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2014-03-05stasis: Made internal_stasis_subscribe() prototype and definition match exactly.Richard Mudgett
........ Merged revisions 409682 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26PJSIP: Fix some bad spacingKinsey Moore
........ Merged revisions 408943 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-25res_pjsip_exten_state: Presence for digium phonesKevin Harwell
Added presence support for digium phones. Review: https://reviewboard.asterisk.org/r/3239/ ........ Merged revisions 408882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21json: Fix off-nominal json ref counting issues.Richard Mudgett
* Fixed off-nominal json ref counting issue with using the following API calls: ast_json_object_set() and ast_json_array_append(). * Fixed off-nominal error reporting in ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal json ref counting issues in report_receive_fax_status() and dial_to_json(). ........ Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20sorcery: Create sorcery instance registry.George Joseph
In order to retrieve an arbitrary sorcery instance from a dialplan function (or any place else) there needs to be a registry of sorcery instances. ast_sorcery_init now creates a hashtab as a registry. ast_sorcery_open now checks the hashtab for an existing sorcery instance matching the caller's module name. If it finds one, it bumps the refcount and returns it. If not, it creates a new sorcery instance, adds it to the hashtab, then returns it. ast_sorcery_retrieve_by_module_name is a new function that does a hashtab lookup by module name. It can be called by the future dialplan function. res_pjsip/config_system needed a small change to share the main res_pjsip sorcery instance. tests/test_sorcery was updated to include a test for the registry. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions 408518 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-17Store SIP User-Agent information in contacts.Mark Michelson
When an endpoint sends a REGISTER request to Asterisk, we now will associate the User-Agent header with all contacts that were bound in that REGISTER request. ........ Merged revisions 408270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07timing: Improve performance for most timing implementations.Joshua Colp
This change allows timing implementation data to be stored directly on the timer itself thus removing the requirement for many implementations to do a container lookup for the same information. This means that API calls into timing implementations can directly access the information they need instead of having to find it. Review: https://reviewboard.asterisk.org/r/3175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07chan_iax2: Block unnecessary control frames to/from the wire.Richard Mudgett
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ ........ Merged revisions 407678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407727 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06res_pjsip: Updates and adds more PJSIP CLI commands.Richard Mudgett
* Adds identify, transport, and registration support to the PJSIP CLI. * Creates three additional callbacks, one for an iterator, one for a comparator, and one for a container. This eliminates the link dependency from higher level modules to lower level ones. * Eliminates duplicate sorting in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * Pushes CLI command registration down to the implementing source file. * Adds several ast_sip_destroy_sorcery functions to complement existing ast_sip_sorcery_initialize functions. The destroy functions unregister PJSIP CLI commands and PJSIP CLI formatters. Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3104/ ........ Merged revisions 407568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-05ARI/AMI: Update versions; update UPGRADE/CHANGES notes for 12.1.0 changesMatthew Jordan
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
2014-02-04devicestate: Make ast_devstate_changed_literal() return value and doxygen ↵Richard Mudgett
consistent. Nothing actually cares about the value anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose ........ Merged revisions 407337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407338 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407339 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31res_pjsip: Config option to enable PJSIP logger at load time.Kevin Harwell
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
2014-01-31Add file that apparently got missed in the merge.Mark Michelson
........ Merged revisions 407031 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31Decouple subscription handling from NOTIFY/PUBLISH body generation.Mark Michelson
When the PJSIP pubsub framework was created, subscription handlers were required to state what event they handled along with what body types they knew how to generate. While this serves well when implementing a base RFC, it has problems when trying to extend the body to support non-standard or proprietary body elements. The code also was NOTIFY-specific, meaning that when the time comes that we start writing code to send out PUBLISH requests with MWI or presence bodies, we would likely find ourselves duplicating code that had previously been written. This changeset introduces the concept of body generators and body supplements. A body generator is responsible for allocating a native structure for a given body type, providing the primary body content, converting the native structure to a string, and deallocating resources. A body supplement takes the primary body content (the native structure, not a string) generated by the body generator and adds nonstandard elements to the body. With these elements living in their own module, it becomes easy to extend our support for body types and to re-use resources when sending a PUBLISH request. Body generators and body supplements register themselves with the pubsub core, similar to how subscription and publish handlers had done. Now, subscription handlers do not need to know what type of body content they generate, but they still need to inform the pubsub core about what the default body type for a given event package is. The pubsub core keeps track of what body generators and body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core will check that there is a subscription handler for the event in the SUBSCRIBE, then it will check that there is a body generator that can provide the content specified in the Accept header(s). Because of the nature of body generators and supplements, it means res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no longer worry about body types, instead calling ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150 ........ Merged revisions 407016 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28cdr_radius, cel_radius: build agains libfreeradius-clientKevin Harwell
Asterisk's RADIUS module currently build against libradiusclient-ng, but this project has been superseeded by libfreeradius-client. The API is 99% compatible except that the header name has changed, the library name has changed, and the configuration file location has changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé Patches: freeradius-client.patch uploaded by sharky (license 6561) ........ Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406802 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406803 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28res_pjsip,compat: INFINITY and NAN undefinedKevin Harwell
On some systems the values for INFINITY and NAN are not defined thus causing a build error on those systems. Added definitions for those if they had not previously been defined. (closes issue ASTERISK-23056) Reported by: capouch Patches: inf-nan-patch.txt uploaded by capouch (license 6564) ........ Merged revisions 406788 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-27Protect ast_filestream object when on a channelRussell Bryant
The ast_filestream object gets tacked on to a channel via chan->timingdata. It's a reference counted object, but the reference count isn't used when putting it on a channel. It's theoretically possible for another thread to interfere with the channel while it's unlocked and cause the filestream to get destroyed. Use the astobj2 reference count to make sure that as long as this code path is holding on the ast_filestream and passing it into the file.c playback code, that it knows it's valid. Bug reported by Leif Madsen. Review: https://reviewboard.asterisk.org/r/3135/ ........ Merged revisions 406566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-17pjsip: fix support for allow=allScott Griepentrog
This change adds improvements to support for allow=all in pjsip.conf so that it functions as intended. Previously, the allow/disallow socery configuration would set & clear codecs from the media.codecs and media.prefs list, but if all was specified the prefs list was not updated. Then a call would fail when create_outgoing_sdp_stream() created an SDP with no audio codecs. A new function ast_codec_pref_append_all() is provided to add all codecs to the prefs list - only those not already on the list. This enables the configuration to specify a codec preference, but still add all codecs, and even then remove some codecs, as shown in this example: allow = ulaw, alaw, all, !g729, !g723 Also, the display order of allow in cli output is updated to match the configuration by using prefs instead of caps when generating a human readable string. Finally, a change to create_outgoing_sdp_stream() skips a codec when it does not have a payload code instead of the call failing. (closes issue ASTERISK-23018) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3131/ ........ Merged revisions 405875 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-15PJSIP: Add Path header supportKinsey Moore
This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14ARI: Add mailboxes resource for controlling and polling external MWIJonathan Rose
Adds the following AMI commands: PUT mailboxes/mailboxName modifies mailbox state and implicitly creates new mailboxes GET mailboxes/mailboxName retrieves a JSON representation of a single mailbox if it exists GET mailboxes retrieves a JSON array of all mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that res_mwi_external must be loaded for these functions to actually do anything. Review: https://reviewboard.asterisk.org/r/3117/ ........ Merged revisions 405553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14verbosity: Fix performance of console verbose messages.Richard Mudgett
The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-12res_ari: Fix various memory leaks.Joshua Colp
This change fixes a few memory leaks that were found based on a mailing list post. 1. Some JSON response messages were never freed. This was caused by the documentation stating that message references were stolen when in reality they were not. The code now follows the documentation and usage has been updated. 2. HTTP response headers were never freed. 3. The variable list for wildcards paths was never freed. (closes issue ASTERISK-23128) Reported by: Kenneth Watson (on list) Review: https://reviewboard.asterisk.org/r/3119/ ........ Merged revisions 405325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-12CDRs: Synchronize dialplan applications that manipulate CDRs with the engineMatthew Jordan
In https://reviewboard.asterisk.org/r/3057/, applications and functions that manipulate CDRs were made to interact over Stasis. This was done to synchronize manipulations of CDRs from the dialplan with the updates the engine itself receives over the message bus. This change rested on a faulty premise: that messages published to the CDR topic or to a topic that forwards to the CDR topic are synchronized with the messages handled by the CDR topic subscription in the CDR engine. This is not the case. There is no ordering guaranteed for two messages published to the same topic; ordering is only guaranteed if a message is published to the same subscriber. Stasis was modified in r405311 to allow a publisher to synchronize on the subscriber. This patch uses that API to synchronize the CDR publishers with the CDR engine message router, which maintains the overall topic subscription. (closes issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-12stasis: Add methods to allow for synchronous publishing to subscriberMatthew Jordan
This patch adds an API call to Stasis that allows a publisher to publish a stasis message that will not return until a specific subscriber handles the message. Since a subscriber can have their own forwarding topic which orders messages from many topics, this allows a publisher who knows of that subscriber to synchronize to that subscriber regardless of the forwarding relationships between topics. This is of particular use for dialplan applications that need to synchronize on a particular subscriber's handling of a message. (issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-09astobj2: Correct ao2_iterator opacity violationsKinsey Moore
This corrects the ao2_iterator opacity violations in res_pjsip_session.c by adding a global function to get the number of elements inside the container hidden behind the iterator. (closes issue ASTERISK-23053) Review: https://reviewboard.asterisk.org/r/3111/ Reported by: Richard Mudgett ........ Merged revisions 405253 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-06External MWI core support.Richard Mudgett
* The core external MWI resource provides for MWI message counts persistence using sorcery. With sorcery, the user is able to configure which sorcery wizzard backend to use if the default astdb is not desired. * The core external MWI resoruce provides some debugging CLI commands enabled by defining MWI_DEBUG_CLI. The debugging CLI commands are: "mwi delete all", "mwi delete like <regex>", "mwi delete mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> [<old>]]". (closes issue AFS-43) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404952 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-02res_pjsip: add 'set_var' support on endpointsKevin Harwell
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
2013-12-20res_pjsip: Add PJSIP CLI commandsMatthew Jordan
Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-20manager: bump version to 2.0.0Matthew Jordan
AMI has received substantial updates over the past year. Not only has the syntax been vastly improved and made consistent (which entails many event changes), but the underlying things that those events convey have changed substantially as well. After some conversation in #asterisk-dev, it was agreed that this is a good time to jump to 2. At the same time, since ARI will most likely use semantic versioning, we might as well use that for AMI as well. That also affords us greater meaning for the AMI version. ........ Merged revisions 404421 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-19Fix a deadlock that occurred due to a conflict of masquerades.Mark Michelson
For the explanation, here is a copy-paste of the review board explanation: Initially, it was discovered that performing an attended transfer of a multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread started a masquerade and reached the point where it was calling the fixup() callback on the "original" channel. For chan_pjsip, this involves pushing a synchronous task to the session's serializer. The problem was that a task ahead of the fixup task was also attempting to perform a channel masquerade. However, since masquerades are designed in a way to only allow for one to occur at a time, the task ahead of the fixup could not continue until the masquerade already in progress had completed. And of course, the masquerade in progress could not complete until the task ahead of the fixup task had completed. Deadlock. The initial fix was to change the fixup task to be asynchronous. While this prevented the deadlock from occurring, it had the frightful side effect of potentially allowing for tasks in the session's serializer to operate on a zombie channel. Taking a step back from this particular deadlock, it became clear that the problem was not really this one particular issue but that masquerades themselves needed to be addressed. A PJSIP attended transfer operation calls ast_channel_move(), which attempts to both set up and execute a masquerade. The problem was that after it had set up the masquerade, the PBX thread had swooped in and tried to actually perform the masquerade. Looking at changes that had been made to Asterisk 12, it became clear that there never is any time now that anyone ever wants to set up a masquerade and allow for the channel thread to actually perform the masquerade. Everyone always is calling ast_channel_move(), performs the masquerade itself before returning. In this patch, I have removed all blocks of code from channel.c that will attempt to perform a masquerade if ast_channel_masq() returns true. Now, there is no distinction between setting up a masquerade and performing the masquerade. It is one operation. The only remaining checks for ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not want to interrupt a masquerade by hanging up the channel. Instead, now ast_hangup() will wait for a masquerade to complete before moving forward with its operation. The ast_channel_move() function has been modified to basically in-line the logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has been killed off for real. ast_channel_move() now has a lock associated with it that is used to prevent any simultaneous moves from occurring at once. This means there is no need to make sure that ast_channel_masq() or ast_channel_masqr() are already set on a channel when ast_channel_move() is called. It also means the channel container lock is not pulling double duty by both keeping the container locked and preventing multiple masquerades from occurring simultaneously. The ast_do_masquerade() function has been renamed to do_channel_masquerade() and is now internal to channel.c. The function now takes explicit arguments of which channels are involved in the masquerade instead of a single channel. While it probably is possible to do some further refactoring of this method, I feel that I would be treading dangerously. Instead, all I did was change some comments that no longer are true after this changeset. The other more minor change introduced in this patch is to res_pjsip.c to make ast_sip_push_task_synchronous() run the task in-place if we are already a SIP servant thread. This is related to this patch because even when we isolate the channel masquerade to only running in the SIP servant thread, we would still deadlock when the fixup() callback is reached since we would essentially be waiting forever for ourselves to finish before actually running the fixup. This makes it so the fixup is run without having to push a task into a serializer at all. (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: https://reviewboard.asterisk.org/r/3069 ........ Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-19udptl: Dead code elimination. ast_udptl_bridge was not used.Richard Mudgett
Removing dead code starting with ast_udptl_bridge() eliminated the code in this change. Note: This code has actually been dead since Asterisk v1.4 when it was first put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-19Voicemail: Remove mailbox identifier format (box@context) assumptions in the ↵Richard Mudgett
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
2013-12-19app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating stateMatthew Jordan
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
2013-12-18channel locking: Add locking for channel snapshot creationKevin Harwell
Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-18channels: Return allocated channels locked.Joshua Colp
This change makes ast_channel_alloc return allocated channels locked. By doing so no other thread can acquire, lock, and manipulate the channel before it is completely set up. (closes issue AST-1256) Review: https://reviewboard.asterisk.org/r/3067/ ........ Merged revisions 404204 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-17Several components: fixing Typos in comments and code, "avaliable" instead ↵Rusty Newton
of "available" (issue ASTERISK-23021) (closes issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty Newton Patches: available.patch uploaded by Jeremy Lainé (license 6561) ........ Merged revisions 404046 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-17bridging: Give bridges a name and a known creatorJonathan Rose
Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ ........ Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-16security: Inhibit execution of privilege escalating functionsDavid M. Lee
This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-14res_pjsip: Apply outbound proxy to all SIP requests.Joshua Colp
Objects which are involved in SIP request creation and sending now allow an outbound proxy to be specified. For cases where an endpoint is used the outbound proxy specified there will be applied. (closes issue ASTERISK-22673) Reported by: Antti Yrjola Review: https://reviewboard.asterisk.org/r/3022/ ........ Merged revisions 403811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-14res_stasis: Expose event for call forwarding and follow forwarded channel.Joshua Colp
This change adds an event for when an originated call is redirected to another target. This event contains the original channel and the newly created channel. If a stasis subscription exists on the original originated channel for a stasis application then a new subscription will also be created on the stasis application to the redirected channel. This allows the application to follow the call path completely. (closes issue ASTERISK-22719) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3054/ ........ Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliablyJonathan Rose
There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be set on channels involved with blind and attended transfers. This would happen with features that were initialized by channel driver specific mechanisms in multiparty calls. This patch resolves those cases while attempted to keep the behavior for setting those variables as consistent as possible. (closes issue AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........ Merged revisions 403781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13bridge_native_rtp: Deadlock during 4-way conference creationKevin Harwell
The change contains a slightly adjusted patch that was on the issue (submitted by kmoore). A fix was made by adding in a bridge lock while calling bridge_start/stop from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. (closes issue ASTERISK-22749) Reported by: Kinsey Moore Review: https://reviewboard.asterisk.org/r/3066/ Patches: lock_inversion.diff uploaded by kmoore (license 6273) ........ Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13ARI: Adding a channel to a bridge while a live recording is active blocksKevin Harwell
Added the ability to have rules that are checked when adding and/or removing channels to/from a bridge. In this case, if a channel is currently recording and someone attempts to add it to a bridge an "is recording" rule is checked, fails, and a 409 conflict is returned. Also command functions now return an integer value that can be descriptive of what kind of problems, if any, occurred before or during execution. (closes issue ASTERISK-22624) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2947/ ........ Merged revisions 403749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-13test_voicemail_api: Add check for a registered voicemail provider before tests.Richard Mudgett
It is much nicer diagnosing a test failure if app_voicemail is actually loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11res_pjsip_messaging: send message to a default outbound endpointKevin Harwell
In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11app_voicemail: Voicemail callback registration/unregistration function ↵Richard Mudgett
improvements. * The voicemail registration/unregistration functions now take a struct of callbacks instead of a lengthy parameter list of callbacks. * The voicemail registration/unregistration functions now prevent a competing module from interfering with an already registered callback supplying module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-11func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsipMatthew Jordan
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan to use the CHANNEL function on a chan_pjsip channel to obtain run-time information about the channel from the PJSIP channel driver and the PJSIP stack. This includes: * RTP information, including source/destination media addresses, whether or not the media is secure, held, and other properties. * RTCP information. This includes sets of parseable information, as well as individual statistic attriutes. * PJSIP information. This includes URIs, local/remote signalling addresses, whether or not the signalling is secure, and other properties. * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT function to obtain more detailed endpoint information. Review: https://reviewboard.asterisk.org/r/3038/ ........ Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09app_page: Add predial handlers for app_page.Jonathan Rose
(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
2013-12-09Switch PJSIP auth to use a vector.Mark Michelson
Since Asterisk has a vector API now, places where arrays are manually resized don't really make sense any more. Since the auth work in PJSIP was freshly-written, it was easy to reform it to use a vector. Review: https://reviewboard.asterisk.org/r/3044 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05Reverting r403311. It's causing ARI tests to hang.David M. Lee
........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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