4606 Commits (27dc2094e98b59b8a50b059ddd6048285a42e6b9)

Author	SHA1	Message	Date
Matt Jordan	0a1a550593	apps/app_dictate: Fix typo in attribution ... Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago
Matt Jordan	49f81ddb85	Makefile: Remove coverage files on 'make clean' ... This patch updates a variety of Makefiles in Asterisk's build system to remove .gcda and .gcno files when 'make clean' is executed. These files are generated when '--enable-coverage' is passed to the Asterisk configure script. Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602	10 years ago
Joshua Colp	e99e654d75	app_dial: Hold reference to calling channel formats when dialing outbound. ... Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db	10 years ago
Richard Mudgett	a2f4d03c87	app_directory: Fix crash when using the alias option 'a'. ... The voicemail.conf mailbox key/value pair is defined as: <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just <mailbox>=<password>,<name> when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374	10 years ago
Corey Farrell	55c8daf88b	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	10 years ago
George Joseph	7fcf0a97b8	app_playback: Suppress warnings on playback if channel hung up ... If a channel hangs up while an audio file is playing, there's no need to clutter up the logs with a warning so suppress it if ast_check_hangup returns true. Also, change warning to debug/2 in file.c if writing a frame fails. Same reasoning. Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Jonathan Rose	d49d64b79c	app_voicemail: fix moving when old messages full ... When completing voicemail playback of a message in the 'INBOX', the message gets moved to the 'Old' messages folder. Without this patch, if the 'Old' folder is already at its set limit, then the 'INBOX' message will simply be deleted. With this patch, the flag to delete the message will be removed if the save_to_folder function indicates that the message could not be moved due to a full folder. ASTERISK-25082 #close Reported by: Jonathan Rose Review: https://gerrit.asterisk.org/#/c/448/ Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f	10 years ago
Ivan Poddubny	87263b47b5	app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter ... This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3 parameters: position, original position and waiting time. ASTERISK-25038 #close Reported by: Etienne Lessard Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618	10 years ago
Kevin Harwell	7e5056b393	app_confbridge: Default the template option to a compatible default profile. ... Confbridge dynamic profiles did not have a default profile unless you explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a template was not set prior to the bridge being created then some options were left with no default values set. This patch makes it so the default templates are set to the default bridge and user profiles. ASTERISK-24749 #close Reported by: philippebolduc Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a	10 years ago
Mark Michelson	4f1a8dbe92	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
Matthew Jordan	e05c8ae68e	apps/app_queue: Prevent possible crash when evaluating queue penalty rules ... Although it only occurred once, a crash occurred when a queue attempted to evaluate a queue penalty rule that appeared to have already been destroyed. In many locations in app_queue, a test is done to see if qe->pr is NULL; however, when we dispose of a queue's penalty rules, we don't set the pointer to NULL after free'ing it. This patch does that to prevent any dangling pointers from lingering on the queue object. Review: https://reviewboard.asterisk.org/r/4522 ASTERISK-23319 #close Reported by: Vadim patches: rb4552.patch submitted by Stefan Engström (License 6691) ........ Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	f324870dab	clang compiler warnings: Fix pointer-bool-converesion warnings ... This patch fixes several warnings pointed out by the clang compiler. * chan_pjsip: Removed check for data->text, as it will always be non-NULL. * app_minivm: Fixed evaluation of etemplate->locale, which will always evaluate to 'true'. This patch changes the evaluation to use ast_strlen_zero. * app_queue: - Fixed evaluation of qe->parent->monfmt, which always evaluates to true. Instead, we just check to see if the dereferenced pointer evaluates to true. - Fixed evaluation of mem->state_interface, wrapping it with a call to ast_strlen_zero. * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero. Review: https://reviewboard.asterisk.org/r/4541 ASTERISK-24917 Reported by: dkdegroot patches: rb4541.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Ashley Sanders	e301185983	stasis: set a channel variable on websocket disconnect error ... Resolve compile errors caused by r433863 by fixing the documentation xml to comply with the schema. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Ashley Sanders	a1f12d9231	stasis: set a channel variable on websocket disconnect error ... Resolve compile errors caused by r433839 by included the missing header file, pbx.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Ashley Sanders	7293ecd90b	stasis: set a channel variable on websocket disconnect error ... When an error occurs while writing to a web socket, the web socket is disconnected and the event is logged. A side-effect of this, however, is that any application on the other side waiting for a response from Stasis is left hanging indefinitely (as there is no mechanism presently available for notifying interested parties about web socket error states in Stasis). To remedy this scenario, this patch introduces a new channel variable: STASISSTATUS. The possible values for STASISSTATUS are: SUCCESS - The channel has exited Stasis without any failures FAILED - Something caused Stasis to croak. Some (not all) possible reasons for this: - The app registry is not instantiated; - The app requested is not registered; - The app requested is not active; - Stasis couldn't send a start message ASTERISK-24802 Reported By: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4519/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	5f8faf16af	clang compiler warnings: Fix -Wabsolute-value warnings ... This patch fixes several warnings caught by clang - in this case, usage of the abs function on non-integer values. This patch uses labs and fabs, as appropriate, in the various affected files. Review: https://reviewboard.asterisk.org/r/4525 ASTERISK-24917 Reported by: dkdegroot patches: rb4525.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e126ab9eeb	clang compiler warnings: Fix a variety of "unused" warnings ... This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable errors caught by clang. Specifically: * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], qsmp_cmd_usage[] * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" * funcs/func_env.c:729: Fixed ast_str_append_substr. * main/editline/np/strlcat.c: removed unused rcsid variable * main/editline/np/strlcpy.c: removed unused rcsid variable * main/security_events.c: removed unused TIMESTAMP_STR_LEN * utils/conf2ael.c: removed unused cfextension_states * utils/extconf.c: removed unused cfextension_states Review: https://reviewboard.asterisk.org/r/4526 ASTERISK-24917 Reported by: dkdegroot patches: rb4526.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	eb70993a50	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	fd434a210f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	a3fe43fbdc	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	2122c205e6	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	a4c27baf47	apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation ... This patch corrects the documentation for the AMD application. Specifically: * It documents the maximum_word_length option, which limits the maximum allowed length of a single utterance. * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was undocumented. Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. ASTERISK-19470 #close Reported by: Frank DiGennaro ........ Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	055001716c	app_voicemail: Fix crash with IMAP backends when greetings aren't present ... When an IMAP backend is in use and greetings are set to be used, but aren't present for a user in their IMAP folder, Asterisk will crash. This occurs due to the mailstream being set to the 'greetings' folder and being left in that particular state, regardless of the success/failure of the attempt to access the folder the mailstream points to. Later access of the mailstream assumes that it points to the 'INBOX' (or some other folder), resulting in either a crash (if the greetings folder didn't exist and the mailstream is invalid) or an inability to read messages from the 'INBOX' folder. This patch restores the mailstream to its correct state after accessing the greetings. This fixes the crash, and sets the mailstream to the state that VoiceMailMain expects. Note that while ASTERISK-23390 also contained a patch for this issue, the patch on ASTERISK-24786 is the one being merged here. Review: https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close Reported by: Ben Smithurst ASTERISK-24786 #close Reported by: Graham Barnett Tested by: Graham Barnett patches: app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685) ........ Merged revisions 432695 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	06fa8db864	app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE. ... There is a leftover "assert" in app_voicemail/__messagecount that references variables that don't exist. This causes the compile to fail when --enable-dev-mode and IMAP_STORAGE are selected. This patch removes the assert. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4461/ ........ Merged revisions 432484 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	169058e73f	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	de86b30dba	make: Remove 'res_features' from libraries to link against with cygwin/mingw32 ... Both the apps and channels Makefiles still listed 'res_features' as modules to link against when compiling for cygwin or mingw32. This module hasn't existed for quite some time. ASTERISK-18105 #close Reported by: feyfre ........ Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432342 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	c40d78c31e	apps/app_voicemail: Demote an ERROR message to a WARNING message ... When using IMAP voicemail with FreePBX, you will often get ERROR messages complaining about not being able to find a mailbox. This is due to how FreePBX handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this a configuration error, as in any other system it would be indicative of someone misconfiguring their system. Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch demotes the message so that system administrators can hopefully reduce some of the noise in their log files. Note that in the original patch this was made into a NOTICE, but that's a too forgiving. ASTERISK-24790 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) ........ Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	c7bdf62a95	apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange ... When interfacing with Microsoft Exchange, custom headers will be returned as all lower case. Currently, the IMAP header code will fail to parse the returned custom headers, as it will be performing a case sensitive comparison. This can cause playback of messages to fail, as needed information - such as origtime - will not be present. This patch updates app_voicemail's header parsing code to perform a case insensitive lookup for the requested custom headers. Since the headers are specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be unique in an IMAP message, this should cause no issues with other systems. ASTERISK-24787 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685) ........ Merged revisions 432012 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	4f4d03fdd1	apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes ... The Test Event for MIXMONITOR_END - which signals that a MixMonitor has completed - technically fired before the filestream was closed. If a test used this to trigger a condition to verify that the file was written, it could result in a race condition where the file size would not be what the test expected. Luckily, no tests were using this (although they should have been). Since the test event needed to be moved after the point where the MixMonitor autochan has been destroyed, the test event no longer emits the channel name. Luckily, nothing needs it. ........ Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	feddab7944	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	05e2832b35	app_agent_pool: Fix initial module load agent device state reporting. ... When the app_agent_pool module initially loads there is a race condition between the thread loading agents.conf and the device state internal processing thread. If the device state internal processing thread handles the agent creation state updates before the thread that loaded agents.conf registers the device state provider callback then the cached agent state is "Invalid". When a consumer module like app_queue asks for the agent state it gets the cached "Invalid" state instead of the real state from the provider. * Moved loading the agents.conf configuration to the last thing setup by app_agent_pool in load_module(). Now the device state provider callback is registered before the config is loaded so the agent creation state updates are guaranteed to get the initial device state. * Removed some now redundant config cleanup on error in load_config(). * Added lock protection when accessing the device state in agent_pvt_devstate_get() and eliminated the RAII_VAR() usage. ASTERISK-24737 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4390/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	eda125f98d	app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel. ... Starting and stopping conference recording more than once causes the recording channels to be leaked. For v13 the channels also show up in the CLI "core show channels" output. * Reworked and simplified the recording channel code to use ast_bridge_impart() instead of managing the recording thread in the ConfBridge code. The recording channel's ref handling easily falls into place and other off nominal code paths get handled better as a result. ASTERISK-24719 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/ Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	1a17693789	app_confbridge: Restore user's menu name to CLI output of 'confbridge list' ... When issuing a 'confbridge list XXXX' CLI command, the resulting output no longer displays the menu associated with a ConfBridge participant. The issue was caused by ASTERISK-22760. When that patch was done, it removed the copying of the menu name associated with the user from the actual user profile. This patch fixes the issue by copying the menu name over to the user profile when the menu hooks are applied to the user. Since that function now does a little bit more than just apply the hooks, the name of the function has been changed to cover the copying of the menu name over as well. In addition, there is a disparity between the menu name length as it is stored on the conf_menu structure and the confbridge_user structure; this patch makes the lengths match so that a strcpy can be used. Review: https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close Reported by: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e302116e40	app_confbridge: Make CBRec channel names more unique. ... Channel names should be different from other channels in the system while the channel exists. * Use a sequence number for CBRec channels instead of a random number because the same random number could be picked again for the next CBRec channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	f8b3fb6e2f	app_confbridge: Whitespace ... Because there is sometimes no sence to any whitespace. ........ Merged revisions 431049 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	e23f07beb8	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	beb20440e0	apps/app_voicemail: Trigger MWI notification with MixMonitor m() option ... The MixMonitor m() option allows a recording to be pushed to a specific voicemail mailbox. If the message is delivered to the mailbox's INBOX, however, no MWI notification is currently raised. This patch corrects the issue by properly calling notify_new_state from the msg_create_from_file function. This will cause MWI to be triggered if the message was placed in the mailbox's INBOX. ASTERISK-24709 #close Reported by: Gareth Palmer patches: app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169) ........ Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	894d4d781c	apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values ... The Dial application has some interesting options with the mid-call Macro (M) and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific values, the Dial application will take some action upon the channels involved in the dial operation (such as hanging up a particular party, etc.) The Dial application ensures that a Stasis message is published in the event that MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so that there is a corresponding DialEnd event published in AMI/ARI for the DialBegin event that preceeded it. A bug exists where that same DialEnd event will be published on Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial application cares about. This causes two DialEnd events to be published - one with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all sorts of wrong. This patch fixes the bug by ensuring that we only publish a DialEnd message to Stasis if the Dial application's mid-call Macro/GoSub returns something that Dial cares about. Review: https://reviewboard.asterisk.org/r/4336 ASTERISK-24682 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e659b3e53d	app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend ... When using ODBC or IMAP storage, temporary files created on the file system must be disposed of using the DISPOSE macro. The DELETE macro will map to a deletion function for the backend storage, but does not clean up any local files created as a result of the operation. When using voicemail with the operator and review options enabled, pressing 0 to enter the menu, followed by 1 to save the message, followed by any other DTMF press to delete the message, will result in the temporary file lingering on the file system. This patch properly calls DISPOSE after the DELETE. This causes the local file to be disposed of. ASTERISK-24288 #close Reported by: LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640) ........ Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	1f94b96749	app_macro: Don't restore the calling location on a channel redirect. ... v11: If a channel redirect to a macro exten of a macro that is active happens, the redirect location doesn't get executed. Instead the original macro location is restored and gets reexecuted. v13: An additional effect happens if a parked call times out to an extension in the macro that parked the call then the macro is reexecuted instead of the expected park return location. * Made not restore the macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. * Increased the locked channel range when setting up the macro execution environment to cover things that should be done while the channel is locked. * Removed unnecessary NULL tests before calling ast_free() in _macro_exec(). ASTERISK-23850 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4292/ ........ Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9ae57e0dd6	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0321d00413	apps/app_meetme: Apply default values on initial load with no config file ... When the app_meetme module is loaded without its configuration file, the module settings aren't initialized. In particular, this impacts the use of logging realtime members. This patch guarantees that we always set the default module settings on initial load. Review: https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno Borges (License 6116) ........ Merged revisions 429027 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429028 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bda506c346	apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously ... The UW IMAP library is instrinsically not thread-safe, and relies upon higher level applications to guarantee thread safety. For the most part, this is provided by the vms object, which provides locking for individual streams. Unfortunately, this is not sufficient for calls to mail_open which create the IMAP stream. mail_open can, on some systems, call into a UW IMAP specific function for determining the address of a system based on a hostname, ip_nametoaddr. In the ip6_unix implementation of this function, static variables are used to hold parsing buffers. This can cause a crash if multiple threads attempt to convert a hostname to an address at the same time. Locking on a single mail stream is not sufficient to prevent simultaneous access to these static variables. In the IMAP library, this function can be called from the mail_open and imap_status functions. As the imap_status function is not used by app_voicemail, locking on access to mail_open is sufficient to prevent any mangling of the buffers. Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516 #close Reported by: David Duncan Ross Palmer Tested by: David Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660) ........ Merged revisions 428863 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6e30ccd242	app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. ... The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428654 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	19ba0d1363	AST-2014-017 - app_confbridge: permission escalation/ class authorization. ... Confbridge dialplan function permission escalation via AMI and inappropriate class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” could also be used to execute arbitrary system commands without first checking for system access. Asterisk now inhibits the CONFBRIDGE function from being executed from an external interface if the live_dangerously option is set to no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed to execute under a user with system level access. ASTERISK-24490 Reported by: Gareth Palmer ........ Merged revisions 428332 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428334 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	45f4fe6f1c	apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves ... When r428077 was made for ASTERISK-24522, it failed to take into account users who are neither wait_marked nor end_marked. These users are *also* supposed to hear the 'leader has left the conference' message. Granted, this behaviour is a bit odd; however, that is how it used to work... and behaviour changes are not good. This patch ensures that if there are any 'normal' users present when the last marked user leaves the conference, the message will still be played to them. Note that this regression was caught by the Asterisk Test Suite's confbridge_nominal test, which has a quirky combination of users. ........ Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6d59f1546e	app_confbridge: Don't play leader leaving prompt if no one will hear it ... Consider the following: - A marked user in a conference - One or more end_marked only users in the conference When the marked users leaves, we will be in the conf_state_multi_marked state. This currently will traverse the users, kicking out any who have the end_marked flags. When they are kicked, a full ast_bridge_remove is immediately called on the channels. At this time, we also unilaterally set the need_prompt flag. When the need_prompt flag is set, we then playback a sound to the bridge informing everyone that the leader has left; however, no one is left in the bridge. This causes some odd behaviour for the end_marked users - they are stuck waiting for the bridge to be unlocked. This results in them waiting for 5 or 6 seconds of dead air before hearing that they've been kicked. Unfortunately, we do have to keep the bridge locked while we're playing back the 'leader-has-left' prompt. If there are any wait_marked users in the conference, this behaviour can't be easily changed - but we do make the case of the end_marked users better with this patch. Review: https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close Reported by: Matt Jordan ........ Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1536b0ecb6	Fix race condition that could result in ARI transfer messages not being sent. ... From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago