan_motif.c
team/file/pimp_sip_media/channels/chan_sip.c
team/file/pimp_sip_media/configs/motif.conf.sample
team/file/pimp_sip_media/configs/xmpp.conf.sample
team/file/pimp_sip_media/doc/appdocsxml.dtd
team/file/pimp_sip_media/include/asterisk/_private.h
team/file/pimp_sip_media/include/asterisk/config_options.h
team/file/pimp_sip_media/include/asterisk/logger.h
team/file/pimp_sip_media/include/asterisk/sorcery.h
team/file/pimp_sip_media/include/asterisk/xml.h
team/file/pimp_sip_media/include/asterisk/xmldoc.h
team/file/pimp_sip_media/main/asterisk.c
team/file/pimp_sip_media/main/autoservice.c
team/file/pimp_sip_media/main/config_options.c
team/file/pimp_sip_media/main/logger.c
team/file/pimp_sip_media/main/named_acl.c
team/file/pimp_sip_media/main/sorcery.c
team/file/pimp_sip_media/main/udptl.c
team/file/pimp_sip_media/main/xml.c
team/file/pimp_sip_media/main/xmldoc.c
team/file/pimp_sip_media/res/res_xmpp.c
Propchange: team/file/pimp_sip_media/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/file/pimp_sip_media/
------------------------------------------------------------------------------
--- pimp-integrated (original)
+++ pimp-integrated Fri Feb 15 14:22:49 2013
@@ -1,1 +1,1 @@
-/team/group/pimp_my_sip:1-381522
+/team/group/pimp_my_sip:1-381579
Modified: team/file/pimp_sip_media/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/Makefile?view=diff&rev=381580&r1=381579&r2=381580
==============================================================================
--- team/file/pimp_sip_media/Makefile (original)
+++ team/file/pimp_sip_media/Makefile Fri Feb 15 14:22:49 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/file/pimp_sip_media/apps/app_skel.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/apps/app_skel.c?view=diff&rev=381580&r1=381579&r2=381580
==============================================================================
--- team/file/pimp_sip_media/apps/app_skel.c (original)
+++ team/file/pimp_sip_media/apps/app_skel.c Fri Feb 15 14:22:49 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/file/pimp_sip_media/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/apps/confbridge/conf_config_parser.c?view=diff&rev=381580&r1=381579&r2=381580
==============================================================================
--- team/file/pimp_sip_media/apps/confbridge/conf_config_parser.c (original)
+++ team/file/pimp_sip_media/apps/confbridge/conf_config_parser.c Fri Feb 15 14:22:49 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: team/file/pimp_sip_media/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/channels/chan_motif.c?view=diff&rev=381580&r1=381579&r2=381580
==============================================================================
--- team/file/pimp_sip_media/channels/chan_motif.c (original)
+++ team/file/pimp_sip_media/channels/chan_motif.c Fri Feb 15 14:22:49 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>
[... 2113 lines stripped ...]
More information about the asterisk-commits
mailing list