asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	b1f34068ac	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@352520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	0abad2d894	Update documentation for MESSAGE_SEND_STATUS variable. (Closes issue ASTERISK-19056) Reported by: Yuri Patches: 348360.diff uploaded by Yuri (license #5242) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@348605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	5d452c2647	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/branches/10@346349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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	15 years ago

4 Commits (98658b0670a4374be271aed0c7b42657cfced657)