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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
2013-12-11func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint detailsMatthew Jordan
This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint, any property configured on an endpoint. This function is a companion to the CHANNEL function, which can be used to extract the endpoint name for a channel. Review: https://reviewboard.asterisk.org/r/3035 ........ Merged revisions 403616 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-10Fix correct authentication behavior for artificial endpoint.Mark Michelson
When switching to using a vector for authentication, I initialized the vector for the artificial endpoint to be of size 1. However, this does not result in AST_VECTOR_SIZE() returning 1 since there isn't actually anything in the vector. Rather than trifle with the vector by putting unnecessary elements in, I simply changed the callback in res_pjsip_authenticator_digest.c to explicitly report that the artificial endpoint requires authentication. Thanks to Joshua Colp for pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09chan_pjsip: Fix a sticking channel lock caused by channel masqueradesJonathan Rose
(closes issue ASTERISK-22936) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/3042/ ........ Merged revisions 403587 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403588 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
2013-12-09Reverting regex part of -r403545 at request of file.Richard Mudgett
res_sorcery_astdb.c: Fix get multiple records by regex. * Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec() function match the stored key values instead of having astdb prefilter them. Previoiusly you could only use a simple regex pattern when the pattern began with '^'. ........ Merged revisions 403559 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09res_sorcery_astdb.c: Fix get multiple records by regex.Richard Mudgett
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec() function match the stored key values instead of having astdb prefilter them. Previoiusly you could only use a simple regex pattern when the pattern began with '^'. * Fix off nominal memory leak in sorcery_astdb_retrieve_regex(). ........ Merged revisions 403545 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09sorcery: Eliminate shadowing a varaible that caused confusion.Richard Mudgett
* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter causing confusion. * Fix potential crash from sorcery.conf user input in __ast_sorcery_apply_config() if the user supplied a malformed config line that is missing the sorcery object type name. * Remove redundant test in __ast_sorcery_apply_config(). !config and config == CONFIGS_STATUS_FILEMISSING are identical. ........ Merged revisions 403541 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09endpoints: Keep a reference to channel ids when creating snapshot.Joshua Colp
The snapshot process for endpoints uses the channel ids present on the endpoint itself. Without keeping a reference it was possible for the strings to be freed underneath any consumer of an endpoint snapshot. A reference is now held by the snapshot to the channel ids and released when the snapshot is destroyed. (issue ASTERISK-22801) Reported by: Matt Jordan ........ Merged revisions 403542 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09sorcery: WhitespaceRichard Mudgett
You would think that a new file would start off without any whitespace oddities. ........ Merged revisions 403527 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09Add a CONFBRIDGE_RESULT channel variable to discern why a channel left a ↵Mark Michelson
ConfBridge. Review: https://reviewboard.asterisk.org/r/3009 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09Create function for retrieving Mixmonitor instance data.Mark Michelson
For the time, this is only useful for retrieving the filename. The purpose of this function is to better facilitate multiple mixmonitors per channel. Setting a MIXMONITOR_FILENAME channel variable is not conducive to such behavior, so allowing finer grained access to individual mixmonitor properties improves the situation. The MIXMONITOR_FILENAME channel variable is still set, though, so there is no worry about backwards compatibility. Review: https://reviewboard.asterisk.org/r/3023 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-09res_pjsip_nat: Add NAT module to session dialogs.Joshua Colp
Due to the way pjproject internally works it was possible for the NAT module to not be invoked on messages with-in a session dialog. This means that the various parts of the message would not get rewritten with the source IP address and port. This change uses a session supplement to add the NAT module to the dialog on the first incoming or outgoing INVITE. (closes issue ASTERISK-22941) Reported by: Leif Madsen ........ Merged revisions 403510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403511 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
2013-12-09res_fax_spandsp: Always init T.38 session to avoid crashes during state changeMatthew Jordan
Prior to this patch, res_fax_spandsp was conservative with how it initialized the spandsp T.38 context. It would only initialize it if the driver thought the current state was a T.38 fax. While this works fine in nominal situations, in certain off nominal situations, res_fax_spandsp can believe that a T.38 fax will not occur when in fact one has started. In particular, this was discovered when res_fax would fall back to audio after timing out on a T.38 upgrade. The SIP channel driver would continue to retry the re-INVITE and - if the remote end responded after res_fax timed out with a 200 OK - a T.38 frame would be delivered to the res_fax stack when it no longer expected it. As it turns out, there does not appear to be any downside to always initializing the T.38 context, other than the actual memory allocation. Since that avoids this off nominal situation (and others which are equally likely hard to predict), this is the safest way to avoid this problem. Much thanks to Torrey as well for providing a scenario that reproduces this issue. (closes issue ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey Searle patches: always-init-t38.patch uploaded by awinters (License 6477) A_PARTY.xml uploaded by tsearle (License 5334) ........ Merged revisions 403449 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403450 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-08res_config_sqlite: Check for CDR unregistration failuresMatthew Jordan
If the CDR unregistration fails due to an inflight CDR, the res_config_sqlite module needs to bail on unloading itself. Otherwise, the config could be unloaded (including the CDR table name) while the CDR engine posts a CDR to the still registered backend, resulting in a crash. ........ Merged revisions 403435 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05app_record: Add an option that allows DTMF '0' to act as an additional ↵Jonathan Rose
terminator Using this terminator when active results in ${RECORD_STATUS} being set to 'OPERATOR' instead of 'DTMF' (closes issue AFS-7) Review: https://reviewboard.asterisk.org/r/3041/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403414 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
2013-12-05ari: Fix deadlock problem with functions that use autoservice.David M. Lee
The code for getting channel variables from ARI assumed that you needed to lock the channel in order to properly execute functions and read channel variables. Apparently, this is not the case, since any dialplan function that puts the channel into autoservice deadlocks when attempting to remove the channel from autoservice. ........ Merged revisions 403342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05Multiple revisions 403304,403310David M. Lee
........ r403304 | dlee | 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line Fixed the filename for the ari.conf docs ........ r403310 | file | 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines Revert revision 403304: Fixed the filename for the ari.conf docs The changed value refers to the name of the module. The name of the configuration file is specified in the configFile section. ........ Merged revisions 403304,403310 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403402 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05Blocked revisions 403291David M. Lee
........ remove unwanted property svn:mergeinfo git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-04res_pjsip_registrar: undefined function pointer symbolKevin Harwell
Used a static wrapper around the offending function to alleviate the issue. Reported by: rmudgett ........ Merged revisions 403377 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-04res_pjsip_t38: Don't pass T.38 control frames through to other hooks.Joshua Colp
This crept up during gateway testing where the gateway would receive the request to negotiate and assume it came from the remote side, causing the gateway state machine to go a little, to a use a technical term, "wonky". ........ Merged revisions 403364 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-04Initialize the hash value argument to pj_hash_get() to 0.Mark Michelson
Passing a non-zero value causes PJLIB to use the given input as the hash value. Passing zero causes the parameter to become an output parameter that receives the hash value that was computed based on the given key. This change essentially makes ast_sip_dict_get() properly retrieve the desired value. ........ Merged revisions 403349 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.Joshua Colp
Newer versions of PJSIP have changed to using a flag for the PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a configure check to detect the presence of the flag and use it if found. ........ Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03sorcery, bucket: Change observer remove calls to take const callbacks struct.Richard Mudgett
* Make ast_sorcery_observer_remove() accept a const callbacks struct. * Make ast_sorcery_observer_remove() tolerant of the sorcery parameter being NULL. Now it can be called within a module unload routine if the sorcery initialization fails. * Fix ast_sorcery_observer_add() to fail if the container link fails. ........ Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03Add channel locking for channel snapshot creation.Mark Michelson
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. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03media_index: Make media indexing tolerable of bad symlinks.Joshua Colp
Media indexing will now skip over files and directories that stat will not return information about. This can occur under normal conditions when a symbolic link points to a location that no longer exists. ........ Merged revisions 403312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-02Check and reject non-digits e164 values on peers and general sections in ↵Alexandr Anikin
ooh323.conf Regenerate e164 endpoint list on reload ooh323 (issue ASTERISK-22901) Reported by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch ........ Merged revisions 403288 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403290 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-01res_pjsip_session: Apply fromuser and fromdomain to all requests as documented.Joshua Colp
........ Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-01res_pjsip_t38: Add the framehook to the channel only on first INVITE.Joshua Colp
The check for determining whether the T.38 framehook should be added to the channel or not has now been changed to guarantee adding only occurs on the first incoming or outgoing INVITE. ........ Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-01res_pjsip_transport_websocket: Fix security events and simplify implementation.Joshua Colp
Transport type determination for security events has been simplified to use the type present on the message itself instead of searching through configured transports to find the transport used. The actual WebSocket transport has also been simplified. It now leverages the existing PJSIP transport manager for finding the active WebSocket transport for outgoing messages. This removes the need for res_pjsip_transport_websocket to store a mapping itself. (closes issue ASTERISK-22897) Reported by: Max E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ ........ Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-30res_ari: Add Recording events to the validator.Joshua Colp
........ Merged revisions 403240 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_pjsip_sdp_rtp: Don't produce an invalid media stream with no formats.Joshua Colp
Depending on configuration it was possible for a media stream to be created without any media formats. The produced SDP would fail internal validation and cause a crash. The code will now no longer add media streams with no formats to the SDP, allowing it to pass validation and work. (closes issue ASTERISK-22858) Reported by: Anthony Messina ........ Merged revisions 403223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_pjsip_header_funcs: Don't add headers to re-INVITEs.Joshua Colp
When sending a re-INVITE to an endpoint it was possible for received headers to be added as well (since they are stored for retrieval using the PJSIP_HEADER dialplan function). This caused a broken (and potentially large) SIP INVITE to be produced and sent. This changes the module so it will no longer add headers to re-INVITEs. (closes issue ASTERISK-22882) Reported by: David M. Lee ........ Merged revisions 403221 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_stasis_playback: Add 'number', 'digits', and 'characters' URI scheme ↵Joshua Colp
implementations. This change adds new URI scheme implementations for playing numbers, digits, and characters. This is done as part of the normal playback mechanism and can be used with queueing to create a combined sentence. Review: https://reviewboard.asterisk.org/r/3028/ ........ Merged revisions 403209 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-28res_pjsip_session: Add configurable behavior for redirects.Joshua Colp
The action taken when a redirect occurs is now configurable on a per-endpoint basis. The redirect can either be treated as a redirect to a local extension, to a URI that is dialed through the Asterisk core, or to a URI that is dialed within PJSIP itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27astdb: Tweak some doxygen comments.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27res_pjsip: Fix crash when reloading certain configurations.Joshua Colp
Certain options available that specify a SIP URI perform validation on the provided URI using the PJSIP URI parser. This operation requires that the thread executing it be registered with the PJLIB library. During reloads this was done on a thread which was NOT registered with it. This fixes the problem by creating a task which reloads the configuration on a PJSIP thread. (closes issue ASTERISK-22923) Reported by: Anthony Messina ........ Merged revisions 403179 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27ari:Add application/json parameter supportDavid M. Lee
The patch allows ARI to parse request parameters from an incoming JSON request body, instead of requiring the request to come in as query parameters (which is just weird for POST and DELETE) or form parameters (which is okay, but a bit asymmetric given that all of our responses are JSON). For any operation that does _not_ have a parameter defined of type body (i.e. "paramType": "body" in the API declaration), if a request provides a request body with a Content type of "application/json", the provided JSON document is parsed and searched for parameters. The expected fields in the provided JSON document should match the query parameters defined for the operation. If the parameter has 'allowMultiple' set, then the field in the JSON document may optionally be an array of values. (closes issue ASTERISK-22685) Review: https://reviewboard.asterisk.org/r/2994/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-27res_pjsip: Update handling of some options to work with new option names.Joshua Colp
Some options (such as call_group and pickup_group) share the same configuration handler and decide what logic to use based on the name of the option. These handlers were not updated to check for the new option names and were treating the options as invalid. This change simply updates the handlers with the proper names of the options. (closes issue ASTERISK-22922) Reported by: Anthony Messina ........ Merged revisions 403173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-26Fix a configure issue with PJSIP transaction group lock detection.Joshua Colp
The configure check did not use the provided paths for pjproject if provided when looking for transaction group lock support. ........ Merged revisions 403160 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23ARI: Implement device state APIKevin Harwell
Created a data model and implemented functionality for an ARI device state resource. The following operations have been added that allow a user to manipulate an ARI controlled device: Create/Change the state of an ARI controlled device PUT /deviceStates/{deviceName}&{deviceState} Retrieve all ARI controlled devices GET /deviceStates Retrieve the current state of a device GET /deviceStates/{deviceName} Destroy a device-state controlled by ARI DELETE /deviceStates/{deviceName} The ARI controlled device must begin with 'Stasis:'. An example controlled device name would be Stasis:Example. A 'DeviceStateChanged' event has also been added so that an application can subscribe and receive device change events. Any device state, ARI controlled or not, can be subscribed to. While adding the event, the underlying subscription control mechanism was refactored so that all current and future resource subscriptions would be the same. Each event resource must now register itself in order to be able to properly handle [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged revisions 403134 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23res_pjsip: AMI commands and events.Kevin Harwell
Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23ari: Add events for playback and recording.Joshua Colp
While there were events defined for playback and recording these were not actually sent. This change implements the to_json handlers which produces them. (closes issue ASTERISK-22710) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/3026/ ........ Merged revisions 403119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23ari: Add Snoop operation for spying/whispering on channels.Joshua Colp
The Snoop operation can be invoked on a channel to spy or whisper on it. It returns a channel that any channel operations can then be invoked on (such as record to do monitoring). (closes issue ASTERISK-22780) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3003/ ........ Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-23app_voicemail: when forwarding a message, play vm-msgforwarded instead of ↵Rusty Newton
vm-msgsaved In the last release of sounds, 1.4.25 we added a vm-msgforwarded prompt for various core languages. Now we use that prompt. (issue ASTERISK-21413) (closes issue ASTERISK-21413) Reported by: netwrkr Tested by: newtonr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-22Make sure unit tests compileKinsey Moore
This fixes the unit tests that were broken by r403069 and several functions requiring a new parameter for sanitization of JSON messages generated from object snapshots. ........ Merged revisions 403094 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-22res_pjsip: convert configuration settings names to snake case some moreKevin Harwell
Updated the alembic script for pjsip. Also, the dtls config parsing stuff was expecting strings with no underscores, so removed the underscores from the option name before passing it to the parser. ........ Merged revisions 403082 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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