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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!

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-08-31 01:59:02 +00:00
parent 5418a7a1a8
commit c6977b9983
29 changed files with 744 additions and 365 deletions
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6
UPGRADE.txt
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@@ -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.