asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	7019ff68db	Fixed crashes from issue8824 review board channel locking changes. The local struct ast_party_connected_line connected_caller variable was uninitialized when the copy function was called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	1ed5422fa9	Merged revisions 192429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 \| file \| 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) \| 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	17dad4e5b0	Commit documentation changes related to issue #14801 . (issue #14801) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	a3af213506	Remove rarely-used event_log/LOG_EVENT support In discussions today at the Europe Asterisk Developer Meet-Up, we determined that the event_log was used in only 9 places in the entire tree, and really was not needed at all. The users have been converted to use LOG_NOTICE, or the messages have been removed since other messages were already in place that provided the same information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	8612c7ac8a	Made security features optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	38720ccd3b	Added routing number support. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	236920485d	Fixed not report source network ID and not export destination network ID issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a866a75900	Merge str_substitution branch. This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1e016da893	Fix app_queue XML documentation. I think it would behoove us to force "make validate-docs" to be run after the XML documentation has been generated if dev-mode is enabled. (closes issue #14989) Reported by: tzafrir Patches: app_queue_xml.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	2e0f8bcbc8	Updated for OSP Toolkit 3.5. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	1d941ad821	Allow for a position to be specified when entering a queue. This would allow for one to add a caller to a specific place in the queue instead of just placing the caller in the back every time. To help facilitate some interesting manipulations, a new channel variable called QUEUEPOSITION has been added. When a caller is removed from a queue, his position in that queue is stored in the QUEUEPOSITION variable. One such strategy an administrator can employ is to allow for the removal of a caller from one queue followed by the insertion of the same caller into a separate queue in the same position. Review: http://reviewboard.digium.com/r/189 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	09cde5a40c	Update warning message to not have pipes and contain all options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	8f81deab25	Fix reversed behavior of leavewhenempty option in queues.conf. (closes issue #14650) Reported by: alecdavis Patches: 14650.patch uploaded by mmichelson (license 60) Tested by: mmichelson, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	ada8ae56e1	Fix a double free issue with the Pickup dialplan application. As part of the pickup process the connected line information is updated. Part of this process does a shallow copy of the target channel's connected line information to a local structure. Once complete the structure contents are freed. As a result any information in the target channel's connected line information structure is no longer valid. This change will now set the contents back to a clean state so that the freeing of the target channel's connected line information structure when the channel is destroyed will no longer try to double free things. (closes issue #14839) Reported by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	7164958d9d	Merged revisions 189465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189465 \| twilson \| 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) \| 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is set ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	f505cb43bf	Merged revisions 189463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189463 \| twilson \| 2009-04-20 16:00:52 -0500 (Mon, 20 Apr 2009) \| 2 lines Don't treat a NOANSWER like a CHANUNAVAIL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0adb04fbbb	Merged revisions 188773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188773 \| tilghman \| 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) \| 4 lines Umask should not be exported into global namespace. (closes issue #14912) Reported by: jcapp ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188774 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fd833f14ed	Make the cancellation of the dial timeout on a call forward optional. This introduces the 'z' option to app_dial. With it set, a call forward will cancel any timeout originally set for this instance of the Dial application. AST-207 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f26878feb2	Fix a couple of queue member reference leaks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	bb03eef676	Making sure we have references to external libraries. Note: Update h.323 with the recent changes too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b6a2f40793	Set all queue variables on both the caller and member channels. This allows for the variables to be accessed if a member macro is run. Thanks to Grigoriy Puzankin for bringing this up on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	df28954a84	Make sure tc is unlocked before calling ast_call since calling a Local channel could result in a deadlock. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	bd23adbc8a	Even more changes concerning r187426. Revised where locks are placed yet once again. ast_call() should not be called with a channel locked. could cause deadlock issues with local channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2f048030bd	revert addition of LOG_SECURITY log channel; after further discussion, a much better solution will be used git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	e6052e79d0	More changes concerning r187426. Revised where locks are placed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	de4af72f9f	Add ability for dialplan execution to continue when caller hangs up. The F option to app_dial has been modified to accept no parameters and perform the above functionality. I don't see anywhere else that is doing function overloading, but this really is the best place for this operation because: - It makes it close to the 'g' option in the argument list which provides similar functionality. - The existing code to support the current F option provides a very convienient location to add this new feature. (closes issue #12381) Reported by: michael-fig git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	19f381b484	Fixes deadlock caused by calling get_cid_name with chan locked. get_cid_name should not be called with a channel lock. get_cid_name calls ast_get_hint which eventually calls pbx_find_extension. pbx_find_extension starts and stops autoservice which should not be done with a channel lock, so get_cid_name should not be called with one. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3a220874cc	Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 \| tilghman \| 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) \| 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	b5f8c632df	add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts (inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e36b632b4a	Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 \| tilghman \| 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) \| 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into 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	16 years ago
Tilghman Lesher	a3c84f9575	Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 \| tilghman \| 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) \| 2 lines Found a conflict in the last commit, due to multiple targets ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	06061491ba	Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 \| tilghman \| 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) \| 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5c53b2226d	Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f43159ba31	Fix trunk's compilation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5c0d934e6b	Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 \| mmichelson \| 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) \| 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	de8a0946c8	Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 \| mmichelson \| 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) \| 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c9c8758d6d	Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	c4e3bfb74c	Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 \| mmichelson \| 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) \| 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	47c3799c99	Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 \| russell \| 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) \| 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	383d4fa05b	Fix a typo in app_ices. (closes issue #14765) Reported by: timeshell Patches: app_ices.svn-1.6.0.diff uploaded by timeshell (license 399) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f745326750	Use ast_random() instead of rand() to ensure we use the best RNG available. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2a4f9f7181	Change global_app_buf to ast_str_thread_global_buf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba0ce88b1e	Merged revisions 184388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184388 \| dvossel \| 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) \| 8 lines pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ee77b475f2	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/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a7028f2bc6	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/trunk@184079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9d3527bddf	Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 \| dvossel \| 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) \| 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b52d2dae2e	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/trunk@183244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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