diff options
| author | Joshua Colp <jcolp@digium.com> | 2006-08-31 01:59:02 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2006-08-31 01:59:02 +0000 |
| commit | c6977b9983db4f58446bfbc65a5b028cda8244ee (patch) | |
| tree | 95a924d57b906428b26d398758facf98b92bdcb2 /UPGRADE.txt | |
| parent | 5418a7a1a8326544f2275819c93649c97261a5f3 (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: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'UPGRADE.txt')
| -rw-r--r-- | UPGRADE.txt | 6 |
1 files changed, 6 insertions, 0 deletions
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. |