diff --git a/CHANGES b/CHANGES
index 15b4d0c670d0509819900f921db9443412b28ef1..9c60b2443ccfc7d9a14403648d77348c179c3959 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
@@ -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.
- * 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 @@ 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".
- 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 @@ 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
--------------------
- * 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 -----------
------------------------------------------------------------------------------