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2014-09-04Manager: Require read permission for SYSTEM in order to send FullyBootedJonathan Rose
Review: https://reviewboard.asterisk.org/r/3969/ ........ Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422625 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422626 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-09-01main/cli: Do not attempt to show CDR data for internal channelsMatthew Jordan
Internal channels don't have CDRs. Querying the CDR engine for their variables will make it cranky. ........ Merged revisions 422506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-30manager: Make WaitEvent action respect eventfiltersGeorge Joseph
A WaitEvent issued via an http session isn't respecting eventfilters defined for the user. I just added a match_filter to the predicate that controls astman_append. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3958/ ........ Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422440 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422441 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-28sched: Fix typo and whitespace change.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-27CallerID: Fix parsing of malformed calleridKinsey Moore
This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-26Fix race condition in the scheduler when deleting a running entry.Mark Michelson
When scheduled tasks run, they are removed from the heap (or hashtab). When a scheduled task is deleted, if the task can't be found in the heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled, this assertion causes a crash. The problem is, sometimes it just so happens that someone attempts to delete a scheduled task at the time that it is running, leading to a crash. This change corrects the issue by tracking which task is currently running. If that task is attempted to be deleted, then we mark the task, and then wait for the task to complete. This way, we can be sure to coordinate task deletion and memory freeing. ASTERISK-24212 Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3927 ........ Merged revisions 422070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-22main/message: Add a new-line to a DEBUG messageMatthew Jordan
........ Merged revisions 421859 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-21ARI: Fix implicit answer when playback is initiated on unanswered channelMatthew Jordan
When issuing a POST /channels/{channel_id}/play on a channel that is not yet answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS on the channel * Start up the playback of the media Instead, we sneak an answer on the channel right before starting playing media. This is due to ARI's usage of control_streamfile. This function implicitly answers the channel (and doesn't give ARI the option to stop it). The answering of the channel here is probably unnecessary: * app_voicemail, by far the biggest consumer of this function, always answers the channels anyway * control stream file (in res_agi) and ControlPlayback probably shouldn't be implicitly answering the channel. Answering should not be tied directly to playing back media. As it turns out, the answering of the channel here is pretty old: 356042 twilson if (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res = ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that others ran into this problem and commented about it on various mailing lists. Review: https://reviewboard.asterisk.org/r/3907/ ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged revisions 421695 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-21Clean up files that do not end with newlinesMatthew Jordan
Trivial patch to add new lines to several files missing them. This fixes warnings when compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close Reported by: Shaun Ruffell patches: 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417) ........ Merged revisions 421677 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-21uri: Quiet warning about type qualifiers ignored on function return typeMatthew Jordan
This patch fixes gcc warnings that occur due to the type qualifier 'const' being ignored on a return type of int. ASTERISK-24246 #close Reported by: Shaun Ruffell patches: 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-20chan_pjsip: Update media translation paths when new SDP negotiated.Richard Mudgett
On a SIP reinvite that changes media strams, the PJSIP channel driver was flooding the log with "Asked to transmit frame type %s, while native formats is %s" warnings. * Fixes PJSIP not setting up translation paths when the formats change on a reinvite. AFS-63 was effectively reintroduced because of the media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the unexpected frame format WARNING message to include more information. * Added protective locking while altering formats on a channel. Reworked set_format() to simplify and protect the formats under manipulation. * Restored some code that got lost in the media_formats work. (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3906/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-20cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.Richard Mudgett
filename_completion_function() returns memory that was not allocated by the MALLOC_DEBUG allocation tracker so the memory must be freed by ast_std_free(). ........ Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421602 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421608 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-20Stasis: Add information to blind transfer eventKinsey Moore
When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-19AMI Docs: Fix Status channel parameter optionalityKinsey Moore
........ Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421443 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421444 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-15Bridging: Fix a behavioral change when checking if a channel is leaving a bridgeJonathan Rose
r420934 introduced some failures in the test suite. Upon investigating, it was discovered that differences in the way we were evaluating whether a channel was in the process of leaving a bridge were causing some reinvites not to occur (mostly reinvites back to Asterisk when ending a call). This patch fixes that behavioral change. ASTERISK-24027 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3910/ ........ Merged revisions 421186 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-15app_voicemail/app: Remove test events that were duplicated by r421059Matthew Jordan
Moving the test event raised when a file is played back (which occurred in r421059) broke the ever loving snot out of the voicemail tests. This caused duplicate test events to get raised, as app_voicemail and main/app were raising events prior to call ast_streamfile. The voicemail tests did not enjoy getting multiple events. Since raising the playback event in ast_streamfile is far more useful to the vast majority of tests, this patch keeps the call there and simply removes the extraneous calls that duplicated the event. ........ Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-14main/file: Move test event to emit PLAYBACK event more consistentlyMatthew Jordan
This is being done in advance of the test for ASTERISK-23953 ........ Merged revisions 421059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421060 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421061 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-14cel: Make sure channels in extra fields include their unique IDs as wellMatthew Jordan
CEL typically tracks a lot of information using the unique ID of the channel. This is typically needed due to tying events together using the linked ID of the various channels involved in a "call", which is derived from the channel ID of the oldest channel involved in a bridge (or in the case of a Dial, the parent channel). Previously, we had updated the extra fields to include the involved channel names, but forgot to put in the unique ID. This patch corrects that error. ........ Merged revisions 421037 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-14channel_internal_api.c: Replace some code with ao2_replace().Richard Mudgett
Use ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace() has the advantange of not altering the ref count if the replaced pointer is the same. Review: https://reviewboard.asterisk.org/r/3904/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-13Bridges: Fix feature interruption/unintended kick caused by external actionsJonathan Rose
If a manager or CLI user attached a mixmonitor to a call running a dynamic bridge feature while in a bridge, the feature would be interrupted and the channel would be forcibly kicked out of the bridge (usually ending the call during a simple 1 to 1 call). This would also occur during any similar action that could set the unbridge soft hangup flag, so the fix for this was to remove unbridge from the soft hangup flags and make it a separate thing all together. ASTERISK-24027 #close Reported by: mjordan Review: https://reviewboard.asterisk.org/r/3900/ ........ Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-13AMI: Improve documentation for Status actionKinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-13logger: Don't store verbose-magic in the log files.Walter Doekes
In r399267, the verbose2magic stuff was edited. This time it results in magic characters in the log files for multiline messages. In trunk (and 13) this was fixed by the "stripping" of those characters from multiline messages (in r414798). This fix is altered to actually strip the characters and not replace them with blanks. Review: https://reviewboard.asterisk.org/r/3901/ Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged revisions 420897 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420898 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11Improve call forwarding reporting, especially with regards to ARI.Mark Michelson
This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11general: Fix memory Corruption in __ast_string_field_ptr_build_va.Walter Doekes
If the space left in a stringfield is between 0 and (alignof(ast_string_field_allocation)-1) adding new data would cause memory corruption, because we would assume enough space (unsigned underrun). Thanks Arnd Schmitter for reporting and finding out the cause! ASTERISK-23508 #close Reported by: Arnd Schmitter Tested by: Arnd Schmitter, JoshE Review: https://reviewboard.asterisk.org/r/3898/ ........ Merged revisions 420680 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420715 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420716 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-11tcptls: Avoid compiler warning on non-dev-mode.Walter Doekes
........ Merged revisions 420654 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420655 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420656 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-08main/message: remove debug messageMatthew Jordan
........ Merged revisions 420533 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07Stasis: Correct blind transfer message generationKinsey Moore
This fixes the json object creation format string and key name for the BridgeBlindTransfer Stasis event allowing it to be published properly. ........ Merged revisions 420414 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07Stasis: Ensure transfer messages follow validation rulesKinsey Moore
This makes Stasis() event generation for transfer messages follow validation rules. Currently, ast_json_null() is being used in place of omitting a key entirely which falls afoul of these validation rules. https://reviewboard.asterisk.org/r/3892/ ........ Merged revisions 420408 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07Ensure bridges exist when trying to determine bridged parties when ↵Mark Michelson
publishing transfer information. ........ Merged revisions 420387 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07Add support for RFC 4662 resource list subscriptions.Mark Michelson
This commit adds the ability for a user to configure a resource list in pjsip.conf. Subscribing to this list simultaneously subscribes the subscriber to all resources listed. This has the potential to reduce the amount of SIP traffic when loads of subscribers on a system attempt to subscribe to each others' states. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07chan_iax2: Several media format fixes.Richard Mudgett
* Fixed the iax.conf bandwidth option. This is the root cause of ASTERISK-24150. * Added checks in iax2_request() to ensure that there are actual formats requested for the new channel to prevent any more fracks from issues like ASTERISK-24150. This is a consequence of the iax.conf bandwidth option not working. * Fixed struct iax2_codec_pref.order member size mismatch issue when converting to and from the codec preference order list passed over the wire. In addition the values sent over the wire are now compatible with previous Asterisk versions. * Fixed several issues dealing with the struct iax2_codec_pref members. Off-by-one, array limit errors, and the order/framing members always need to be updated together. * Made iax2_request() setup the channel's native format preference order according to the user's wishes. The new media format strategy needs the order specified earler. * Fixed usage of ast_format_compatibility_bitfield2format(). The function can return NULL if the bitfield was not associated with a function. * Deleted dead code iax2_codec_pref_getsize() and iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8. * Renamed prefs to prefs_global so it won't get confused with the local pref versions. * Fixed too small buffer in handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete lines. * Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an optimization so iax2_request() and iax2_call() do less work. * Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when the pbx could not get started. * Made set_config() setup a local prefs list along side the local capability format bitfield. Once the config is loaded, then the local copies are put into the global versions. * Fix unininialized codec_buf in function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3890/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07Stasis: Convey transfer information to applicationsKinsey Moore
This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-07pbx: Filter out pattern matching hints in responses sent to ExtensionStateListMatthew Jordan
Hints that are a pattern match are technically stored in the hint container in the same fashion as concrete implementations of hints. The pattern matching hints, however, are not "real" in the sense that things can subscribe to them: rather, they are stored in the hints container so that when a subscription is made a "real" hint can be generated for the subscription if one does not yet exist. The extension state core takes care of this correctly by matching against non-pattern matching extensions prior to pattern matching extensions. Because of this, however, the ExtensionStateList AMI action was returning pattern matching hints when executed. These hints are meaningless from the perspective of AMI clients: their state will never change, they cannot be subscribed to, and events would never normally be generated from them. As such, we now filter these out of the response. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-06pbx_lua: fix regression with global sym export and context clash by pbx_config.George Joseph
ASTERISK-23818 (lua contexts being overwritten by contexts of the same name in pbx_config) surfaced because pbx_lua, having the AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before pbx_config. Since I couldn't find any reason for pbx_lua to export it's symbols to the rest of Asterisk, I simply changed the flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't realize was that the symbols need to be exported not because Asterisk needs them but because any external Lua modules like luasql.mysql need the base Lua language APIs exported (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's an issue in pbx.c where context_merge was only merging includes, switches and ignore patterns if the context was already existing AND has extensions, or if the context was brand new. If pbx_lua is loaded before pbx_config, the context will exist BUT pbx_lua, being implemented as a switch, will never place extensions in it, just the switch statement. The result is that when pbx_config loads, it never merges the switch statement created by pbx_lua into the final context. This patch sets pbx_lua's modflag back to AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge that catches the case where an existing context has includes, switchs or ingore patterns but no actual extensions. ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo Teräs Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3891/ ........ Merged revisions 420146 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 420147 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 420148 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-06Stasis: Allow message types to be blockedKinsey Moore
This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-05Multiple revisions 420089-420090,420097Matthew Jordan
........ r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-05chan_iax2: Fix a crash that occurs when using allow=all for an IAX2 peerJonathan Rose
Or any combination of codecs that includes Opus. ASTERISK-24107 #close Review: https://reviewboard.asterisk.org/r/3885/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-04Remove duplicate definitions of ast_format_vp8.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-04Manager - Improve documentation for manager commands Getvar and Setvar.Rusty Newton
The documentation for these commands did not make it clear that they could accept expressions and functions. Modified to make this clear, but tried not to be overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3854 ........ Merged revisions 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419943 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419944 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-31res_hep_rtcp: Add module that sends RTCP information to a Homer ServerMatthew Jordan
This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes to the RTCP topics in Stasis and receives RTCP information back from the message bus. It encodes into HEPv3 packets and sends the information to the res_hep module for transmission. Using this, someone with a Homer server can get live call quality monitoring for all RTP-based channels in their Asterisk 12+ systems. In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered by the tests written for the Asterisk Test Suite. This patch fixes the following: 1) chan_pjsip failed to set its channel unique ids on its RTP instance on outbound calls. It now does this in the appropriate location, in the serialized call callback. 2) The rtp_engine was overflowing some values when packed into JSON. Specifically, some longs and unsigned ints can't be be packed into integer values, for obvious reasons. Since libjansson only supports integers, floats, strings, booleans, and objects, we print these values into strings. 3) res_rtp_asterisk had a few problems: (a) it would emit a source IP address of 0.0.0.0 if bound to that IP address. We now use ast_find_ourip to get a better IP address, and properly marshal the result into an ast_strdupa'd string. (b) Reports can be generated with no report bodies. In particular, this occurs when a sender is transmitting information to a receiver (who will send no RTP back to the sender). As such, the sender has no report body for what it received. We now properly handle this case, and the sender will emit SR reports with no body. Likewise, if we receive an RTCP packet with no report body, we will still generate the appropriate events. ASTERISK-24119 #close ........ Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-31xmldocs: Add support for an <example> tag in the Asterisk XML DocumentationMatthew Jordan
This patch adds support for an <example /> tag in the XML documentation schema. For CLI help, this doesn't change the formatting too much: - Preceeding white space is removed - Unlike with para elements, new lines are preserved However, having an <example /> tag in the XML schema allows for the wiki documentation generation script to surround the documentation with {code} or {noformat} tags, generating much better content for the wiki - and allowing us to put dialplan examples (and other code snippets, if desired) into the documentation for an application/function/AMI command/etc. Review: https://reviewboard.asterisk.org/r/3807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-30manager: Add state list commandsKinsey Moore
This patch adds three new AMI commands: * ExtensionStateList (pbx.c) - list all known extension state hints and their current statuses. Events emitted by the list action are equivalent to the ExtensionStatus events. * PresenceStateList (res_manager_presencestate) - list all known presence state values. Events emitted are generated by the stasis message type, and hence are PresenceStateChange events. * DeviceStateList (res_manager_devicestate) - list all known device state values. Events emitted are generated by the stasis message type, and hence are DeviceStateChange events. Patch-by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3799/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-29Do not omit the first header of a UserEvent AMI action from the ↵Mark Michelson
corresponding emitted UserEvent. ASTERISK-24124 #close Reported by Matt Jordan AFS-131 #close Reported by Matt Jordan Patches: userevent.patch uploaded by Matt Jordan (License #6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-28datastores: Audit ast_channel_datastore_remove usage.Richard Mudgett
Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in app_speech_utils and func_frame_trace. * Fixed app_speech_utils not locking the channel when accessing the channel datastore list. Review: https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leak in func_jitterbuffer. (Was not in v12) Review: https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in abstract_jb. * Fixed leak in ast_channel_unsuppress(). Used by ARI mute control and res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used by ARI mute control and res_mutestream. Review: https://reviewboard.asterisk.org/r/3861/ ........ Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419685 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419686 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25loader: Fix an infinite loop when printing modules using "module show".Joshua Colp
When creating the alphabetical sorted list each module is added to a list temporarily. On the second iteration each module already has a pointer to another module, causing stuff to go into a loop. ASTERISK-24123 #close Reported by: Malcolm Davenport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25Add module support level to ast_module_info structure. Print it in CLI ↵Mark Michelson
"module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25module loader: Unload modules in reverse order of their start orderMatthew Jordan
When Asterisk starts a module (calling its load_module function), it re-orders the module list, sorting it alphabetically. Ostensibly, this was done so that the output of 'module show' listed modules in alphabetic order. This had the unfortunate side effect of making modules with complex usage patterns unloadable. A module that has a large number of modules that depend on it is typically abandoned during the unloading process. This results in its memory not being reclaimed during exit. Generally, this isn't harmful - when the process is destroyed, the operating system will reclaim all memory allocated by the process. Prior to Asterisk 12, we also didn't have many modules with complex dependencies. However, with the advent of ARI and PJSIP, this can make make unloading those modules successfully nearly impossible, and thus tracking memory leaks or ref debug leaks a real pain. While this patch is not a complete overhaul of the module loader - such an effort would be beyond the scope of what could be done for Asterisk 13 - this does make some marginal improvements to the loader such that modules like res_pjsip or res_stasis *may* be made properly un-loadable in the future. 1. The linked list of modules has been replaced with a doubly linked list. This allows traversal of the module list to occur backwards. The module shutdown routine now walks the global list backwards when it attempts to unload modules. 2. The alphabetic reorganization of the module list on startup has been removed. Instead, a started module is placed at the end of the module list. 3. The ast_update_module_list function - which is used by the CLI to display the modules - now does the sorting alphabetically itself. It creates its own linked list and inserts the modules into it in alphabetic order. This allows for the intent of the previous code to be maintained. This patch also contains a fix for res_calendar. Without calendar.conf, the calendar modules were improperly bumping the use count of res_calendar, then failing to load themselves. This patch makes it so that we detect whether or not calendaring is enabled before altering the use count. Review: https://reviewboard.asterisk.org/r/3777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25bridge: Make "bridge destroy" only available in developer mode and add "all" ↵Joshua Colp
to "bridge kick". The "bridge destroy" CLI command is invasive to bridges and can leave them in an unexpected state for the users of them. Since this command may be useful for developers it is now only available when developer mode is available. To take its place "all" has been added as a valid option to the "bridge kick" CLI command. It will kick all of the channels in the bridge out. ASTERISK-23987 Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/ ........ Merged revisions 419536 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-24accountcode: Slightly change accountcode propagation.Richard Mudgett
The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-24core/db: Revert Patch Added In Attempt To Improve I/O PerformanceMichael L. Young
Reverting the patch since it was causing a regression and after fixing the regression, there were no performance gains. At least based on my method for measurement. ASTERISK-24050 Review: https://reviewboard.asterisk.org/r/3841/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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