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authorfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-08-31 01:59:02 +0000
committerfile <file@f38db490-d61c-443f-a65b-d21fe96a405b>2006-08-31 01:59:02 +0000
commit3f22aa53af0a7bb3efb280a1adda11f00a7f68fc (patch)
tree95a924d57b906428b26d398758facf98b92bdcb2 /UPGRADE.txt
parent0fcb352fba60344329a582f891ddd4433f48c6fa (diff)
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
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diff --git a/UPGRADE.txt b/UPGRADE.txt
index 9a75748a8..f0e7246b7 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -303,6 +303,12 @@ The SIP channel:
option in sip.conf is removed to osp.conf as authpolicy. allowguest option
in sip.conf cannot be set as osp anymore.
+* The Asterisk RTP stack has been changed in regards to RFC2833 reception
+ and transmission. Packets will now be sent with proper duration instead of all
+ at once. If you are receiving calls from a pre-1.4 Asterisk installation you
+ will want to turn on the rfc2833compensate option. Without this option your
+ DTMF reception may act poorly.
+
* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
in coming versions of Asterisk. Please use the dialplan function
SIPCHANINFO(useragent) instead.