asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	8b295a2792	security_events: Fix assertion failure in dev-mode on optional IE parsing When formatting an optional IE, the value is, of course, optional. As such, it is entirely appropriate for ast_json_object_get to return NULL. If that occurs, we now simply skip the IE that was requested, as it was not provided by the entity that raised the event. Thanks to George Joseph (gtjoseph) for catching this and reporting it in #asterisk-dev ........ Merged revisions 407750 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e8e2f91bba	timing: Improve performance for most timing implementations. This change allows timing implementation data to be stored directly on the timer itself thus removing the requirement for many implementations to do a container lookup for the same information. This means that API calls into timing implementations can directly access the information they need instead of having to find it. Review: https://reviewboard.asterisk.org/r/3175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	42d3fe8772	security_events: Fix error caused by DTD validation error The appdocsxml.dtd specifies that a "required" attribute in a parameter may have a value of yes, no, true, or false. On some systems, specifying "False" instead of "false" would cause a validation error. This patch fixes the casing to explicitly match the DTD. ........ Merged revisions 407676 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	cbaa27142c	security_events: Add AMI documentation; output optional fields This patch adds documentation for the Security Events that are emited over AMI. It also notes these events in the UPGRADE/CHANGES file. ........ Merged revisions 407589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	6f4a834870	Logger: Fix handling of absolute paths This fixes path handling for log files so that an extra / is not appended to the file path when the path is absolute (begins with /). This would previously result in different but functionally equivalent paths in the output of 'logger show channels'. ........ Merged revisions 407455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407456 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407459 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	dd0c6e9cc1	devicestate: Make ast_devstate_changed_literal() return value and doxygen consistent. Nothing actually cares about the value anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose ........ Merged revisions 407337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407338 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407339 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	12668b6659	tcptls.c: Made TLS handle a certificate chain file. Thanks to Guillaume Martres for doing the necessary research to validate the change. (closes issue ASTERISK-17727) Reported by: LN Patches: use_certificate_chain.patch (license #5864) patch uploaded by st documente_certificate_chain.patch (license #6576) patch uploaded by Guillaume Martres ........ Merged revisions 407272 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407273 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	01af8d6e12	cdrs: Check for applications to lock onto during dial begin handling This patch brings CDR processing further in line with r407085. During some dial operations, the application would not be locked to the Dial application and would instead continue to show the previously known application. In particular, this would occur when a Parked call would time out. This was due to a previous snapshot already locking the application to Park - processing this in a Dial Begin allows the Dial application to reassert its rightful place. (CDRs. Ugh.) But hooray for the Parked Call tests for catching this in the Asterisk Test Suite. ........ Merged revisions 407166 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e5899852cc	res_stasis: Enable transfers and provide events when they occur. This change enables transfers within ARI created bridges and adds events for when they occur. Unlike other events these will be received if any subscribed object is involved in the transfer. (closes issue ASTERISK-22984) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/3120/ ........ Merged revisions 407153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	66c46fba24	CDRs: fix a variety of dial status problems, h/hangup handler creating CDRs This patch fixes a number of small-ish problems that were noticed when witnessing the records that the FreePBX dialplan produces: (1) Mid-call events (as well as privacy options) have the ability to change the overall state of the Dial operation after the called party answers. This means that publishing the DialEnd event when the called party is premature; we have to wait for the execution of these subroutines to complete before we can signal the overall status of the DialEnd. This patch moves that publication and adds handlers for the mid-call events. (2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto datastore is detected. This flag was preventing CDRs from being recorded for all outbound channels that had a 'continue' option enabled on them by the Dial application. (3) The CDR engine now locks the 'Dial' application as being the CDR application if it detects that the current CDR has entered that app. This is similar to the logic that is done for Parking. In general, if we entered into Dial, then we want that CDR to record the application as such - this prevents pre-dial handlers, mid-call handlers, and other shenaniganry from changing the application value. (4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places to determine if the channel is in hangup logic or dead. In either case, we don't want to record changes in the channel. (5) The default option for "endbeforehexten" has been changed to "yes". In general, you don't want to see CDRs in the 'h' exten or in hangup logic. Since the semantics of that option changed in 12, it made sense to update the default value as well. (6) Finally, because we now have the ability to synchronize on the messages published to the CDR topic, on shutdown the CDR engine will now synchronize to the messages currently in flight. This helps to ensure that all in-flight CDRs are written before shutting down. (closes issue ASTERISK-23164) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3154 ........ Merged revisions 407084 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407085 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Corey Farrell	c35d07950f	res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions ast_bind to a port reserved for another program by SELinux causes errno == EACCES. This caused random failures when binding rtp or udptl sockets. Treat EACCES as a non-fatal error, try next port. (closes issue ASTERISK-23134) Reported by: Corey Farrell ........ Merged revisions 406933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406934 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406935 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	98de7719dd	Make a NOTICE about an invalid channel name more useful. ........ Merged revisions 406918 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406919 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	601692a7e4	rtp_engine: improved handling of get_rtp_info failure In ast_rtp_instance_make_compatible(), after a failure of channel tech call get_rtp_info() to return peer_instance, the null pointer would be passed to ao2_ref, producing an error that looked like a refernce counting problem but is not. This patch corrects that and adds helpful LOG_ERROR messages to indicate which failure path occurred. (issue AST-1276) Review: https://reviewboard.asterisk.org/r/3156/ ........ Merged revisions 406721 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406722 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406723 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	f9479fbcbd	manager: ExtensionStatus event status human readable When an 'ExtensionStatus' event was raised it included the status as a numerical value, but did not include a text description of the status. Added a 'StatusText' field to the event which is a string representation of the extension status. Also added this to the 'Extension State' command response. (closes issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	8a762efb35	Allow nested #includes in extconfig.conf extconfig.conf was hard-coded to not allow nested includes for some reason. The code has been this way since a patch was merged for ASTERISK-3333 (revision 4889), which was a significant update to this code ("Merge config updates"). I can't figure out any good reason why this should be limited. This patch just removes the limit and uses the default nesting depth limit. Closes issue ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/ ........ Merged revisions 406643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406644 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406645 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	cc42229f26	manager: The eventfilter= option now takes an extended regex. In pre-trunk versions (...12) it accepts a basic regex, which is confusing because all other regexes in asterisk are of the extended kind. Review: https://reviewboard.asterisk.org/r/3147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	33071d636c	Protect ast_filestream object when on a channel The ast_filestream object gets tacked on to a channel via chan->timingdata. It's a reference counted object, but the reference count isn't used when putting it on a channel. It's theoretically possible for another thread to interfere with the channel while it's unlocked and cause the filestream to get destroyed. Use the astobj2 reference count to make sure that as long as this code path is holding on the ast_filestream and passing it into the file.c playback code, that it knows it's valid. Bug reported by Leif Madsen. Review: https://reviewboard.asterisk.org/r/3135/ ........ Merged revisions 406566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	45261449ec	tcptls.c: Add missing cleanup on off nominal path. ........ Merged revisions 406514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406515 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406516 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	42c15dfa6e	CEL: Protect data structures during reload and shutdown. The CEL data structures need to be protected during a configuration reload and shutdown. Asterisk crashed during a shutdown because CEL events were still in flight and the CEL data structures were already destroyed. * Protected the cel_backends, cel_dialstatus_store, and cel_linkedids ao2 containers with a global ao2 object wrapper. * Added NULL checks before use of the cel_backends, cel_dialstatus_store, and cel_linkedids ao2 containers in case the CEL module is already shutdown. * Fixed overloading of the cel_linkedids held objects reference count. During shutdown any held objects would be leaked. * Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is not being tracked. The objects in the cel_linkedids container were not removed if the LINKEDID_END event is not used. * Added access protection to the cel_backends container during the CLI "cel show status" command. * Made cel_backends, cel_dialstatus_store, and cel_linkedids use the standard ao2 callback templates for the hash and cmp functions. * Eliminated unnecessary uses of RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated resources on failure. (closes issue AST-1253) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3128/ ........ Merged revisions 406417 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406418 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	2a9d15b400	Thread Debugging: Add LWP to core show locks output This patch adds the LWP to core show locks output if it is available. Review: https://reviewboard.asterisk.org/r/3142/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a9911f027e	manager: Register atexit shutdown routine only once. * Made register atexit shutdown routine only once in __init_manager(). * Fixed some initial load failure conditions in __init_manager(). * Made reset options to defaults on reload when the reload will actually happen. * Removed unnecessary container traversals of the white/black filters during manager_free_user(). * ast_free() does not need a NULL check before calling. ........ Merged revisions 406359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406401 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	82cce81737	manager: Protect data structures during shutdown. Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic number" error on a "core restart gracefully" command if an AMI connection is established. * Added ao2_global_obj protection to the sessions global container. * Fixed the order of unreferencing a session object in session_destroy(). * Removed unnecessary container traversals of the white/black filters during session_destructor(). (closes issue AST-1242) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3144/ ........ Merged revisions 406341 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	64e2e1d5d8	pbx.c: Pre-initialize timezone to avoid crash on destroy In ast_build_timing, initialize the timezone value to NULL in order to avoid deferencing an uninitialized value later when calling ast_destroy_timing. The timezone value could be uninitialized if ast_build_timing were to fail due to a zero length time string. (closes issue ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review: https://reviewboard.asterisk.org/r/3134/ Patches: ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406245 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406264 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	9a88cc33f8	manager: Clarify eventfilter documentation. Textual changes only. Review: https://reviewboard.asterisk.org/r/3133/ ........ Merged revisions 406079 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406080 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406081 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	2b14601bdc	pjsip: fix support for allow=all This change adds improvements to support for allow=all in pjsip.conf so that it functions as intended. Previously, the allow/disallow socery configuration would set & clear codecs from the media.codecs and media.prefs list, but if all was specified the prefs list was not updated. Then a call would fail when create_outgoing_sdp_stream() created an SDP with no audio codecs. A new function ast_codec_pref_append_all() is provided to add all codecs to the prefs list - only those not already on the list. This enables the configuration to specify a codec preference, but still add all codecs, and even then remove some codecs, as shown in this example: allow = ulaw, alaw, all, !g729, !g723 Also, the display order of allow in cli output is updated to match the configuration by using prefs instead of caps when generating a human readable string. Finally, a change to create_outgoing_sdp_stream() skips a codec when it does not have a payload code instead of the call failing. (closes issue ASTERISK-23018) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3131/ ........ Merged revisions 405875 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	2704b49c1b	http: supported chunked Transfer-Encoding This change implements support for HTTP Transfer-Encoding chunked in both JSON and Form (post vars) body content. A new function ast_http_get_contents() handles both regular and chunked mode body, returning after the entire body is received. (closes issue ASTERISK-23068) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3125/ ........ Merged revisions 405861 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	f6647d2362	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	7054e12ef2	manager: Originate doesn't abort on failed format_cap allocation action_originate responds to the remote system with an error when cap==NULL, but doesn't return (abort the originate). Patched to return. (closes issue ASTERISK-23034) Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 405745 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405746 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ed0d083596	string container: Remove unnecessary RAII_VAR usage and string object lock. ........ Merged revisions 405541 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	828f339a9c	verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	373965dbff	CDRs: Synchronize dialplan applications that manipulate CDRs with the engine In https://reviewboard.asterisk.org/r/3057/, applications and functions that manipulate CDRs were made to interact over Stasis. This was done to synchronize manipulations of CDRs from the dialplan with the updates the engine itself receives over the message bus. This change rested on a faulty premise: that messages published to the CDR topic or to a topic that forwards to the CDR topic are synchronized with the messages handled by the CDR topic subscription in the CDR engine. This is not the case. There is no ordering guaranteed for two messages published to the same topic; ordering is only guaranteed if a message is published to the same subscriber. Stasis was modified in r405311 to allow a publisher to synchronize on the subscriber. This patch uses that API to synchronize the CDR publishers with the CDR engine message router, which maintains the overall topic subscription. (closes issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f8aaf585a3	stasis: Add methods to allow for synchronous publishing to subscriber This patch adds an API call to Stasis that allows a publisher to publish a stasis message that will not return until a specific subscriber handles the message. Since a subscriber can have their own forwarding topic which orders messages from many topics, this allows a publisher who knows of that subscriber to synchronize to that subscriber regardless of the forwarding relationships between topics. This is of particular use for dialplan applications that need to synchronize on a particular subscriber's handling of a message. (issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4bde62eb40	Logging callid: Fix some sizeof() references per coding guidelines. ........ Merged revisions 405281 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405282 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	51901aa2ed	astobj2: Correct ao2_iterator opacity violations This corrects the ao2_iterator opacity violations in res_pjsip_session.c by adding a global function to get the number of elements inside the container hidden behind the iterator. (closes issue ASTERISK-23053) Review: https://reviewboard.asterisk.org/r/3111/ Reported by: Richard Mudgett ........ Merged revisions 405253 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	3eee3f21ad	asterisk.c: suppress live_dangerously warning on rasterisk Even since the fixes of AST-2013-007, Asterisk prints the following warning on startup if the user decided to live dangerously: Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This message is intended for the logs and interactive startup. No need for it to appear on a remote console. This commit removes it from there. (closes issue ASTERISK-23084) Review: https://reviewboard.asterisk.org/r/3101/ ........ Merged revisions 404861 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404888 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404911 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	bd4be333fb	manager: UserEvent including action on output AMI action UserEvent event response would include the action header in its keyvalue pairs list. Adjusted the start of the header loop to skip over the action part. (closes issue ASTERISK-22899) Reported by: outtolunc Patches: svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license 5198) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	20d02c1288	channels.c: core show channeltypes slicing 'core show channeltypes' type column is being sliced, resulting in incomplete type names. (closes issue ASTERISK-22919) Reported by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded by outtolunc (license 5198) ........ Merged revisions 404579 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404581 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	a952abb9da	http: Properly reject requests with Transfer-Encoding set Asterisk does not support any of the transfer encodings specified in HTTP/1.1, other than the default "identity" encoding. According to RFC 2616: A server which receives an entity-body with a transfer-coding it does not understand SHOULD return 501 (Unimplemented), and close the connection. A server MUST NOT send transfer-codings to an HTTP/1.0 client. This patch adds the 501 Unimplemented response, instead of the hard work of actually implementing other recordings. This behavior is especially problematic for Node.js clients, which use chunked encoding by default. (closes issue ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/ ........ Merged revisions 404565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b172d369c4	res_pjsip: Add PJSIP CLI commands Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	a0c288bb23	say.c: correct time for polish In ast_say_date_with_format_pl(), change ast_say_number() to use tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported by: Robert Mordec Review: https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch uploaded by veilen (license 6555) ........ Merged revisions 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9e4f80a4f6	Whitespace fixes. ........ Merged revisions 404419 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	efd7c40712	security_events: log events with descriptive names This patch updates the log messages to include descriptive names for event types. This is an improvement over having only cryptic type numbers. (closes issue ASTERISK-22909) Reported by: outtolunc Review: https://reviewboard.asterisk.org/r/3081/ Patches: svn_security_events.c.names.diff.txt uploaded by outtolunc (license 5198) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	1b91ee6c4b	Fix a deadlock that occurred due to a conflict of masquerades. For the explanation, here is a copy-paste of the review board explanation: Initially, it was discovered that performing an attended transfer of a multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread started a masquerade and reached the point where it was calling the fixup() callback on the "original" channel. For chan_pjsip, this involves pushing a synchronous task to the session's serializer. The problem was that a task ahead of the fixup task was also attempting to perform a channel masquerade. However, since masquerades are designed in a way to only allow for one to occur at a time, the task ahead of the fixup could not continue until the masquerade already in progress had completed. And of course, the masquerade in progress could not complete until the task ahead of the fixup task had completed. Deadlock. The initial fix was to change the fixup task to be asynchronous. While this prevented the deadlock from occurring, it had the frightful side effect of potentially allowing for tasks in the session's serializer to operate on a zombie channel. Taking a step back from this particular deadlock, it became clear that the problem was not really this one particular issue but that masquerades themselves needed to be addressed. A PJSIP attended transfer operation calls ast_channel_move(), which attempts to both set up and execute a masquerade. The problem was that after it had set up the masquerade, the PBX thread had swooped in and tried to actually perform the masquerade. Looking at changes that had been made to Asterisk 12, it became clear that there never is any time now that anyone ever wants to set up a masquerade and allow for the channel thread to actually perform the masquerade. Everyone always is calling ast_channel_move(), performs the masquerade itself before returning. In this patch, I have removed all blocks of code from channel.c that will attempt to perform a masquerade if ast_channel_masq() returns true. Now, there is no distinction between setting up a masquerade and performing the masquerade. It is one operation. The only remaining checks for ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not want to interrupt a masquerade by hanging up the channel. Instead, now ast_hangup() will wait for a masquerade to complete before moving forward with its operation. The ast_channel_move() function has been modified to basically in-line the logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has been killed off for real. ast_channel_move() now has a lock associated with it that is used to prevent any simultaneous moves from occurring at once. This means there is no need to make sure that ast_channel_masq() or ast_channel_masqr() are already set on a channel when ast_channel_move() is called. It also means the channel container lock is not pulling double duty by both keeping the container locked and preventing multiple masquerades from occurring simultaneously. The ast_do_masquerade() function has been renamed to do_channel_masquerade() and is now internal to channel.c. The function now takes explicit arguments of which channels are involved in the masquerade instead of a single channel. While it probably is possible to do some further refactoring of this method, I feel that I would be treading dangerously. Instead, all I did was change some comments that no longer are true after this changeset. The other more minor change introduced in this patch is to res_pjsip.c to make ast_sip_push_task_synchronous() run the task in-place if we are already a SIP servant thread. This is related to this patch because even when we isolate the channel masquerade to only running in the SIP servant thread, we would still deadlock when the fixup() callback is reached since we would essentially be waiting forever for ourselves to finish before actually running the fixup. This makes it so the fixup is run without having to push a task into a serializer at all. (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: https://reviewboard.asterisk.org/r/3069 ........ Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3ccd5dee18	udptl: Dead code elimination. ast_udptl_bridge was not used. Removing dead code starting with ast_udptl_bridge() eliminated the code in this change. Note: This code has actually been dead since Asterisk v1.4 when it was first put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e4803bbd9e	Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Scott Griepentrog	2882c5f9f1	astdb: crash in sqlite3 during shutdown When Asterisk is shut down, the astdb_atexit() function releases (finalize) the previously initiated (prepared) SQL statements in sqlite3. Another thread making a subsequent request can cause a crash in sqlite3. This patch eliminates that issue by resetting the statement pointer after it is released/cleared. The sqlite3 code detects the null pointer, and aborts the operation cleanly. (closes issue AST-1265) Reported by: Alexander Hömig (closes issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged revisions 404344 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404345 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	eb235ad05f	channel: Add a missing ast_channel_unlock when allocating a Surrogate channel. ........ Merged revisions 404332 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7e9febbf86	app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state When doing the rework of the CDR engine that pushed all of the logic into cdr.c and made it respond to changes in channel state over Stasis, we knew that accessing the CDR engine from the dialplan would be "slightly" non-deterministic. Dialplan threads would be accessing CDRs while Stasis threads would be updating the state of said CDRs - whereas in the past, everything happened on the dialplan threads. Tests have shown that "slightly" is in reality "very". This patch synchronizes things by making the dialplan applications/functions that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to send their requests over to the CDR engine, and synchronize on the channel Stasis topic via a subscription so that they return their values/control to the dialplan at the appropriate time. While going through this, the following changes were also made: * DISA, which can reset the CDR when a user successfully authenticates, now just uses the ResetCDR app to do this. This prevents having to duplicate the same Stasis synchronization logic in that application. * Answer no longer disables CDRs. It actually didn't work anyway - calling DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer time - it just kills all CDRs on that channel, which isn't what the caller would intend. (closes issue ASTERISK-22884) (closes issue ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/ ........ Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	04dfe2d77f	Add AMI event for presence state. Review: https://reviewboard.asterisk.org/r/3039/ ........ Merged revisions 404275 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404279 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	28c0cb28d0	channel locking: Add locking for channel snapshot creation Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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