4822 Commits (e9520dbe0db9a4f3c5c2cd90a9740880b18d1e90)

Author	SHA1	Message	Date
Matthew Jordan	e9520dbe0d	clang compiler warnings: Fix -Wparantheses-equality warnings ... Clang will treat ((a == b)) as a warning, as it reasonably expects that the developer may have intended to write (a == b) or ((a = b)). This patch cleans up all instances where equality, not assignment, was intended between two parantheses. Review: https://reviewboard.asterisk.org/r/4531/ ASTERISK-24917 Repoted by: dkdegroot patches: rb4531.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433688 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d6173cd1d0	clang compiler warnings: Fix -Wunused-function; make inline function static ... This patch fixes clang compilers warnings for unused functions. Specifically: * channels/chan_iax2: removed user_ref function * main/dsp.c: removed goertzel_update function * main/config.c: made variable_list_switch static Review: https://reviewboard.asterisk.org/r/4527 ASTERISK-24917 Reported by: dkdegroot patches: rb4527.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433680 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	28e3bd0af7	Improved and portable ast_log recursion avoidance ... This introduces a new logger routine ast_log_safe. This routine should be used for all error messages in code that can be run as a result of ast_log. ast_log_safe does nothing if run recursively. All error logging in astobj2.c, strings.c and utils.h have been switched to ast_log_safe. This required adding support for raw threadstorage. This provides direct access to the void* pointer in threadstorage. In ast_log_safe, NULL is used to signify that this thread is not already running ast_log_safe, (void*)1 when it is already running. This was done since it's critical that ast_log_safe do nothing that could log during recursion checking. ASTERISK-24155 #close Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/4502/ ........ Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433523 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	554eb74516	Fix compile errors caused by r4500 / r4501. ... * Add ast_register_cleanup to utils/clicompat.c to deal with any utils that copy sources from main. * Asterisk 13+: remove unused variables from core_local.c. Review: https://reviewboard.asterisk.org/r/4534/ ........ Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433500 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	3ddd92902a	Replace most uses of ast_register_atexit with ast_register_cleanup. ... Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	ab674f67b5	app_confbridge: file playback blocks dtmf ... Attempting to execute DTMF in a confbridge while file playback (prompt, announcement, etc) is occurring is not allowed. You have to wait until the sound file has completed before entering DTMF. This patch fixes it so that app_confbridge now monitors for dtmf key presses during menu driven file playback. If a key is pressed playback stops and it executes the matched menu option. ASTERISK-24864 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4510/ ........ Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433446 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433447 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e953d15223	A couple minor cleanup tweaks. ... * In res/res_sorcery_realtime.c: Broke long line. * In main/bucket.c: Eliminated unnecessary NULL check as ast_sorcery_unref() is NULL tolerant and set the global object to NULL after unref in the system shutdown bucket_cleanup(). ........ Merged revisions 433420 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	abf3e40902	dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. ... This change adds an abstracted core DNS API which resembles the API described here[1]. The API provides a pluggable mechanism for resolvers and also a consistent view for records. Both synchronous and asynchronous queries are supported. This change also adds a res_resolver_unbound module which uses the libunbound library to provide resolution. Unit tests have also been written for all of the above to confirm the API and functionality. ASTERISK-24834 #close Reported by: Matt Jordan ASTERISK-24836 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4474/ Review: https://reviewboard.asterisk.org/r/4512/ [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	60f01520e7	Fix compilations errors on 64-bit OpenBSD systems ... In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to (long) when printing members of certain time structs. Review: https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close Reported by: snuffy Tested by: snuffy patches: openbsd-time64.diff uploaded by snuffy (License 5024) ........ Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433269 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	66670f02e6	Fix compilation issues for OpenBSD ... This patch addresses compilation issues for OpenBSD. Specifically, it addresses: * It allows including <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast in xmldoc.c In 13+, it also addresses a conditional inclusion in loader.c. Review: https://reviewboard.asterisk.org/r/4506 ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy patches: misc-openbsd.diff uploaded by snuffy (License 5024) ........ Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433247 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	148e8799fe	res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively. ... Valgrind found a memory leak and invalid access. * Fix invalid access by sscanf() being fed a non-nul terminated string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix memory leak in main/sorcery.c:sorcery_object_field_destructor(). * Fix potential NULL pointer dereference in main/xmldoc.c:xmldoc_get_syntax_config_option(). Review: https://reviewboard.asterisk.org/r/4513/ ........ Merged revisions 433199 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	3aa0a869c2	logger: Apply default console logging when configuration cannot be loaded. ... When logger.conf is missing or invalid enable console logging and display an error message. ASTERISK-24817 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4497/ ........ Merged revisions 433122 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433126 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	8c65c9167e	Various: bugfixes found via chaos ... Using DEBUG_CHAOS several instances of a null pointer crash, and one uninitialized variable were uncovered and fixed. Also added details on why Asterisk failed to initialize. Review: https://reviewboard.asterisk.org/r/4468/ ........ Merged revisions 433064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c41dd32b94	Audit ast_sockaddr_resolve() usage for memory leaks. ... Valgrind found some memory leaks associated with ast_sockaddr_resolve(). Most of the leaks had already been fixed by earlier memory leak hunt patches. This patch performs an audit of ast_sockaddr_resolve() and found one more. * Fix ast_sockaddr_resolve() memory leak in apps/app_externalivr.c:app_exec(). * Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs parameter for safety so the pointer will never be uninitialized on return. The same goes for res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs variable use ast_free instead of ast_free_ptr to provide better MALLOC_DEBUG information. Review: https://reviewboard.asterisk.org/r/4509/ ........ Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433057 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e89f83b3ad	main/frame: Don't report empty disallow values as an error ... In realtime, it is normal to have a database with both 'allow' and 'disallow' columns in the schema. It is perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files, you can't *not* provide the disallow value. Thus, the empty disallow value causes a spurious WARNING message, which is kind of annoying. This patch makes it so that a 'disallow' value with no ... value ... is ignored. Granted, you can still screw this up as well, as technically specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no 'allow' value in your database. But really, why would you do that? WHY? ASTERISK-16779 #close Reported by: Atis Lezdins ........ Merged revisions 432970 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432971 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b3fa35786f	core: Fix tab completion of "core set debug channel" CLI command. ... The "core set debug channel" CLI command mistakenly had source filenames added to its tab completion. This occurred because the CLI generator fell back to the "core set debug" command which permits setting debug at a source filename level. ASTERISK-21038 #close Reported by: Richard Kenner ........ Merged revisions 432944 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432945 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	f2c21ead1f	Logger: Fix MALLOC_DEBUG build error. ... Revision 432834 introduced a build error when MALLOC_DEBUG is used. Switch callid threadstorage to simple AST_THREADSTORAGE since we no longer need custom cleanup. Reported by: Corey Farrell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. ... Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	38ee441ea7	main/audiohook: Update internal sample rate on reads ... When an audiohook is created (which is used by the various Spy applications and Snoop channel in Asterisk 13+), it initially is given a sample rate of 8kHz. It is expected, however, that this rate may change based on the media that passes through the audiohook. However, the read/write operations on the audiohook behave very differently. When a frame is written to the audiohook, the format of the frame is checked against the internal sample rate. If the rate of the format does not match the internal sample rate, the internal sample rate is updated and a new SLIN format is chosen based on that sample rate. This works just fine. When a frame is read, however, we do something quite different. If the format rate matches the internal sample rate, all is fine. However, if the rates don't match, the audiohook attempts to "fix up" the number of samples that were requested. This can result in some seriously large number of samples being requested from the read/write factories. Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)). However, if the audiohook is still expecting an internal sample rate of 8000, we'll attempt to "fix up" the requested samples to: samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate); which is: 92160 = 3840 * (192000 / 8000) This results in us attempting to read 92160 samples from our factories, as opposed to the 3840 that we actually wanted. On a 64-bit machine, this miraculously survives - despite allocating up to two buffers of length 92160 on the stack. The 32-bit machines aren't quite so lucky. Even in the case where this works, we will either (a) get way more samples than we wanted; or (b) get about 3840 samples, assuming the timing is pretty good on the machine. Either way, the calculation being performed is wrong, based on the API users expectations. My first inclination was to allocate the buffers on the heap. As it is, however, there's at least two drawbacks with doing this: (1) It's a bit complicated, as the size of the buffers may change during the lifetime of the audiohook (ew). (2) The stack is faster (yay); the heap is slower (boo). Since our calculation is flat out wrong in the first place, this patch fixes this issue by instead updating the internal sample rate based on the format passed into the read operation. This causes us to read the correct number of samples, and has the added benefit of setting the audihook with the right SLIN format. Note that this issue was caught by the Asterisk Test Suite as a result of r432195 in the 13 branch. Because this issue is also theoretically possible in Asterisk 11, the change is being made here as well. Review: https://reviewboard.asterisk.org/r/4475/ ........ Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432811 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	29304d10a0	Add support for the clang compiler; update RAII_VAR to use BlocksRuntime ... RAII_VAR, which is used extensively in Asterisk to manage reference counted resources, uses a GCC extension to automatically invoke a cleanup function when a variable loses scope. While this functionality is incredibly useful and has prevented a large number of memory leaks, it also prevents Asterisk from being compiled with clang. This patch updates the RAII_VAR macro such that it can be compiled with clang. It makes use of the BlocksRuntime, which allows for a closure to be created that performs the actual cleanup. Note that this does not attempt to address the numerous warnings that the clang compiler catches in Asterisk. Much thanks for this patch goes to: * The folks on StackOverflow who asked this question and Leushenko for providing the answer that formed the basis of this code: http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang * Diederik de Groot, who has been extremely patient in working on getting this patch into Asterisk. Review: https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133 ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by: Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik de Groot (License 6600) ........ Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432808 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2889f074a0	core: Don't create snapshots with locks. ... Snapshots are immutable and are never changed. Allocating them with a lock is wasteful. Review: https://reviewboard.asterisk.org/r/4469/ ........ Merged revisions 432742 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	79e9b37ad0	localtime: Fix file descriptor leak on kqueue(2) systems ... The localtime management in the Asterisk core contains a thread that watches for changes in the local timezone. On systems where the directory containing /etc/localtime is modified frequently, the thread monitoring the changes will be woken up to determine if any changes in timezone have occurred. When using kqueue(2), this can cause a leak of file descriptors due to some improper management of resources. This patch updates the kqueue(2) handling in localtime, such that is no longer leaks resources. Review: https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License 6680) ........ Merged revisions 432691 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432693 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b85cb7ea1b	app: Add functions to swap voicemail function table for testing purposes ... ........ Merged revisions 432556 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	41ba8fd7c0	translate: Prevent invalid memory accesses on fast shutdown ... When a 'core restart now' or 'core stop now' is executed and a channel is currently in a media operation, the translator matrix can be destroyed while a channel is currently blocked on getting the best translation choice (see ast_translator_best_choice). When the channel gets the mutex, the translation matrix now has invalid memory, and Asterisk crashes. This patch does two things: (1) We now only clean up the translation matrix on a graceful shutdown. In that case, there are no channels, and so there is no risk of this occurring. (2) We also now set the __matrix and __indextable to NULL. In some initial backtraces when this occurred, it looked as if there was a memory corruption occurring, and it wasn't until we determined that something had restarted Asterisk that the issue became clear. By setting these to NULL on shutdown, it becomes a bit easier to determine why a crash is occurring. Note that we could litter the code with NULL checks on the __matrix, but the act of making the translation matrix cleaned up on shutdown should preclude this issue from occurring in the first place, and this part of the code needs to be as fast as possible. Review: https://reviewboard.asterisk.org/r/4457/ ........ Merged revisions 432453 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9e841e4fb6	ARI: Fix crash if integer values used in JSON payload 'variables' object. ... Sending the following ARI commands caused Asterisk to crash if the JSON body 'variables' object passes values of types other than strings. POST /ari/channels POST /ari/channels/{channelid} PUT /ari/endpoints/sendMessage PUT /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(), ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751 #close Reported by: jeffrey putnam Review: https://reviewboard.asterisk.org/r/4447/ ........ Merged revisions 432404 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432405 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	d79670b269	Dial API: add self destruct option when complete ... This patch adds a self-destruction option to the dial api. The usefulness of this is mostly when using async mode to spawn a separate thread used to handle the new call, while the calling thread is allowed to go on about other business. The only alternative to this option would be the calling thread spawning a new thread, or hanging around itself waiting to destroy the dial struct after completion. Example of use (minus error checking): struct ast_dial *dial = ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo"); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1); The dial_run call will return almost immediately after spawning the new thread to run and monitor the dial. If the call is answered, it is placed into the echo app. When completed, it will call ast_dial_destroy() on the dial structure. Note that any allocations made to pass values to ast_dial_set_user_data() or dial options must be free'd in a state callback function on any of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED, AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review: https://reviewboard.asterisk.org/r/4443/ ........ Merged revisions 432385 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d04fbb0f9d	app_chanspy, channel: fix frame leaks ... Fixed a couple of frame leaks that were found during testing. ASTERISK-24828 #close Reported by: John Hardin Review: https://reviewboard.asterisk.org/r/4445/ ........ Merged revisions 432362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432363 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d68012d1a3	channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI ... Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would be rejected if those crypto attributes contained either a key lifetime or a MKI parameter. While from a theoretical point of view this was defensible - Asterisk does not support key lifetimes or multiple crypto keys - from a practical point of view, this is quite a problem. A large number of endpoints offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually have to support anything more than a single key or refresh the key. In reality, this is (so far as we've seen) always the case. This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters in the following fashion: > The Lingon branch now handle lifetime and MKI parameters. > > We only accept lifetimes up to max for the crypto and higher than 10 hours > for packetization of 20 ms (50 pps). > > We only handle MKI with index 1. > > We do not really bother with counting packets and reinviting at end of > lifetime, so the min of 10 hours kind of takes care of most calls. If there > are longer ones, we rely on the other side for re-invites. > > It's still not perfect, but I personally think this is an improvement. A > configuration option for minimum lifetime accepted could be added. When the patch was ported forward, I decided against adding a configuration option as Olle's handling was more than sufficient for every case I've seen come through the issue tracker or through interoperability testing. We can revisit that decision if it proves to be false. A few small other tweaks were made to the surrounding code to reduce indentation and provide better type safety for the 'tag' parameter. Review: https://reviewboard.asterisk.org/r/4419/ Review: https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close Reported by: Terry Wilson ASTERISK-17899 #close Reported by: Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432258 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a528dfc9a7	ARI/PJSIP: Apply requesting channel's format cap to created channels ... This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan ........ Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2ea7ccbf70	http: Add missing html tag to 'httpstatus' functionality. ... ASTERISK-24724 #close Reported by: Ashley Sanders ........ Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e66b874f5d	Allow shutdown to unload modules that register bucket scheme's or codec's. ... * Change __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules that call ast_bucket_scheme_register or ast_codec_register to be unloaded during graceful shutdown only (13+ only). ASTERISK-24796 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4428/ ........ Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432059 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bb71672a47	main/asterisk.c: Reverse #if statement in listener() to fix code folding. ... listener() opens the same code block in two places (#if and #else). This confuses some folding editors causing it to think that an extra code block was opened. Folding in 'geany' causes all code after listener() to be folded as if it were part of that procedure. ASTERISK-24813 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4435/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	252aee4228	ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association. ... Processing an AOC-E event that does not or no longer has a channel association causes a crash. The problem with posting AOC events to the channel topic is that AOC-E events don't always have a channel association and posting the event to the all channels topic is just wrong. AOC-E events do however have their own charging association method to refer to the agreement with the charging entity. * Changed the AOC events to post to the AMI manager topic instead of the channel topics. If a channel is associated with the event then channel snapshot information is supplied with the AMI event. * Eliminated RAII_VAR() usage in aoc_to_ami() and ast_aoc_manager_event(). This patch supercedes the patch on Review: https://reviewboard.asterisk.org/r/4427/ ASTERISK-22670 #close Reported by: klaus3000 ASTERISK-24689 #close Reported by: Marcel Manz ASTERISK-24740 #close Reported by: Panos Gkikakis Review: https://reviewboard.asterisk.org/r/4430/ ........ Merged revisions 431974 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e3fd826cdb	tcptls: Handle new OpenSSL compile time option to disable SSLv3 ... Some distributions are going to disable SSLv3 at compile time. This option can be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling in Asterisk to look for that directive before attempting to use the SSLv3 specific methods. ASTERISK-24799 #close Reported by: Alexander Traud patches: no-ssl3-method.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 431936 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431937 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a4774ceaa5	Create work around for scheduler leaks during shutdown. ... * Added ast_sched_clean_by_callback for cleanup of scheduled events that have not yet fired. * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. Cleanup of replace_callno events is only run 11, since it no longer releases any references or allocations in 13+. ASTERISK-24451 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........ Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431917 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	455a98a2f8	sorcery: Output an error message if a wizard is specified for an object type and it isn't found. ... ASTERISK-24612 #close Reported by: Joshua Colp ........ Merged revisions 431771 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e2d3215b83	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8cc50b1ebc	Enable REF_DEBUG for ast_module_ref / ast_module_unref. ... Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431663 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	7ca1a0da04	various: cleanup issues found during leak hunt ... In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ ........ Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	03ce56d6c5	sorcery: Don't try to load object types which haven't been defined. ... The act of defining wizards for an object type in sorcery.conf will create a minimal object type. This can cause a problem when a module has multiple sorcery instances (which all get the wizards from sorcery.conf applied) but the sorcery instances do not all contain full information about the object types. Upon loading errors will occur stating that the objects can not be created. This is confusing and is actually perfectly fine. This change makes it so that only object types which have been fully defined will be loaded. ASTERISK-24748 #close Reported by: Joshua Colp ........ Merged revisions 431538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Ashley Sanders	6a76740b83	HTTP: For httpd server, need option to define server name for security purposes ... Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ ........ Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	bd0bdf1e41	Fix some memory leaks. ... These memory leaks were found and fixed by John Hardin. I'm just committing them for him. ASTERISK-24736 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4389 ........ Merged revisions 431468 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	388d691f34	stasis transfer: fix stasis bridge push race part two ... When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ ........ Merged revisions 431450 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	69e107b24e	res_pjsip_outbound_registration: Fix reload race condition. ... Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	c7591ef6bc	tcptls: Bad file descriptor error when reloading chan_sip ... While running through some scenarios using chan_sip and tcp a problem would occur that resulted in a flood of bad file descriptor messages on the cli: tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor The message is received because the underlying socket has been closed, so is valid. This is probably happening because unloading of chan_sip is not atomic. That however is outside the scope of this patch. This patch simply stops the logging of multiple occurrences of that message. ASTERISK-24728 #close Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/ ........ Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431219 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	e826cb8a26	Manager: Fix Manager Action ModuleLoad to give correct response when reloading ... Prior to this patch, ModuleLoad would respond with an error indicating that the requested module wasn't found in spite of finding and reloading the module. Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 #close ........ Merged revisions 431153 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	e62bd46511	res_pjsip: make it unloadable (take 2) ... Due to the original patch causing memory corruptions it was removed until the problem could be resolved. This patch is the original patch plus some added locking around stasis router subcription that was needed to avoid the memory corruption. Description of the original problem and patch (still applicable): The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4363/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 431179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a43d24a9d3	bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media. ... This change fixes two issues: 1. During a swap operation bridging added the new channel before having the swap channel leave. This was not handled in bridge_native_rtp and could result in a channel not getting reinvited back to Asterisk. After this change the swap channel will leave first and the new channel will then join. 2. If a re-invite was received after a session had been established any upstream elements (such as bridge_native_rtp) were not notified that they may want to re-evaluate things. After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs and upstream can react. AST-1524 #close Review: https://reviewboard.asterisk.org/r/4378/ ........ Merged revisions 431157 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	965777ccfc	Various fixes for OS X ... This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a8ae5a7bcb	dynamic realtime: Updates fail to work due to update fields being passed over ... When a crash was fixed due to usage of the REALTIME function in r423003, a regression was introduced into ast_update2_realtime where the update fields passed to the function would be skipped and the lookup field processed twice. The use of this function is a bit interesting: A variable argument list is used with two sentinel values - the first marks the end of the lookup fields/values; the second marks the end of the update fields/values. Unfortunately, ast_update2_realtime parses over the lookup fields twice, as opposed to parsing over the update fields. This causes the lookups to succeed, but the updates itself to have no effect. Note that the most common instance of this problem occurred in app_voicemail during the updating of a mailbox password. Thanks to the issue reporter, Paddy Grice, for pointing out the problem. Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231 ASTERISK-24626 #close Reported by: Paddy Grice ........ Merged revisions 431072 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago