Commit graph

6 commits

Author SHA1 Message Date
Terry Wilson
a9d607a357 Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-29 16:52:47 +00:00
Terry Wilson
57f42bd74f ast_channel opaquification of pointers and integral types
Review: https://reviewboard.asterisk.org/r/1753/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Richard Mudgett
cbe57b11cb Fixes for sending SIP MESSAGE outside of calls.
* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.

* Pass up better From header contents for SIP to use.  Now is in the
"display-name" <URI> format expected by MessageSend.  (Note that this is a
behavior change that could concievably affect some people.)

* Block user from adding standard headers that are added automatically.
(To, From,...)

* Allow the user to override the Content-Type header contents sent by
MessageSend.

* Decrement Max-Forwards header if the user transferred it from an
incoming message.

* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.

* Documents what SIP expects in the MessageSend(from) parameter.

(closes issue ASTERISK-18992)
Reported by: Yuri

(closes issue ASTERISK-18917)
Reported by: Shaun Clark

Review: https://reviewboard.asterisk.org/r/1683/
........

Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-25 17:23:25 +00:00
Leif Madsen
eb37d38b7d Update documentation for MESSAGE_SEND_STATUS variable.
(Closes issue ASTERISK-19056)
Reported by: Yuri
Patches:
     348360.diff uploaded by Yuri (license #5242)
........

Merged revisions 348605 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-12-19 19:55:18 +00:00
David Vossel
d7dec4f14f Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines
  
  Fixes memory leak in message API.
  
  The ast_msg_get_var function did not properly decrement
  the ref count of the var it retrieves.  The way this is
  implemented is a bit tricky, as we must decrement the var and then
  return the var's value.  As long as the documentation for the
  function is followed, this will not result in a dangling pointer as
  the ast_msg structure owns its own reference to the var while it
  exists in the var container.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-11-29 00:03:36 +00:00
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