asterisk

Commit Graph

Author	SHA1	Message	Date
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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 years ago
David Vossel	2764c2821f	Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 \| dvossel \| 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) \| 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bdd99ad64	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/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	e559cae4ec	Option to send DTMF when receiving PROGRESS status The D() option in app_dial is only able to send DTMF after the call has been answered. A progress option has been added to D() to allow DTMF to be sent upon receiving PROGRESS. This allows DTMF to be sent before the call is answered. (closes issue #12123) Reported by: VoipForces Patches: app_dial.c_patch_trunk_valid uploaded by VoipForces (license 419) dtmf_progress.patch uploaded by dvossel (license 671) Tested by: VoipForces, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 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	17 years ago

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3443 Commits (25cea89d90efca01c69f5749e4ee01b38b0c96b1)