asterisk/channels/sip
Russell Bryant 3f4d0e8743 Support routing text messages outside of a call.
Asterisk now has protocol independent support for processing text messages
outside of a call.  Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported.  There are options in sip.conf
and jabber.conf that enable these features.

There is a new application, MessageSend().  There are two new functions,
MESSAGE() and MESSAGE_DATA().  Documentation will be available on
the project wiki, wiki.asterisk.org.

Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.

Review: https://reviewboard.asterisk.org/r/1042/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-01 21:31:40 +00:00
..
include Support routing text messages outside of a call. 2011-06-01 21:31:40 +00:00
config_parser.c Merged revisions 281687 via svnmerge from 2010-08-11 13:31:39 +00:00
dialplan_functions.c Merged revisions 310088 via svnmerge from 2011-03-08 20:34:05 +00:00
reqresp_parser.c Merged revisions 321273 via svnmerge from 2011-05-27 16:35:49 +00:00
sdp_crypto.c Merged revisions 317474 via svnmerge from 2011-05-05 22:44:52 +00:00
srtp.c Add SRTP support for Asterisk 2010-06-08 05:29:08 +00:00