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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2002-03-15 14:58:01 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2002-03-15 14:58:01 +0000
commite55a55fb0ea8753f36febbc49fb7f08e195e0cbd (patch)
treed1919f5f8a590ca65cae03708cca45fc3c836cf8
parent814c6e3a554b450a50579c58f14928ea54dd7d93 (diff)
Version 0.1.11 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@427 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-xCHANGES8
1 files changed, 8 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 36b222c79..51eb71205 100755
--- a/CHANGES
+++ b/CHANGES
@@ -1,4 +1,12 @@
+Asterisk 0.1.11
+ -- Add ISDN RAS capability
+ -- Add stutter dialtone to Chan Zap
+ -- Add "#include" capability to config files.
+ -- Add call-forward variable to Chan Zap (*72, *73)
+ -- Optimize IAX flow when transfer isn't possible
+ -- Allow transmission of ANI over IAX
Asterisk 0.1.10
+ -- Make ast_readstring parameter be the max # of digits, not the max size with \0
-- Make up any missing messages on the fly
-- Add support for specific DTMF interruption to saying numbers
-- Add new "u" and "b" options to condense busy/unavail handling