The following patch with updates for trunk. Works much better in trunk.
authorOlle Johansson <oej@edvina.net>
Tue, 27 Nov 2007 19:24:17 +0000 (19:24 +0000)
committerOlle Johansson <oej@edvina.net>
Tue, 27 Nov 2007 19:24:17 +0000 (19:24 +0000)
commitdf7ba90b2085bab375774687938b98725db07ae8
tree010456feea8c290b3ad46768771d566dba273121
parent35ecd08b46453971d59020099fb638c81afb1312
The following patch with updates for trunk. Works much better in trunk.
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...

Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines

If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.

With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.

(closes issue #11376)
Reported by: lasse
Patches:
      bug11376.txt uploaded by oej (license 306)
Tested by: lasse

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/rtp.h
main/rtp.c