CHANGES: Update formatting of items

* Provide consistent indenting of lines in bulleted paragraphs
* Respect the 80 character column width
* Group all like items together, e.g., all dialplan applications under
  "Applications", etc.
* Use a single blank line to break up functionality changes within a
  larger section
* Use two blanks lines to delineate larger sections

Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647

CHANGES

@@ -15,13 +15,13 @@
 ARI
 -----------------
  * A new ARI method has been added to the channels resource. "create" allows for
- you to create a new channel and place that channel into a Stasis application. This
+   you to create a new channel and place that channel into a Stasis application.
- is similar to origination except that the specified channel is not dialed. This
+   This is similar to origination except that the specified channel is not
- allows for an application writer to create a channel, perform manipulations on it,
+   dialed. This allows for an application writer to create a channel, perform
- and then delay dialing the channel until later.
+   manipulations on it, and then delay dialing the channel until later.
 
- * To complement the "create" method, a "dial" method has been added to the channels
+ * To complement the "create" method, a "dial" method has been added to the
- resource in order to place a call to a created channel.
+   channels resource in order to place a call to a created channel.
 
  * All operations that initiate playback of media on a resource now support
    a list of media URIs. The list of URIs are played in the order they are
@@ -32,6 +32,7 @@ ARI
    back to the resource. The "PlaybackFinished" event is raised when all media
    URIs are done.
 
+
 Applications
 ------------------
 
@@ -73,6 +74,17 @@ Playback
    provided, including the file extension. Currently, on HTTP and HTTPS URI
    schemes are supported.
 
+Queue
+-------------------
+ * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
+   the queue member was paused.
+
+ * Added field LastPause on QueueMemberStatus for time when started the last
+   pause for a queue member.
+
+ * Show the time when started the last pause for queue member on CLI for command
+   'queue show'.
+
 SMS
 ------------------
  * Added the 'n' option, which prevents the SMS from being written to the log
@@ -80,20 +92,6 @@ SMS
    providers to not log SMS content.
 
 
-CDRs
-------------------
-cdr_odbc
-------------------
- * Added a new configuration option, "newcdrcolumns", which enables use of the
-   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
-
-------------------
-cdr_csv
-------------------
- * Added a new configuration option, "newcdrcolumns", which enables use of the
-   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
-
-
 Channel Drivers
 ------------------
 
@@ -101,6 +99,7 @@ chan_dahdi
 ------------------
  * The CALLERID(ani2) value for incoming calls is now populated in featdmf
    signaling mode.  The information was previously discarded.
+
  * Added the force_restart_unavailable_chans compatibility option.  When
    enabled it causes Asterisk to restart the ISDN B channel if an outgoing
    call receives cause 44 (Requested channel not available).
@@ -110,6 +109,7 @@ chan_iax2
  * The iax.conf forcejitterbuffer option has been removed.  It is now always
    forced if you set iax.conf jitterbuffer=yes.  If you put a jitter buffer
    on a channel it will be on the channel.
+
  * A new configuration parameters, 'calltokenexpiration', has been added that
    controls the duration before a call token expires. Default duration is 10
    seconds. Setting this to a higher value may help in lagged networks or those
@@ -120,9 +120,11 @@ chan_sip
  * New 'rtpbindaddr' global setting. This allows a user to define which
    ipaddress to bind the rtpengine to. For example, chan_sip might bind
    to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
+
  * DTLS related configuration options can now be set at a general level.
    Enabling DTLS support, though, requires enabling it at the user
    or peer level.
+
  * Added the possibility to set the From: header through the the SIP dial
    string (populating the fromuser/fromdomain fields), complementing the
    [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
@@ -132,17 +134,22 @@ chan_sip
 chan_pjsip
 ------------------
  * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
-   to the request URI and From URI if the user is determined to be a phone number.
+   to the request URI and From URI if the user is determined to be a phone
- * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
+   number.
-   through using SIP re-invites with sendonly and sendrecv accordingly.
+
+ * New 'moh_passthrough' endpoint setting. This will pass hold and unhold
+   requests through using SIP re-invites with sendonly and sendrecv accordingly.
+
  * Added the pjsip.conf system type disable_tcp_switch option.  The option
    allows the user to disable switching from UDP to TCP transports described
    by RFC 3261 section 18.1.1.
- * New 'line' and 'endpoint' options added on outbound registrations. This allows some
+
-   identifying information to be added to the Contact of the outbound registration.
+ * New 'line' and 'endpoint' options added on outbound registrations. This
-   If this information is present on messages received from the remote server
+   allows some identifying information to be added to the Contact of the
-   the message will automatically be associated with the configured endpoint on the
+   outbound registration. If this information is present on messages received
-   outbound registration.
+   from the remote server the message will automatically be associated with the
+   configured endpoint on the outbound registration.
+
 
 Core
 ------------------
@@ -190,6 +197,7 @@ Core
    context. If enabled then a hint will be automatically created with the name of
    the device.
 
+
 Functions
 ------------------
 
@@ -208,8 +216,9 @@ CURL
 DTMF Features
 ------------------
  * The transferdialattempts default value has been changed from 1 to 3. The
-   transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
+   transferinvalidsound has been changed from "pbx-invalid" to
-   These were changed to make DTMF transfers be more user-friendly by default.
+   "privacy-incorrect". These were changed to make DTMF transfers be more
+   user-friendly by default.
 
 
 Resources
@@ -250,6 +259,7 @@ res_pjsip_outbound_registration
   outbound registration, registration is retried at the given interval up to
   'max_retries'.
 
+
 CEL Backends
 ------------------
 
@@ -262,6 +272,7 @@ cel_pgsql
    configurable for cel_pgsql via the 'schema' in configuration file
    cel_pgsql.conf.
 
+
 CDR Backends
 ------------------
 
@@ -272,15 +283,18 @@ cdr_adaptive_odbc
    names. This setting is configurable for cdr_adaptive_odbc via the
    quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
 
-Queue
+cdr_odbc
--------------------
+------------------
- * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
+ * Added a new configuration option, "newcdrcolumns", which enables use of the
-   the queue member was paused.
+   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
- * Added field LastPause on QueueMemberStatus for time when started the last
-   pause for a queue member.
- * Show the time when started the last pause for queue member on CLI for command
-   'queue show'.
 
+cdr_csv
+------------------
+ * Added a new configuration option, "newcdrcolumns", which enables use of the
+   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
+
+
+------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
 ------------------------------------------------------------------------------