3411 Commits (36edf175df5945f34e18fa05bc26a213d2f9236d)

Author	SHA1	Message	Date
Russell Bryant	36edf175df	Merged revisions 184726 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	64f1321f14	Merged revisions 184693 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 | russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines Change global_app_buf to ast_str_thread_global_buf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c2f4651a4b	Merged revisions 184339 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ffbca9c0bb	Merged revisions 184079 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e4ae79101d	Merged revisions 183244 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@183247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b085426cda	Merged revisions 182847 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@182947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b4fcc4a098	Change faulty comparison used when announcing average hold minutes and seconds ... (closes issue #14227) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a1a9006163	Run the macro on the queue member's channel when he answers, not the caller's channel. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	499ca8de65	Fix crash when sleep and retries argument was not given to RetryDial application. ... (closes issue #14647) Reported by: sherpya git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d7d817d687	Fix segfault when dialing a typo'd queue ... If trying to dial a non-existent queue, there would be a segfault when attempting to access q->weight, even though q was NULL. This problem was introduced during the queue-reset merge and thus only affects trunk. (closes issue #14643) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	3a92673356	Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet ... option is enabled. (reported by Vadim Lebedev on the asterisk-dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	09df92d485	Merged revisions 180567 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	96405af1a8	Merged revisions 180464 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e69803a2be	Merged revisions 180380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	4c9ab0df8c	Merge phase 1 support for the new bridging architecture. ... This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	15090ba1df	Remove duplicate 'k' and 'K' Dial options. ... (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	979eb709ae	app_read does not break from prompt loop with user terminated empty string ... In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	8970f8caaa	Merged revisions 180006 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ae786501f1	app_meetme not setting filename and fileformat correctly for realtime ... When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brian Degenhardt	a7092f0acc	fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	8f6a933e30	Merged revisions 179532 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	932cef55fd	Swap reversed timevals. ... This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	22ea970494	These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	345a6fd1cb	Permit emailsubject and emailbody to be set per mailbox. ... (closes issue #14372) Reported by: fhackenberger Patches: voicemail_individual_subject_and_body_1.6.1 uploaded by fhackenberger (license 592) with additional fixes by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Dwayne M. Hubbard	83bbf83f49	Make app_fax compatible with spandsp-0.0.6pre4 ... Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3af1c558df	Allow semicolons to be escaped, when passing arguments to the System command. ... (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	fefa97001a	Oops, merge broke trunk ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	306f229d64	Merged revisions 177536 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2e2494591e	Merged revisions 177383 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8065dc57af	Re-add 'o' option to MeetMe, reverting rev 62297. ... Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	d13cdd43a4	Remove a dependency that no longer exists. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	184872fdfd	Fix a race condition that caused device states to become incorrect for hints. ... The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	20c5bad34b	Merged revisions 176249,176252 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3c9667ae12	Merge queue-reset branch to Asterisk ... From a user point-of-view, this adds new CLI commands and Manager Actions to better facilitate the reloading of queues and the resetting of their statistics. The new CLI commands are the "queue reload" and "queue reset stats" commands. The new manager actions are the QueueReload and QueueReset commands. Review: http://reviewboard.digium.com/r/115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b5bde28d9f	Add manager events for chanspy starting or stopping ... (closes issue #14469) Reported by: caio1982 Patches: chanspy_events2.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	10281bf183	Merged revisions 175590 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0cfdaa2dc0	Add an option to keep the recorded file upon hangup. ... (closes issue #14341) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0d5da5f436	Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	34161542e9	Fix odd "thank you" sound playing behavior in app_queue.c ... If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	47ebea6a8d	Fix 'd' option for app_dial and add new option to Answer application ... The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c9670b24a4	Fix potential for stack overflows in app_chanspy.c ... When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d4df1c6a78	Fix0ring build ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	51c80f5b40	Remove the usage of the KeepAlive app, as it no longer exists. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	3084e9ce93	This patch removes the use of AST_PBX_KEEPALIVE ... from app_rpt.c. (closes issue #14435) Reported by: D_McNaul git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	43462557c0	More intptr_t work. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	dbe56b6647	Merged revisions 174369 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	829b14fc3e	Fixes issue with hangups not being sent and external process never terminating. ... The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b2484d7db9	Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. ... (closes issue #14414) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8f9c8f8b40	Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage ... (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago