mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-22 12:52:33 +00:00
Note jitterbuffer support for chan_local in CHANGES
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
12
CHANGES
12
CHANGES
@@ -141,9 +141,9 @@ Voicemail Changes
|
||||
* SMDI is now enabled in voicemail using the smdienable option.
|
||||
* A "lockmode" option has been added to asterisk.conf to configure the file
|
||||
locking method used for voicemail, and potentially other things in the
|
||||
future. The default is the old behavior, lockfile. However, there is a
|
||||
new method, "flock", that uses a different method for situations where the
|
||||
lockfile will not work, such as on SMB/CIFS mounts.
|
||||
future. The default is the old behavior, lockfile. However, there is a
|
||||
new method, "flock", that uses a different method for situations where the
|
||||
lockfile will not work, such as on SMB/CIFS mounts.
|
||||
|
||||
Queue changes
|
||||
-------------
|
||||
@@ -289,3 +289,9 @@ Miscellaneous
|
||||
to just UNKNOWN if the extension exists.
|
||||
* When originating a call using AMI or pbx_spool that fails the reason for failure
|
||||
will now be available in the failed extension using the REASON dialplan variable.
|
||||
* Added jitterbuffer support for chan_local. This allows you to use the
|
||||
generic jitterbuffer on incoming calls going to Asterisk applications.
|
||||
For example, this would allow you to use a jitterbuffer for an incoming
|
||||
SIP call to Voicemail by putting a Local channel in the middle. This
|
||||
feature is enabled by using the 'j' option in the Dial string to the Local
|
||||
channel in conjunction with the existing 'n' option for local channels.
|
||||
|
Reference in New Issue
Block a user