asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	d014ad2261	func_channel: Read/Write after_bridge_goto option Allows reading and setting of a channel's after_bridge_goto datastore (closes issue ASTERISK-21875) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	68103abba3	Handle variable substitution in dummy variables When func_cdr is used for variable substitution, there is no channel name and hence no run-time information available for CDR variable substitution. In that case, the correct thing to do is to use the CDR object on the channel passed to the function. This patch checks to see if the channel passed in has a name - if not, it uses ast_cdr_format_var instead of ast_cdr_get_var. This allows CDR backends to continue to use variable substitution in order to resolve ast_cdr object properties. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4aa47d6893	Fix build warnings related to printf/scanf of tv_usec. The type of tv_usec is suseconds_t. On Linux, this is usually a long int, but the specification is actually pretty lax on what it might actually be. And, sadly, there's no printf/scanf width specifier for suseconds_t. So it could bit an int or a long, but there's not a great way to tell which it is. This patch fixes scanf by reading into a long temporary variable that's then stored into the tv_usec. It fixes printf by casting the tv_usec to a long first. This patch also adds some missing width specifiers for some debug statements, which would cause ".000001" to be displayed at ".1". git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392076 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now very reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b97c71bb11	Fix shutdown assertions in stasis-core In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b3bb6608ef	Migrate SHARED's use of the VarSet AMI event to Stasis-Core This patch removes the direct call to AMI from the SHARED function and instead call Stasis-Core. Stasis-Core delivers the notification that a shared variable has changed on a channel to all interested consumers. (issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	71a01725b8	Move presence state distribution to Stasis-core Convert presence state events to Stasis-core messages and remove redundant serializers where possible. Review: https://reviewboard.asterisk.org/r/2410/ (closes issue ASTERISK-21102) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1cafccdf87	Update documentation for CHANNEL function Document that you can read/write the 'accountcode' and 'amaflags' on a channel. ........ Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	220cc9b9af	Have func_curl log a warning when a curl request fails. Review: https://reviewboard.asterisk.org/r/2403/ ........ Merged revisions 383460 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383461 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	0b1e78cace	Minor cleanup in func_curl near hashcompat code. Review: https://reviewboard.asterisk.org/r/2402/ ........ Merged revisions 383457 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d7a616c945	PRESENCE_STATE: Provide better documentation for the 'e' option. Notes that the 'e' option actually decodes data when used as a write function such as with the SET application while it encodes data when used to read. Review: https://reviewboard.asterisk.org/r/2335/ ........ Merged revisions 381655 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c209e85ad3	Fix crash in PresenceState AMI action when specifying an invalid provider This patch fixes a crash in Asterisk that could be caused by using the PresenceState AMI action while providing an invalid provider. This patch also adds some additional warnings when a user attempts to provide the PresenceState action with invalid data, and removes some NOTICE statements that were still lurking in the code from testing. (closes issue AST-1084) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 381594 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	32ac38ea37	Improve func FRAME_TRACE DTMF digit format. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7d9871b394	Add ControlPlayback manager action This patch adds the capability for asynchronous manipulation of audio being played back to a channel though a new AMI action "ControlPlayback". The ControlPlayback action supports a number of operations, the availability of which depend on the application being used to send audio to the channel. When the audio playback was initiated using the ControlPlayback application or CONTROL STREAM FILE AGI command, the audio can be paused, stopped, restarted, reversed, or skipped forward. When initiated by other mechanisms (such as the Playback application), the audio can be stopped, reversed, or skipped forward. Review: https://reviewboard.asterisk.org/r/2265/ (closes issue ASTERISK-20882) Reported by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	89f9e077d7	Prevent crashes from occurring when reading from data sources with large values When reading configuration data from an Asterisk .conf file or when pulling data from an Asterisk RealTime backend, Asterisk was copying the data on the stack for manipulation. Unfortunately, it is possible to read configuration data or realtime data from some data source that provides a large blob of characters. This could potentially cause a crash via a stack overflow. This patch prevents large sets of data from being read from an ARA backend or from an Asterisk conf file. (issue ASTERISK-20658) Reported by: wdoekes Tested by: wdoekes, mmichelson patches: * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674) * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8fb5bdce9a	Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	0877ba5b37	Minor spelling fix to the VOLUME documentation. ........ Merged revisions 376919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376920 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376921 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a0c363e227	Refactor ast_timer_ack to return an error and handle the error in timer users Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	da85f8489f	Make evaluation of channel variables consistently case-sensitive. Due to inconsistencies in how variable names were evaluated, the decision was made to make all evaluations case-sensitive. See the UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for more details. (closes issue ASTERISK-20163) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2160 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	b106b77041	Title update Update title that was left behind many years ago. Used revision 6596 as my guide for what it should be. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fdfb3ae5fa	Allow for redirecting reasons to be set to arbitrary strings. This allows for the REDIRECTING dialplan function to be used to set the reason to any string. The SIP channel driver has been modified to set the redirecting reason string to the value received in a Diversion header. In addition, SIP 480 response reason text will set the redirecting reason as well. (closes issue AST-942) reported by Malcolm Davenport (closes issue AST-943) reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/2101 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	d9d7b1f3e3	"He who go through turnstile sideways is going to Bangkok" ........ Merged revisions 373582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	87370eeced	func_audiohookinherit: Document some missed sources. This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373468 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373470 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	6f61cb50c5	Doxygen Updates - janitor work Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen. Further updates coming. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	da5944fc56	Named call pickup groups. Fixes, missing functionality, and improvements. * ASTERISK-20383 Missing named call pickup group features: CHANNEL(callgroup) - Need CHANNEL(namedcallgroup) CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() - Needs to also select from named pickup groups. * ASTERISK-20384 Using the pickupexten, the pickup channel selection could fail even though there was a call it could have picked up. In a call pickup race when there are multiple calls to pickup and two extensions try to pickup a call, it is conceivable that the loser will not pick up any call even though it could have picked up the next oldest matching call. Regression because of the named call pickup group feature. * See ASTERISK-20386 for the implementation improvements. These are the changes in channel.c and channel.h. * Fixed some locking issues in CHANNEL(). (closes issue ASTERISK-20383) Reported by: rmudgett (closes issue ASTERISK-20384) Reported by: rmudgett (closes issue ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2112/ ........ Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8b90b7a565	Remove annoying unconditional debug message from INC/DEC functions. (closes issue AST-1001) Reported by: Guenther Kelleter ........ Merged revisions 372628 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372629 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372630 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	6a539ace84	Fix misuses of asprintf throughout the code. This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371591 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	78fea20ea7	Make the name of the "HangupCauseClear" application consistent The name of the "HangupCauseClear" application is "HangupCauseClear", not "HangupcauseClear". The incorrect case of 'cause' caused the XML documentation to not register properly. As an aside, this commit message felt very awkward, but I'm not sure how else to note that "X", which has to be "X", was referred to as "x". (closes issue ASTERISK-20253) Reported by: Andrew Latham Patches: hangupcause.diff uploaded by Andrew Latham (license #5985) ........ Merged revisions 371516 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fb6238899b	Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	49a6b4935e	Fix some presence-state unit test typos. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b6a0ae0b35	Unit tests for the Jitter Buffer API; remove unnecessary resync This patch includes the following: * Unit tests for the abstract Jitter Buffer API. This includes both fixed and adaptive flavors, testing nominal creation, frame input, frame retrieval, resyncing; off nominal frame input overflow, out of order, and others. * Tweaks to the abstract_jb API to remove the unnecessary resync_threshold parameter from the create function (resync_threshold is already in the struct passed into the create function) * Ensure the fixed jitter buffer is empty before destroying it, to avoid an ASSERT * Don't "resync" the adaptive jitter buffer. The mechanism that was being used actually causes the jitter buffer to think its being overflowed by going around the jitterbuf API and attempting to 'resynch' it improperly. If a resync is needed, the jitter buffer will do it properly by itself. Note that this is only an optimization needed for trunk, as the worst that happens is the loss of three voice packets before the adaptive jitter buffer will resync anyway. Review: https://reviewboard.asterisk.org/r/2035 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	ec14c2563e	Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370384 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	cb9756daa2	Add hangupcause translation support The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	79087cbbd5	Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	6761812566	Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369971 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	9bd9eb809c	Reverting last merge since it wasn't completed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	a8c12c6e67	Correct Documentation For DEC Function The documentation for DEC in func_math.c was incorrect. Looks like a copy and paste error. (Closes issue ASTERISK-20095) Reported by: Billy Chia Tested by: Michael L. Young Patches: func_math.patch uploaded by Billy Chia (license 6381) ........ Merged revisions 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369972 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tilghman Lesher	6190ae4430	Allow the REALTIME() function to report errors back to the caller. Also, do more error checking on the arguments specified to the REALTIME() function and clarify the documentation. While I was editing the file, a few coding guidelines fixups, as well. Review: https://reviewboard.asterisk.org/r/2031/ ........ Merged revisions 369937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ac35b92b62	Hangup handlers - Dialplan subroutines that run when the channel hangs up. Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	166b4e2b30	Multiple revisions 369001-369002 ........ r369001 \| kpfleming \| 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) \| 11 lines Add support-level indications to many more source files. Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. ........ r369002 \| kpfleming \| 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) \| 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	21997aa7bb	Fix a deadlock that occurs when func_volume is used on a local channel. This was discovered by trying to perform a call forward to an extension that makes use of func_volume. When the local channel is optimized away, the datastore on the local;2 channel would have its audiohook destroyed rather than detaching the audiohook from the channel and then destroying it. With this patch, func_volume's datastore destructor takes the proper route of detaching the audiohook and then destroying it. (closes issue ASTERISK-19611) reported by Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark Michelson (license #5049) ........ Merged revisions 368898 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368899 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	14a985560e	Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	a7a3050de9	Add documentation to function CHANNEL for options echocan_mode and buffers The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368093 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	dd2427c141	Coverity Report: Fix issues for error type REVERSE_INULL (core modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue ASTERISK-19648) Reported by: Matt Jordan ........ Merged revisions 368039 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368042 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6eb4e81033	Fix more memory leaks This patch adds to what was fixed in r366880. Specifically, it addresses the following: * chan_sip: dispose of an allocated frame in off nominal code paths in sip_rtp_read * func_odbc: when disposing of an allocated resultset, ensure that any rows that were appended to that resultset are also disposed of * cli: free the created return string buffer in another off nominal code path * chan_dahdi: free a frame that was allocated by the dsp layer if we choose not to process that frame (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........ Merged revisions 366944 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366948 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	7b51320642	Fix a variety of memory leaks This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar: responses are allocated via the various _request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	b5a6de76fc	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	dd81b047db	Resolve FORWARD_NULL static analysis warnings This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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671 Commits (c3c031569312fe85fb8e7f3c61c5140aadc2904a)