asterisk

Commit Graph

Author	SHA1	Message	Date
Olle Johansson	e509d2709f	Small doxygen updates git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b6b1bf3213	Merge changes from team/russell/events This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	90f152b24f	Merged revisions 61774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61774 \| russell \| 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) \| 5 lines Add a few more state changes in handle_frame_ownerless() so that the SLA code will get notified of these changes even when an owner channel is not provided. This isn't from a specific bug report, it's just something I noticed while poking around. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ab31bec392	Add an option to the dial API for playing music instead of ringing to the 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	18 years ago
Russell Bryant	9138e53bc9	Merged revisions 56277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56277 \| russell \| 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) \| 18 lines Merge changes from team/russell/sla_updates. This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	8f8df3e5a9	Merged revisions 54103 via svnmerge from 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	19 years ago
Russell Bryant	2a5477b35e	Merged revisions 54066 via svnmerge from 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	19 years ago
Russell Bryant	5715b49c30	Merged revisions 53810 via svnmerge from 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	19 years ago
Joshua Colp	9826fc599b	Merged revisions 52049 via svnmerge from 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	19 years ago

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59 Commits (e5c409da6e9a0651fa4041e5a02d916aadec858f)