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2009-06-01Add the ability to execute connected line interception macros.Mark Michelson
When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵Mark Michelson
Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 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
2008-10-30fix a few small things found by using sparseKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03The dialing API should inherit datastores as well as variablesTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-13Convert one more delimiter to use comma.Tilghman Lesher
(closes issue #12850) Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded by Corydon76 (license 14) Tested by: bcnit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01Modify TIMEOUT() to be accurate down to the millisecond.Tilghman Lesher
(closes issue #10540) Reported by: spendergrass Patches: 20080417__bug10540.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20Add missing unlockMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-16Merged revisions 108961 via svnmerge from Michiel van Baak
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Merged revisions 106235 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04Whitespace changes onlyTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28Merged revisions 104841 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105060 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
2008-01-24Test hopefully over.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-24Testing something...Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-16Merged revisions 98960 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 lines Introduce a lock into the dialing API that protects it when destroying the structure. (closes issue #11687) Reported by: callguy Patches: 11687.diff uploaded by file (license 11) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-21AST_LIST_REMOVE_CURRENT only takes one argument in trunkMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-21Merged revisions 94468 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines Since we are freeing list elements within a list traversal, we need to use the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Merged revisions 89610 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove a bunch of useless #include "options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-10Bring up to date with poll changes.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79074 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
2007-07-26Do a massive conversion for using the ast_verb() macroRussell Bryant
(closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵Tilghman Lesher
guidelines changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Small doxygen updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-28Merge changes from team/russell/eventsRussell Bryant
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
2007-04-24Merged revisions 61774 via svnmerge from Russell Bryant
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
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
2007-02-22Merged revisions 56277 via svnmerge from Russell Bryant
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
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 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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