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authorMatthew Jordan <mjordan@digium.com>2014-03-28 17:41:23 +0000
committerMatthew Jordan <mjordan@digium.com>2014-03-28 17:41:23 +0000
commit597f25db69c71d342e8661983df47e457625b5e2 (patch)
treed39f3071d46b4f2c3cf8082e4b0c2503747b47e2 /CHANGES
parenta438a0e65fa183860ed04eb1487ecd991db57225 (diff)
Update API versions and UPGRADE/CHANGES for 12.2.0
This patch does the following: * It updates the AMI version to 2.2.0 to indicate backwards compatible changes have been made since the last release * It updates the ARI version to 1.2.0 to indicate backwards compatible changes have been made since the last release * It updates the UPGRADE/CHANGES files with changes that were not mentioned ........ Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES35
1 files changed, 35 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 2551c0b5b..3b8461e48 100644
--- a/CHANGES
+++ b/CHANGES
@@ -133,11 +133,46 @@ AMI
second channel when dialing LOCAL, or defaults to appending ;2 if only
the single Id is given.
+RealTime
+------------------
+ * A new set of Alembic scripts has been added for CDR tables. This will create
+ a 'cdr' table with the default schema that Asterisk expects.
+
res_pjsip
------------------
* transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
+ * Added the following new CLI commands:
+ - "pjsip show contacts" - list all current PJSIP contacts.
+ - "pjsip show contact" - show specific information about a current PJSIP
+ contact.
+ - "pjsip show channel" - show detailed information about a PJSIP channel.
+
+res_pjsip_multihomed
+------------------
+ * A new module, res_pjsip_multihomed handles situations where the system
+ Asterisk is running out has multiple interfaces. res_pjsip_multihomed
+ determines which interface should be used during message sending.
+
+res_pjsip_pidf_digium_body_supplement
+------------------
+ * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
+ request body formatting for presence support in Digium phones.
+
+res_pjsip_send_to_voicemail
+------------------
+ * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
+ particular headers to transfer a PJSIP channel directly to a particular
+ extension that has VoiceMail. This is intended to be used with Digium
+ phones that support this feature.
+
+res_pjsip_outbound_registration
+------------------
+ * A new CLI command has been added: "pjsip show registrations", which lists
+ all configured PJSIP registrations
+
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
------------------------------------------------------------------------------