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2013-08-23Fix sorcery unit testsMatthew Jordan
When strict XML documentation checking was re-enabled, the test objects used in sorcery would fail to register as the types were not marked internal and the nodoc option wasn't used for the options. This fixes that problem, such that, as one would hope, they once again pass. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Fix memory corruption when trying to get "core show locks".Richard Mudgett
Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch in memory pools but had a math error determining the buffer size and didn't address other similar memory pool mismatches. * Effectively reverted the previous patch to go in the same direction as trunk for the returned memory pool of ast_bt_get_symbols(). * Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is enabled. * Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved __dump_backtrace() because of compile issues with the utils directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Prevent seg fault in off nominal path when registered option fails to validateMatthew Jordan
If an option is registered to a type and it is the last known type in the list of registered types, and the option fails to register, an overrun of the types array can occur due to the index variable having been already incremented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23PSJIP - sip.conf to res_sip.conf scriptKevin Harwell
Most, if not all, of the backing features of a conf file should now be implemented (e.g. multi-line comments, includes, templates, etc...). A few of the options still need to be mapped. Those are currently listed in the 'sip_to_res_sip.py' file. Things to do: (1) There is more work to do here, at least for the sip.conf items that aren't currently parsed. An issue will be created for that. (2) All of the scripts should probably be passed through pylint and have as many PEP8 issues fixed as possible. (3) A public review is probably warranted at that point of the entire script. Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23ARI: Correct error codes for bridge operationsDavid M. Lee
This patch adds error checking to ARI bridge operations, when adding/removing channels to/from bridges. In general, the error codes fall out as follows: * Bridge not found - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel not found - 400 Bad Request * Channel not in Stasis - 422 Unprocessable Entity * Channel not in this bridge (on remove) - 422 Unprocessable Entity (closes issue ASTERISK-22036) Review: https://reviewboard.asterisk.org/r/2769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Update CHANGES file to reflect pass through support for Opus/VP8Matthew Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Add pass through support for Opus and VP8; Opus format attribute negotiationMatthew Jordan
This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Update config framework/sorcery with types/options without documentationMatthew Jordan
There are times when a configuration option should not have documentation. 1. Some options are registered with a particular object merely as a warning to users. These options aren't even really 'deprecated' - which has its own separate API call - they are actually provided by a different configuration file. The options are merely registered so that the user gets a warning that a different configuration file provides the item. 2. Some object types - most notably some used by modules that use sorcery - are completely internal and should never be shown to the user. 3. Sorcery itself has several 'hidden' fields that should never be shown to a user. This patch updates the configuration framework and sorcery with additional API calls that allow a module to register types as internal and options as not requiring documentation. This bypasses the XML documentation checking. This patch also re-enables the strict XML documentation checking in trunk, as well as updates some documentation that was missing. Review: https://reviewboard.asterisk.org/r/2785/ (closes issue ASTERISK-22359) Reported by: Matt Jordan (closes issue ASTERISK-22112) Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Fix crash when answering after a transport error occurs.Joshua Colp
If a response to an initial incoming INVITE results in a transport error the INVITE transaction is removed from the INVITE session. Any attempts to answer the INVITE session after this results in a crash as it requires the INVITE transaction to exist. This change explicitly locks the dialog and checks to ensure that the INVITE transaction exists before answering. (closes issue AST-1203) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Update CEL sample configKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23ARI: Music on Hold/Background Music for bridgesJonathan Rose
Adds ARI functions to be able to turn on/off music on hold in a bridge. It actually functions more as a background music without further actions on the bridge since if the rest of the channels in the bridge aren't explicitly muted, they will still be able to communicate. (closes issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2688/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Minor tweaks with ast_moh_start() callers.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Add SayAlphaCase and similar functionality for AGIKinsey Moore
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
2013-08-22res_sip_dtmf_info: Support sending of 'raw' DTMFKevin Harwell
Added the ability to handle 'raw' DTMF within the body of an INFO message. Also made it so values 10-16 are mapped to valid DTMF values. (closes issue ASTERISK-22144) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Add missing configOption close tagsKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Update MOH start/stop routine doxygen.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Fix missing xml doc configOption 'type' for for both 'system' and 'global' ↵Rusty Newton
configObjects (issue ASTERISK-22344) (closes issue ASTERISK-22344) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Bridge API: Set a cause code on a channel when it is ejected from a bridge.Richard Mudgett
The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Ensure CEL creates a default config if it isn't provided with oneKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Remove set but unused variable 'meid'.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Fix crash when getting CEL configKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Massively clean up app_queue.Mark Michelson
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
2013-08-22Handle default body types for SIP event packages in res_pjsip_pubsubMark Michelson
Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Made the abstract jitter buffer resync on some more control frames.Richard Mudgett
Resync the abstract jitter buffer on the following additional control frames: AST_CONTROL_HOLD AST_CONTROL_UNHOLD AST_CONTROL_T38_PARAMETERS git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Make CEL behavior conform to the documentationKinsey Moore
This modifies the behavior of the CEL engine to conform to documented behavior for Asterisk 12 as defined on the wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification The primary changes deal with removal of the peer field from function calls since it is no longer directly relevant to the bridging system and removal of the layer of CDR-like business logic that was providing a partial emulation of Asterisk 11 CEL functionality. With this change, there is no longer a distinction between "bridges" and "conferences" and all participation changes are denoted with bridge enter and bridge exit messages. This updates the CEL unit tests to handle these changes and simplifies some of the macros used in the process. This also fixes a segfault when attempting to ref a configuration that failed to load. Review: https://reviewboard.asterisk.org/r/2788/ (issue ASTERISK-21567) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Update BUGBUG comment.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Don't store repeated commands in the editline history buffer.Walter Doekes
The equivalent of bash HISTCONTROL=ignoredups. Review: https://reviewboard.asterisk.org/r/2775/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22Add _IO_stdin_used in version-script to fix SIGBUSes on Sparc.Walter Doekes
The --version-script,asterisk.exports linker flag (and the module exports) didn't provide _IO_stdin_used in the list of exported symbols. That causes some kind of libc compatibility mode to kick in, where stdio file structures (stdout/stderr) land somewhere else. In the case of the Sparc, they landed on misaligned memory. This became apparent first after r376428 (Reorder startup sequence) when a lot of ast_log's were replaced with fprintf's. Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64 architectures, the Sparc is very picky about memory alignment.) (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy Kister Review: https://reviewboard.asterisk.org/r/2760/ ........ Merged revisions 397377 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397378 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21UDPTL: Fix a regression where UDPTL won't load default settingsJonathan Rose
If the file udptl.conf is unavailable at startup, UDPTL will fail to initialize and while it makes some noise, it isn't immediately obvious why consumers start to fail when using it. This patch makes UDPTL load as though an empty config was provided when udptl is unavailable at startup. (closes issue ASTERISK-22349) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2773/ ........ Merged revisions 397365 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21* Move ast_bridge_channel_setup_features() into bridge_basic.c.Richard Mudgett
* Made application map hooks be removed on a basic bridge personality change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Deferred some more BUGBUG comments to a JIRA issue or XXX comment.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Complete http_shutdown.David M. Lee
This patch frees up some resources allocated in http.c. * tcp listeners stopped * tls settings freed * uri redirects freed * unregister internal http.c uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell Patches: http.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 397308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397309 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Set 14400 as the default max bit rate if T38MaxBitRate is not specifiedMatthew Jordan
If an endpoint fails to include the T38MaxBitRate attribute during negotiation, Asterisk will negotiate a bit rate of 2400 instead of the ITU recommended bit rate of 14400. This patch fixes this by making AST_T38_RATE_14400 the 'default' value of the enum by assigning it a value of 0, such that if an endpoint fails to include the attribute, the default will be 14400. Note that Walter Doekes included the nice comment in frame.h about why we are purposefully assigning AST_T38_RATE_14400 a value of 0. (closes issue ASTERISK-22275) Reported by: Andreas Steinmetz patches: fax-fix.patch uploaded by anstein (License 6523) ........ Merged revisions 397256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397257 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21ARI: Correct segfault with /variable calls are missing ?variable parameter.David M. Lee
Both /asterisk/variable and /channel/{channelId}/variable requires a ?variable parameter to be passed into the query. But we weren't checking for the parameter being missing, which caused a segfault. All calls now properly return 400 Bad Request errors when the parameter is missing. The Swagger api-docs were updated accordingly. (closes issue ASTERISK-22273) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21ARI: Remove the 'channel:' scheme from endpoint's channel list.David M. Lee
For times when a reference in ARI might be ambiguous, the reference is built as an URI (such as channel:1376341790.3). An endpoint's channel list is not ambiguous, and in fact the field is named 'channel_ids', but it had channel URI's instead of channel id's. This patch changes the list to be the raw id instead of the URI. (closes issue ASTERISK-22291) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21res_stasis: remove call to missing function control_continue.David M. Lee
In the shuffling around of res_stasis, control_continue was renamed to stasis_app_control_continue, but the call in res_stasis wasn't updated. In looking into it, it turns out it wasn't really the right thing to do in res_stasis anyways. This patch changes the handling of received a AST_CONTROL_HANGUP frame to be the same as receiving a NULL frame, and removed the declaration of control_continue(), since it doesn't exist any more. (closes issue ASTERISK-22292) Reported by: Denis Smirnov git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Fix several interrelated issues dealing with the holding bridge technology.Richard Mudgett
* Added an option flags parameter to interval hooks. Interval hooks now can specify if the callback will affect the media path or not. * Added an option flags parameter to the bridge action custom callback. The action callback now can specify if the callback will affect the media path or not. * Made the holding bridge technology reexamine the participant idle mode option whenever the entertainment is restarted. * Fixed app_agent_pool waiting agents needlessly starting and stopping MOH every second by specifying the heartbeat interval hook as not affecting the media path. * Fixed app_agent_pool agent alert from restarting the MOH after the alert beep. The agent entertainment is now changed from MOH to silence after the alert beep. * Fixed holding bridge technology to defer starting the entertainment. It was previously a mixture of immediate and deferred. * Fixed holding bridge technology to immediately stop the entertainment. It was previously a mixture of immediate and deferred. If the channel left the bridging system, any deferred stopping was discarded before taking effect. * Miscellaneous holding bridge technology rework coding improvements. Review: https://reviewboard.asterisk.org/r/2761/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Prevent a crash on outbound SIP MESSAGE requests.Mark Michelson
If a From header on an outbound out-of-call SIP MESSAGE were malformed, the result could crash Asterisk. In addition, if a From header on an incoming out-of-call SIP MESSAGE request were malformed, the message was happily accepted rather than being rejected up front. The incoming message path would not result in a crash, but the behavior was bad nonetheless. (closes issue ASTERISK-22185) reported by Zhang Lei ........ Merged revisions 397254 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Allow channels in app_stasis to hangup properlyKinsey Moore
This detects hangups that occur while bridged to allow channels to exit app_stasis even if the hangup frame was absorbed by the bridge the channel was in. Reported by: David Lee (closes issue ASTERISK-22297) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21Allow the SIP_CODEC family of variables to specify more than one codecMatthew Jordan
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
2013-08-21Fix Not Storing Current Incoming Recv AddressMichael L. Young
In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping the old recv address since recv was already set. This has caused a problem when a proxy is involved since responses to incoming requests from the proxy server, after an outbound call is established, are never sent to the correct recv address. In 11, r382322 introduced this regression. The fix is to revert that change and always store the recv address on incoming requests. Thank you Walter Doekes for helping to point out this error and Mark Michelson for your input/review of the fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin Tested by: Alex Zarubin, Karsten Wemheuer Patches: asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026) ........ Merged revisions 397204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397205 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Localize and rename ACL configuration.Mark Michelson
This is more-or-less a reversion of previous ACL behavior so that it is more self-contained. ACL sections are now only parsed if res_pjsip_acl.so is loaded. Moreover, the configuration section is now "type=acl" instead of "type=security". The original reason for having ACLs configured in a "type=security" section was to lump ACLs and other security-related items into the same section. The problem is that ACLs really should be in their own sections and there are no other security-related options implemented anyways. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Remove REF_DEBUG definition.Mark Michelson
........ Merged revisions 397156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397157 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Fix refcounting of sip_pvt in test_sip_rtpqos test and unlink it from the ↵Mark Michelson
list of pvts. (closes issue ASTERISK-22248) reported by Corey Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell (license #5909) ........ Merged revisions 397112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397133 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Clarify documentation for the "identify_by" option for SIP endpoints.Mark Michelson
This also removes documentation for the options that no longer exist. (closes issue ASTERISK-22306) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Unregister CLI commands on exitKinsey Moore
This patch ensures that CLI commands enabled by DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell Tested by: Corey Farrell Patches: debug_cli_unregister.patch uploaded by Corey Farrell ........ Merged revisions 397106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397107 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Add debug message to res_pjsip_endpoint_identifier_ip to indicate when an ↵Mark Michelson
endpoint is successfully retrieved. (closes issue ASTERISK-22101) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Add warning messages for registration failure paths.Mark Michelson
(closes issue ASTERISK-22089) reported by Rusty Newton patches: patch1.txt uploaded by John Bigelow (License #5091) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Add note to transport configuration that a restart is required to change ↵Mark Michelson
transports. (closes issue ASTERISK-22094) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20Recorded merge of revisions 397067 from ↵Kinsey Moore
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Fix xmldoc memory leak This fixes a single-attribute memory leak that was occurring when the "required" attribute was not true. (closes issue ASTERISK-22249) Reported by: Corey Farrell Tested by: Corey Farrell Patches: xmldoc-free_attr_required.patch uploaded by Corey Farrell ........ Merged revisions 397064 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 00:26:06 +0000