[svn-commits] twilson: branch group/aco_xmldocs r374031 - in /team/group/aco_xmldocs: apps/...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Sep 28 13:28:46 CDT 2012
Author: twilson
Date: Fri Sep 28 13:28:42 2012
New Revision: 374031
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=374031
Log:
Add res_xmpp and chan_motif docs and remove "defaults" from docs
I forgot that I have the default/regex options filled at runtime.
res_xmpp and chan_motif had some options that were not documented in the sample files.
I'm running out of time, so I didn't document them either, but made note of what they were.
If I have time today, I'll go in and make docs for them.
Modified:
team/group/aco_xmldocs/apps/confbridge/conf_config_parser.c
team/group/aco_xmldocs/channels/chan_motif.c
team/group/aco_xmldocs/configs/motif.conf.sample
team/group/aco_xmldocs/configs/xmpp.conf.sample
team/group/aco_xmldocs/res/res_xmpp.c
Modified: team/group/aco_xmldocs/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/group/aco_xmldocs/apps/confbridge/conf_config_parser.c?view=diff&rev=374031&r1=374030&r2=374031
==============================================================================
--- team/group/aco_xmldocs/apps/confbridge/conf_config_parser.c (original)
+++ team/group/aco_xmldocs/apps/confbridge/conf_config_parser.c Fri Sep 28 13:28:42 2012
@@ -52,25 +52,25 @@
<configOption name="type">
<synopsis>Define this configuration category as a user profile</synopsis>
</configOption>
- <configOption name="admin" default="no">
+ <configOption name="admin">
<synopsis>Sets if the user is an admin or not</synopsis>
</configOption>
- <configOption name="marked" default="no">
+ <configOption name="marked">
<synopsis>Sets if this is a marked user or not</synopsis>
</configOption>
- <configOption name="startmuted" default="no">
+ <configOption name="startmuted">
<synopsis>Sets if all users should start out muted</synopsis>
</configOption>
- <configOption name="music_on_hold_when_empty" default="no">
+ <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" default="no">
+ <configOption name="quiet">
<synopsis>Silence enter/leave prompts and user intros for this user</synopsis>
</configOption>
- <configOption name="announce_user_count" default="no">
+ <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" default="no">
+ <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
@@ -78,22 +78,22 @@
count is above the specified number.
</para></description>
</configOption>
- <configOption name="announce_only_user" default="yes">
+ <configOption name="announce_only_user">
<synopsis>Announce to a user when they join an empty conference</synopsis>
</configOption>
- <configOption name="wait_marked" default="no">
+ <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" default="no">
+ <configOption name="end_marked">
<synopsis>Kick the user from the conference when the last marked user leaves</synopsis>
</configOption>
- <configOption name="talk_detection_events" default="no">
+ <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" default="no">
+ <configOption name="dtmf_passthrough">
<synopsis>Sets whether or not DTMF should pass through the conference</synopsis>
</configOption>
- <configOption name="announce_join_leave" default="no">
+ <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">
@@ -105,7 +105,7 @@
<configOption name="announcement">
<synopsis>Sound file to play to the user when they join a conference</synopsis>
</configOption>
- <configOption name="denoise" default="no">
+ <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
@@ -126,7 +126,7 @@
due to its performance enhancements.
</para></description>
</configOption>
- <configOption name="dsp_silence_threshold" default="2500">
+ <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
@@ -158,7 +158,7 @@
By default this value is 2500ms. Valid values are 1 through 2^31
</para></description>
</configOption>
- <configOption name="dsp_talking_threshold" default="60">
+ <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
@@ -188,7 +188,7 @@
By default this value is 160 ms. Valid values are 1 through 2^31
</para></description>
</configOption>
- <configOption name="jitterbuffer" default="no">
+ <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
@@ -207,10 +207,10 @@
<configOption name="type">
<synopsis>Define this configuration category as a bridge profile</synopsis>
</configOption>
- <configOption name="jitterbuffer" default="no">
+ <configOption name="jitterbuffer">
<synopsis>Place a jitter buffer on the conference's audio stream</synopsis>
</configOption>
- <configOption name="internal_sample_rate" default="auto">
+ <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
@@ -222,7 +222,7 @@
will be used.
</para></description>
</configOption>
- <configOption name="mixing_interval" default="20">
+ <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
@@ -233,7 +233,7 @@
or 80.
</para></description>
</configOption>
- <configOption name="record_conference" default="no">
+ <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
@@ -296,7 +296,7 @@
regardless if this limit is reached or not.
</para></description>
</configOption>
- <configOption name="^sound_" regex="true">
+ <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.
Modified: team/group/aco_xmldocs/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/team/group/aco_xmldocs/channels/chan_motif.c?view=diff&rev=374031&r1=374030&r2=374031
==============================================================================
--- team/group/aco_xmldocs/channels/chan_motif.c (original)
+++ team/group/aco_xmldocs/channels/chan_motif.c Fri Sep 28 13:28:42 2012
@@ -74,7 +74,7 @@
<configFile name="motif.conf">
<configObject name="endpoint">
<configOption name="context">
- <synopsis></synopsis>
+ <synopsis>Default dialplan context that incoming sessions will be routed to</synopsis>
</configOption>
<configOption name="callgroup">
<synopsis></synopsis>
@@ -92,25 +92,25 @@
<synopsis></synopsis>
</configOption>
<configOption name="accountcode">
- <synopsis></synopsis>
+ <synopsis>Accout code for CDR purposes</synopsis>
</configOption>
<configOption name="allow">
- <synopsis></synopsis>
+ <synopsis>Codecs to allow</synopsis>
</configOption>
<configOption name="disallow">
- <synopsis></synopsis>
+ <synopsis>Codecs to disallow</synopsis>
</configOption>
<configOption name="connection">
- <synopsis></synopsis>
+ <synopsis>Connection to accept traffic on and on which to send traffic out</synopsis>
</configOption>
<configOption name="transport">
- <synopsis></synopsis>
+ <synopsis>The transport to use (ice-udp, google, or google-v1)</synopsis>
</configOption>
<configOption name="maxicecandidates">
- <synopsis></synopsis>
+ <synopsis>Maximum number of ICE candidates to offer</synopsis>
</configOption>
<configOption name="maxpayloads">
- <synopsis></synopsis>
+ <synopsis>Maximum number of pyaloads to offer</synopsis>
</configOption>
</configObject>
</configFile>
Modified: team/group/aco_xmldocs/configs/motif.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/group/aco_xmldocs/configs/motif.conf.sample?view=diff&rev=374031&r1=374030&r2=374031
==============================================================================
--- team/group/aco_xmldocs/configs/motif.conf.sample (original)
+++ team/group/aco_xmldocs/configs/motif.conf.sample Fri Sep 28 13:28:42 2012
@@ -70,18 +70,25 @@
;maxpayloads = 30 ; Maximum number of payloads we will offer
; Sample configuration entry for Jingle
-[jingle-endpoint](default)
-transport=ice-udp ; Change the default protocol of outgoing sessions to Jingle ICE-UDP
-allow=g722 ; Add G.722 as an allowed format since the other side may support it
-connection=local-jabber-account ; Connection to accept traffic on and send traffic out
-accountcode=jingle ; Account code for CDR purposes
+;[jingle-endpoint](default)
+;transport=ice-udp ; Change the default protocol of outgoing sessions to Jingle ICE-UDP
+;allow=g722 ; Add G.722 as an allowed format since the other side may support it
+;connection=local-jabber-account ; Connection to accept traffic on and send traffic out
+;accountcode=jingle ; Account code for CDR purposes
; Sample configuration entry for Google Talk
[gtalk-endpoint](default)
-transport=google ; Since this is a Google Talk endpoint we want to offer Google Jingle for outgoing sessions
-connection=gtalk-account
+;transport=google ; Since this is a Google Talk endpoint we want to offer Google Jingle for outgoing sessions
+;connection=gtalk-account
; Sample configuration entry for Google Voice
-[gvoice](default)
-transport=google-v1 ; Google Voice uses the original Google Talk protocol
-connection=gvoice-account
+;[gvoice](default)
+;transport=google-v1 ; Google Voice uses the original Google Talk protocol
+;connection=gvoice-account
+
+; Additional options
+; callgroup
+; pickupgroup
+; language
+; musicclass
+; parkinglot
Modified: team/group/aco_xmldocs/configs/xmpp.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/group/aco_xmldocs/configs/xmpp.conf.sample?view=diff&rev=374031&r1=374030&r2=374031
==============================================================================
--- team/group/aco_xmldocs/configs/xmpp.conf.sample (original)
+++ team/group/aco_xmldocs/configs/xmpp.conf.sample Fri Sep 28 13:28:42 2012
@@ -37,3 +37,6 @@
;sendtodialplan=yes ; Send incoming messages into the dialplan. Off by default.
;context=messages ; Dialplan context to send incoming messages to. If not set,
; "default" will be used.
+;forceoldssl=no ; Force the use of old-style SSL.
+;keepalive=
+
Modified: team/group/aco_xmldocs/res/res_xmpp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/aco_xmldocs/res/res_xmpp.c?view=diff&rev=374031&r1=374030&r2=374031
==============================================================================
--- team/group/aco_xmldocs/res/res_xmpp.c (original)
+++ team/group/aco_xmldocs/res/res_xmpp.c Fri Sep 28 13:28:42 2012
@@ -278,91 +278,106 @@
<configFile name="xmpp.conf">
<configObject name="global">
<configOption name="debug">
- <synopsis></synopsis>
+ <synopsis>Enable debugging</synopsis>
</configOption>
<configOption name="autoprune">
- <synopsis></synopsis>
+ <synopsis>Auto-remove users from buddy list.</synopsis>
+ <description><para>Auto-remove users from buddy list. Depending on the setup
+ (e.g., using your personal Gtalk account for a test) this could cause loss of
+ the contact list.
+ </para></description>
</configOption>
<configOption name="autoregister">
- <synopsis></synopsis>
+ <synopsis>Auto-register users bfrom buddy list</synopsis>
</configOption>
<configOption name="collection_nodes">
- <synopsis></synopsis>
+ <synopsis>Enable support for XEP-0248 for use with distributed device state</synopsis>
</configOption>
<configOption name="pubsub_autocreate">
- <synopsis></synopsis>
+ <synopsis>Whether or not the PubSub server supports/is using auto-create for nodes</synopsis>
</configOption>
<configOption name="auth_policy">
- <synopsis></synopsis>
+ <synopsis>Whether to automatically accept or deny users' subscription requests</synopsis>
</configOption>
</configObject>
<configObject name="client">
<configOption name="username">
- <synopsis></synopsis>
+ <synopsis>XMPP username with optional resource</synopsis>
</configOption>
<configOption name="secret">
- <synopsis></synopsis>
+ <synopsis>XMPP password</synopsis>
</configOption>
<configOption name="serverhost">
- <synopsis></synopsis>
+ <synopsis>Route to server, e.g. talk.google.com</synopsis>
</configOption>
<configOption name="statusmessage">
- <synopsis></synopsis>
+ <synopsis>Custom status message</synopsis>
</configOption>
<configOption name="pubsub_node">
- <synopsis></synopsis>
+ <synopsis>Node for publishing events via PubSub</synopsis>
</configOption>
<configOption name="context">
- <synopsis></synopsis>
+ <synopsis>Dialplan context to send incoming messages to</synopsis>
</configOption>
<configOption name="priority">
- <synopsis></synopsis>
+ <synopsis>XMPP resource priority</synopsis>
</configOption>
<configOption name="port">
- <synopsis></synopsis>
+ <synopsis>XMPP server port</synopsis>
</configOption>
<configOption name="timeout">
- <synopsis></synopsis>
+ <synopsis>Timeout in seconds to hold incoming messages</synopsis>
+ <description><para>Timeout (in seconds) on the message stack. Messages stored longer
+ than this value will be deleted by Asterisk. This option applies to incoming messages only
+ which are intended to be processed by the JABBER_RECEIVE dialplan function.
+ </para></description>
</configOption>
-
<configOption name="debug">
- <synopsis></synopsis>
+ <synopsis>Enable debugging</synopsis>
</configOption>
<configOption name="type">
- <synopsis></synopsis>
+ <synopsis>Connection is either a client or a component</synopsis>
</configOption>
<configOption name="distribute_events">
- <synopsis></synopsis>
+ <synopsis>Whether or not to distribute events using this connection</synopsis>
</configOption>
<configOption name="usetls">
- <synopsis></synopsis>
+ <synopsis>Whether to use TLS for the connection or not</synopsis>
</configOption>
<configOption name="usesasl">
- <synopsis></synopsis>
+ <synopsis>Whether to use SASL for the connection or not</synopsis>
</configOption>
<configOption name="forceoldssl">
- <synopsis></synopsis>
+ <synopsis>Force the use of old-style SSL for the connection</synopsis>
</configOption>
<configOption name="keepalive">
<synopsis></synopsis>
</configOption>
<configOption name="autoprune">
- <synopsis></synopsis>
+ <synopsis>Auto-remove users from buddy list.</synopsis>
+ <description><para>Auto-remove users from buddy list. Depending on the setup
+ (e.g., using your personal Gtalk account for a test) this could cause loss of
+ the contact list.
+ </para></description>
</configOption>
<configOption name="autoregister">
- <synopsis></synopsis>
+ <synopsis>Auto-register users bfrom buddy list</synopsis>
</configOption>
<configOption name="auth_policy">
- <synopsis></synopsis>
+ <synopsis>Whether to automatically accept or deny users' subscription requests</synopsis>
</configOption>
<configOption name="sendtodialplan">
- <synopsis></synopsis>
+ <synopsis>Send incoming messages into the dialplan</synopsis>
</configOption>
<configOption name="status">
- <synopsis></synopsis>
+ <synopsis>XMPP status-chat, available, away, xaway, or dnd</synopsis>
</configOption>
<configOption name="buddy">
- <synopsis></synopsis>
+ <synopsis>Manual addition of buddy to list</synopsis>
+ <description><para>
+ Manual addition of buddy to the buddy list. For distributed events, these budies are
+ automatically added in the whitelist as 'owners' of the node(s).
+ </para></description>
</configOption>
</configObject>
</configFile>
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