[asterisk-commits] CHANGES: Update formatting of items (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 18 18:35:32 CDT 2016
Joshua Colp has submitted this change and it was merged.
Change subject: CHANGES: Update formatting of items
......................................................................
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
---
M CHANGES
1 file changed, 52 insertions(+), 38 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
Richard Mudgett: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, but someone else must approve; Verified
diff --git a/CHANGES b/CHANGES
index 15b4d0c..9c60b24 100644
--- a/CHANGES
+++ b/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
- is similar to origination except that the specified channel is not dialed. This
- allows for an application writer to create a channel, perform manipulations on it,
- and then delay dialing the channel until later.
+ you to create a new channel and place that channel into a Stasis application.
+ This is similar to origination except that the specified channel is not
+ dialed. This allows for an application writer to create a channel, perform
+ 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
- resource in order to place a call to a created channel.
+ * To complement the "create" method, a "dial" method has been added to the
+ 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
@@ -31,6 +31,7 @@
"next_media_uri", which specifies the next media URI in the list to be played
back to the resource. The "PlaybackFinished" event is raised when all media
URIs are done.
+
Applications
------------------
@@ -73,25 +74,22 @@
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
file. This is needed for those countries with privacy laws that require
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 @@
------------------
* 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 @@
* 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 @@
* 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_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.
- * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
- through using SIP re-invites with sendonly and sendrecv accordingly.
+ to the request URI and From URI if the user is determined to be a phone
+ number.
+
+ * 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.
- If this information is present on messages received from the remote server
- the message will automatically be associated with the configured endpoint on the
- outbound registration.
+
+ * New 'line' and 'endpoint' options added on outbound registrations. This
+ allows some identifying information to be added to the Contact of the
+ outbound registration. If this information is present on messages received
+ from the remote server the message will automatically be associated with the
+ configured endpoint on the outbound registration.
+
Core
------------------
@@ -190,6 +197,7 @@
context. If enabled then a hint will be automatically created with the name of
the device.
+
Functions
------------------
@@ -208,8 +216,9 @@
DTMF Features
------------------
* The transferdialattempts default value has been changed from 1 to 3. The
- transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
- These were changed to make DTMF transfers be more user-friendly by default.
+ transferinvalidsound has been changed from "pbx-invalid" to
+ "privacy-incorrect". These were changed to make DTMF transfers be more
+ user-friendly by default.
Resources
@@ -250,6 +259,7 @@
outbound registration, registration is retried at the given interval up to
'max_retries'.
+
CEL Backends
------------------
@@ -262,6 +272,7 @@
configurable for cel_pgsql via the 'schema' in configuration file
cel_pgsql.conf.
+
CDR Backends
------------------
@@ -272,15 +283,18 @@
names. This setting is configurable for cdr_adaptive_odbc via the
quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
-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'.
+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'.
+
+
+------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
------------------------------------------------------------------------------
--
To view, visit https://gerrit.asterisk.org/2842
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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