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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-10-09 15:12:59 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2007-10-09 15:12:59 +0000
commitcf9d2ba42b43caa26d5e5f0a0cb9eb807506cd6c (patch)
tree6a9b8fd835371073272027ba4a20f628b985b719 /CHANGES
parent4c29367323959fe219d1657610eec6a236d8f67b (diff)
Note jitterbuffer support for chan_local in CHANGES
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85098 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES12
1 files changed, 9 insertions, 3 deletions
diff --git a/CHANGES b/CHANGES
index 6a37d2061..71f4f1b9d 100644
--- a/CHANGES
+++ b/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.