From c6977b9983db4f58446bfbc65a5b028cda8244ee Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Thu, 31 Aug 2006 01:59:02 +0000 Subject: 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: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- UPGRADE.txt | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'UPGRADE.txt') 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. -- cgit v1.2.3