asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	a610bfa9e8	CHANGES: Update changes log to include r403414 entry Adds note of additional 0 for operator option on app_record git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	7f18df5699	CHANGES: Update changes log to include new bridge fields added in r404042 ........ Merged revisions 407419 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9b93917896	ARI/AMI: Update versions; update UPGRADE/CHANGES notes for 12.1.0 changes Due to backwards compatible changes made to AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0, respectively. ........ Merged revisions 407402 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	10e38fb10c	res_pjsip: Config option to enable PJSIP logger at load time. Added a "debug" configuration option for res_pjsip that when set to "yes" enables SIP messages to be logged. It is specified under the "system" type. Also added an alembic script to add the option to realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged revisions 407036 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	aeb4466656	ChanSpy: Add ability to specify channel uniqueids as well as channel names. * Made ChanSpy accept a channel uniqueid or a fully specified channel name as the chanprefix parameter if the 'u' option is specified. (closes issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	8f82eb0098	manager: ExtensionStatus event status human readable Added a note in the changes file about the new 'StatusText' field that was added to the 'ExtensionStatus' event. (issue ASTERISK-23154) Reported by: Jonathan Rose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Walter Doekes	72cb7a254f	Enable wide band audio in musiconhold streams. Review: https://reviewboard.asterisk.org/r/3112/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3c90fc0bfd	Make 12 - 12.1 CHANGES log the same as in 12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	8ba05ae67e	Include CHANGES info for r405553 ........ Merged revisions 405585 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	821ab51381	res_pjsip: add 'set_var' support on endpoints Added a new 'set_var' option for ast_sip_endpoint(s). For each variable specified that variable gets set upon creation of a pjsip channel involving the endpoint. (closes issue ASTERISK-22868) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3095/ ........ Merged revisions 404663 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	dbead14c3b	Put notice in CHANGES as well as UPGRADE.txt. ........ Merged revisions 404375 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 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	12 years ago
Jonathan Rose	f6e92c35df	app_page: Add predial handlers for app_page. (closes issue AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d421818c3d	Add a CONFBRIDGE_RESULT channel variable to discern why a channel left a ConfBridge. Review: https://reviewboard.asterisk.org/r/3009 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	5730410861	Create function for retrieving Mixmonitor instance data. For the time, this is only useful for retrieving the filename. The purpose of this function is to better facilitate multiple mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel variable is not conducive to such behavior, so allowing finer grained access to individual mixmonitor properties improves the situation. The MIXMONITOR_FILENAME channel variable is still set, though, so there is no worry about backwards compatibility. Review: https://reviewboard.asterisk.org/r/3023 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	18c2cfa7b7	PickupChan: Add ability to specify channel uniqueids as well as channel names. * Made PickupChan() search by channel uniqueids if the search could not find a channel by name. * Ensured PickupChan() never considers the picking channel for pickup. * Made PickupChan() option p use a common search by name routine. The original search was erroneously case sensitive. (issue AFS-42) Review: https://reviewboard.asterisk.org/r/3017/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a60764d61e	app_directory: Set variable indicating reason directory exited By the time the directory application exits, a channel variable DIRECTORY_RESULT will be set for the channel that invoked it which can be used to determine the reason for exit. The changes log and the app_directory documentation contain specific details about each of the possible values for DIRECTORY_RESULT. Review: https://reviewboard.asterisk.org/r/3016/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	7950118e18	Confbridge: Add option to review the recording similar to announce_join_leave Review: https://reviewboard.asterisk.org/r/3008/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ad0e70ba83	Say: If SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF Similar to how background works, if a say application is called with this variable set to 'true', 'yes', 'on', etc. then using DTMF while the say action is in progress will result in the channel jumping to that extension in the dialplan. Review: https://reviewboard.asterisk.org/r/3011/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bf5492abd2	security_events: Push out security events over AMI events Security Events will now be written to any listener of the new 'security' class Review: https://reviewboard.asterisk.org/r/2998/ ........ Merged revisions 402584 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4b7ff87492	app_confbridge: Make the CONFBRIDGE function be able to create dynamic menus Also adds the ability to clear all profile items and makes behavior more consistent with documentation as when choosing whether to use CONFBRIDGE datastore profiles or the application arguments to the confbridge application. (closes issue ASTERISK-22760) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2971/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f87086b374	app_confbridge: Can now set the language used for announcements to the conference. ConfBridge now has the ability to set the language of announcements to the conference. The language can be set on a bridge profile in confbridge.conf or by the dialplan function CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983) Reported by: Jonathan White Patches: M19983_rev2.diff (license #5138) patch uploaded by junky (modified) Tested by: rmudgett ........ Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	2af53640c8	Add IPv6 Support To chan_iax2 This patch adds IPv6 support to chan_iax2. Yay! (closes issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged revisions 400567 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b44ce141e5	chan_sip: Allow Asterisk to retry after 403 on register This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. This also adds a similar per-outbound-registration option to chan_pjsip which allows the retry interval to be altered for 403 responses to REGISTER requests. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400140 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9bed50db41	optional_api: Fix linking problems between modules that export global symbols With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	449afdd9e8	Revert r394939 due to (numerous) objections The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter and Tzafrir have pointed out numerous issues with the approach and have propsed an alternative in r/2757. Since it's not a time critical issue and is not worth holding up the release of 12 for it, I've gone ahead and reverted r394939 from 12/trunk and re-opened ASTERISK-21965. ........ Merged revisions 397938 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	629f43d2b6	Add database schema management using Alembic This patch replaces contrib/realtime/ with a new setup for managing the database schema required for database integration with Asterisk. In addition to initializing a database with the proper schema, alembic can do a database migration to assist with upgrading Asterisk in the future. Hopefully this helps make setting up and operating Asterisk with a database easier. With this the schema only needs to be maintained in one place instead of once per database. The schemas I have added here have a bit of improvement over the examples that were there before (some added consistency and added some missing indexes). Managing the schema in one place here also applies to all databases supported by SQLAlchemy. See contrib/ast-db-manage/README.md for more details. Review: https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant (license 6300) ........ Merged revisions 397874 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b7ec25ec2e	Update CHANGES file for Asterisk 12 This updates the Asterisk 12 CHANGES file with the things that were missed during the development cycle. Review: https://reviewboard.asterisk.org/r/2795/ ........ Merged revisions 397870 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9f4849724f	Update CHANGES file to reflect pass through support for Opus/VP8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7b032c1adb	Add SayAlphaCase and similar functionality for AGI This adds a new dialplan application, SayAlphaCase, that performs much the same function as SayAlpha except that it takes additional options which allow the user to specify whether the case of each letter should be announced for uppercase, lowercase, or all letters. Similar functionality has been added to the SAY ALPHA AGI command via an optional parameter. Original Patch by: Kevin Scott Adams Reported by: Kevin Scott Adams Review: https://reviewboard.asterisk.org/r/2725/ (closes issue ASTERISK-20782) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	00baddb906	Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e85dd76945	Allow the SIP_CODEC family of variables to specify more than one codec The SIP_CODEC family of variables let you set the preferred codec to be offered on an outbound INVITE request. However, for video calls, you need to be able to set both the audio and video codecs to be offered. This patch lets the SIP_CODEC variables accept a comma delineated list of codecs. The first codec in the list is set as the preferred codec; additional codecs are still offered however. This lets a dialplan writer set both audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264) Note that this feature was written by both Dennis Guse and Frank Haase Review: https://reviewboard.asterisk.org/r/2728 (closes issue ASTERISK-21976) Reported by: Denis Guse Tested by: mjordan, sysreq patches: patch-channels-chan__sip.c-393919 uploaded by dennis.guse (license 6513) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	38236e54a8	Remove dead code from features.c; refactor pickup code into pickup.c This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c8a91b5b01	Add queue member paused hints This patch adds the ability in Queue to raise a hint when a member's paused state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name} are the name of the queue and the name of the member to subscribe to, respectively. For example: exten => 8501,hint,Queue:sales_pause_mark. Members will show as In Use when paused. Note that the format of the queue pause hint was changed slightly from what is on the issue to accomodate suggestion on the code review. Review: https://reviewboard.asterisk.org/r/2254 (closes issue ASTERISK-20842) Reported by: Philippe Lindheimer patches: qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9ea74182ef	Update CONTROL STREAM FILE to accept an 'offsetms' parameter This patch allows starting playback of audio through the CONTROL STREAM FILE AGI command to start at a particular offset. It will also return the final position of the file in the 'endpos' attribute. (closes issue ASTERISK-17803) Reported by: Murray Melvin patches: res_agi.c.r316293.diff uploaded by murraytm (license 6221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	54803338b4	Always install safe_asterisk; add configuration file support This patch modifies the behavior of safe_asterisk in two ways: (1) It modifies the Asterisk Makefile such that safe_asterisk is always installed on a 'make install'. This was done as bugfixes in the safe_asterisk script were not applied in previous version of Asterisk without first removing the old version of the script. (2) In order to keep a newly installed version of safe_asterisk from impacting local modifications, a new config file - safe_asterisk.conf.sample - has been provided. Settings that were previously modified in safe_asterisk can be set there instead. (closes issue ASTERISK-21965) Reported by: Jeremy Kister patches: safe_asterisk.patch uploaded by jkister (License 6232) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3a5b68f07c	Allow setting allowmultiplelogin on an account basis This patch modifies manager to allow the allowmultiplelogin setting to be set on an account by account basis. When set in the general context, it will act as the default for the defined accounts. Setting it in the account will override the general setting. (closes issue ASTERISK-21324) Reported by: vldmr patches: asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c3b8939be8	Add CEL local optimization record type This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent local channel optimizations. Local channel optimizations were one of several things conveyed by the now defunct BRIDGE_UPDATE event type. This also adds a unit test to test generation of this new CEL event. Review: https://reviewboard.asterisk.org/r/2676/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	684c83b29b	Add transfer support to CEL This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	19d8f8c8e4	Add 'kick all' capability to ConfBridge CLI command This patch adds the ability to kick all users out of a conference from the ConfBridge kick CLI command. It is invoked by passing 'all' as the channel parameter to the CLI command, i.e., "confbridge kick <conf> all". Note that this patch was modified slightly to conform to trunk. (closes issue ASTERISK-21827) Reported by: dorianlogan patches: kickall-patch_v2.diff uploaded by dorianlogan (License 6504) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d43b17a872	Replace chan_agent with app_agent_pool. The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	30d379851e	Create Local channel messages on the Stasis message bus and produce AMI events This patch does the following: * It adds a virtual table of callbacks to core_unreal. These callbacks can be supplied by concrete implementations of "unreal" channel drivers, which lets the unreal channel driver call specific functionality when it performs some action. Currently, this is done to notify implementations when an optimization operation has begun, and when an optimization operation has succeeded. * It adds Stasis-Core messages for Local channel bridging and Local channel optimization. Local channel optimization is now two events: a Begin and an End. Some consumers of Stasis-Core may want to know when an operation is beginning so that they can 'prepare' their information; others will be more concerned about when the operation has completed, so that they can 'fix up' information. Stasis-Core allows for both, as does AMI. Review: https://reviewboard.asterisk.org/r/2552 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b193c2873d	Handle hangup logic in the Stasis message bus and consumers of Stasis messages This patch does the following: * It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a channel is executing dialplan hangup logic, i.e., the 'h' extension or a hangup handler. Stasis messages now also convey the soft hangup flag so consumers of the messages can know when a channel is executing said hangup logic. * It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs, and other consumers of Stasis have been updated to look for this flag to know when the channel should by lying six feet under. * The CDR engine has been updated to better handle a channel entering and leaving a bridge. Previously, a new CDR was automatically created when a channel left a bridge and put into the 'Pending' state; however, this way of handling CDRs made it difficult for the 'endbeforehexten' logic to work correctly - there was always a new CDR waiting in the hangup logic and, even if 'ended', wouldn't be the CDR people wanted to inspect in the hangup routine. This patch completely removes the Pending state and instead defers creation of the new CDR until it gets a new message that requires a new CDR. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	93ed5ef0ff	res_parking: Replace Parker snapshots with ParkerDialString This process also involved a large amount of rework regarding how to redial the Parker when a channel leaves a parking lot due to timeout. An attended transfer channel variable has been added to attended transfers to extensions that will eventually park (but haven't at the time of transfer) as well. This resolves one of the two BUGBUG comments remaining in res_parking. (issues ASTERISK-21877) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2638/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	02f55a36a0	Revert accidental overcommit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b4e9a3fc2f	Add BUGBUG note for ASTERISK-22009 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	85ba063329	Add a SystemName field to all AMI events. This only gets sent out if configured in asterisk.conf (closes issue ASTERISK-21494) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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847 Commits (a610bfa9e8341bebfb44d656458c97189bdf701d)