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path: root/include/asterisk/dial.h
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2013-12-09app_page: Add predial handlers for app_page.Jonathan Rose
(closes issue AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-22Add dial events to app_queue and app_followme.Jason Parker
Also fixes an issue in app_dial, where the channels were swapped on dial events. (closes issue ASTERISK-21551) (closes issue ASTERISK-21550) Review: https://reviewboard.asterisk.org/r/2549/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-18Move origination to use the dialing API and send Stasis messages on dial ↵Joshua Colp
begin and end. (closes issue ASTERISK-21549) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2512/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-10Transition app_page to using app_confbridge internally for the conference ↵Joshua Colp
bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles. Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsMark Michelson
had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-25Add an API call that steals the answered channel so that a destruction of ↵Joshua Colp
the dialing structure does not hang it up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30Add support for call forwarding and timeouts to the dialing API.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Add an option to the dial API for playing music instead of ringing to the ↵Russell Bryant
caller. I started this for use with SLA but ended up deciding not to use it. However, there is no reason not to put this part in, anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54103 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines Change ast_set_state_callback() to ast_dial_set_state_callback() ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54066 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10Merged revisions 53810 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24Merged revisions 52107 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24 Jan 2007) | 3 lines Fix the formatting of doxygen comments to properly indicate that the comment documents the previous entity, as opposed to the next one. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24Merged revisions 52049 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines Merge in dialing API and the app_page that uses it. (issue #BE-118) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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