asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	44409401ec	main/formats: Fix crash in ast_format_cmp during non-clean shutdown. * Update asterisk.h to reflect availability of ast_register_cleanup in 11.9. * Use ast_register_cleanup for format_attr_shutdown. (closes issue ASTERISK-23103) Reported by: JoshE ........ Merged revisions 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	a8629e53c1	Give sorcery instances a reference to their wizards. On graceful shutdown, sorcery wizards are all killed off, but it is possible for sorcery instances to still have dangling pointers after this, possibly causing a crash. Giving the sorcery instances a reference to their wizards ensures that the wizard reference will remain valid for the lifetime of the sorcery instance. Review: https://reviewboard.asterisk.org/r/3401 ........ Merged revisions 411295 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7dddd694cb	say: Fix a bug where SayNumber in Polish tries to play incorrect sound. This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ Merged revisions 411243 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c8ebf3e3c7	Revert -r411073. It didn't help and blew up the system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	89e12de79d	locking: Add temporary sanity checks. Add some temporary sanity checks to hunt for locking problems with the masquerade supertest. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	df2d959d7d	Remove a LOG_NOTICE from ast_config_engine_register. There is enough indication from the CLI that we are loading a realtime engine as it is. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	1ba13718fc	assigned-uniqueids: Miscellaneous cleanup and fixes. * Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2 uniqueid on a stack variable instead of mallocing it. * Made send error response to ARI and AMI requests instead of just logging excessive uniqueid length and allowing truncation. action_originate() and ari_channels_handle_originate_with_id(). * Fixed minor truncating uniqueid hole when generating the ;2 uniqueid string length. Created public and internal lengths of uniqueid. The internal length can handle a max public uniqueid plus an appended ;2. * free() and ast_free() are NULL tolerant so they don't need a NULL test before calling. * Made use better struct initialization format instead of the position dependent initialization format. Also anything not explicitly initialized in the struct is initialized to zero by the compiler. * Made ast_channel_internal_set_fake_ids() use the safer ast_copy_string() instead of strncpy(). Review: https://reviewboard.asterisk.org/r/3371/ ........ Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	eecb74a9a7	ARI: allow json content type with zero length body When a request was received with a Content-type of json, the body was sent for json parsing - even if it was zero length. This resulted in ARI requests failing that were valid, such as a channel DELETE with no parameters. The code has now been changed to skip json parsing with zero content length. (closes issue SWP-6748) Reported by: Samuel Galarneau Review: https://reviewboard.asterisk.org/r/3360/ ........ Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	77db94a25a	cdr: Add asserts for when we don't know about a CDR for a channel In the CDR core, every channel should either be filtered out (due to being an 'internal' channel used as an implementation detail, such as playing media back into a bridge) or it should get a CDR. Even if that CDR ends up being discarded, we still give the channel a CDR in case we end up needing it. If we hit a situation where a channel does not have a CDR, we should blow up in -dev-mode. Asserts are appropriate for that. This patch adds those asserts, as they would have quickly caught the error fixed by r410814. ........ Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	614b6abc38	stasis_cache: Use the right variable in the cache entry ao2 cmp function. ........ Merged revisions 410813 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410814 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cc40bf5317	res_pjsip: Enable PJSIP DNS client support. This change enables DNS client support within PJSIP. System nameservers are automatically discovered using res_init or res_ninit. If this fails then PJSIP will resort to using gethostbyname for resolution. By enabling this support we gain SRV support, failover, and weight support. (closes issue ASTERISK-23435) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3343/ ........ Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russ Meyerriecks	ed50ef4dc8	callerid: Logic error in checksum processing Callerid checksum-ing was being handled incorrectly here. When the checksum is calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This value will then fail the otherwise correct callerid message. This patch changes the logic to simply add the calculated checksum to the transmitted 2's compliment checksum. Review: https://reviewboard.asterisk.org/r/3356/ (closes issue ASTERISK-23488) ........ This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a broken patch to be comitted to trunk so I pre-merge merged them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	eba91d2a98	Revert changes to sorcery that accidentally got committed. These changes were still up for review and have not been approved yet. I must have had the changes in my working copy when making a different change. ........ Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d44aefeef4	Fix stuck channel in ARI through the introduction of synchronous bridge actions. Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3a565767d7	manager: fix memory leak in manager_add_filter function (closes issue ASTERISK-23420) Reported by: Etienne Lessard Patches: manager_eventfilter_leak uploaded by Etienne Lessard (license 6394) ........ Merged revisions 410609 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410623 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	510a6e6e64	Remove an extra ast_cond_wait() that slipped through the patch. ........ Merged revisions 410606 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410607 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	9665c2d3eb	Handle the return values of realtime updates and stores more accurately. Realtime backends' update and store callbacks return the number of rows affected, or -1 if there was a failure. There were a couple of issues: * The config API was treating 0 as a successful return, and positive values as a failure. Now the config API treats anything >= 0 as a success. * res_sorcery_realtime was treating 0 as a successful return from the store procedure, and any positive values as a failure. Now sorcery treats anything > 0 as a success. It still considers 0 a "failure" since there is no change to report to observers. Review: https://reviewboard.asterisk.org/r/3341 ........ Merged revisions 410592 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ff63012c4e	PJSIP: TOS values should be represented as decimals in sorcery objects (closes issue ASTERISK-23235) Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3324/ ........ Merged revisions 410574 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c1e9d2f177	Prevent delayed astdb syncs. The syncing thread sleeps for a second before waiting to be told to attempt to sync again. If a signal were sent during this sleeping period, we would end up having to wait until the next sync signal occurred in order to sync up the astdb. This code rearrangement also ensures that any pending transactions will be synced prior to Asterisk shutting down. Patches: db_sync.patch by John Hardin (License #6512) ........ Merged revisions 410556 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410559 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	66718a06f7	res_mwi_external: Clear the stasis cache entry when the external MWI is deleted. One of the things missing when external MWI support was added was the ability to clear the stasis cache entry of deleted external MWI mailboxes. Review: https://reviewboard.asterisk.org/r/3325/ ........ Merged revisions 410555 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	251868dc57	cdr.c: Add missing aow_unlock(cdr) in off nominal path of handle_dial_message(). * Trivial common code hoisting in handle_bridge_leave_message(). * Some whitespace fixing. ........ Merged revisions 410541 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f627a0aca0	res_musiconhold.c: Generate MOH start/stop events whenever the MOH stream is started/stopped. * Made res_musiconhold.c always post the MusicOnHoldStart/MusicOnHoldStop events when it actually starts/stops the music streams. This allows the events to always happen when MOH starts/stops. The event posting code was moved to the MOH alloc/release routines. * Made channel_do_masquerade() stop any MOH on the original channel before masquerading so the original channel will get a stop event with correct information. * Cleaned up a couple odd codings in moh_files_alloc() and moh_alloc() dealing with the music state variable. (issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ ........ Merged revisions 410493 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7c854d65af	AST-2014-001: Stack overflow in HTTP processing of Cookie headers. Sending a HTTP request that is handled by Asterisk with a large number of Cookie headers could overflow the stack. Another vulnerability along similar lines is any HTTP request with a ridiculous number of headers in the request could exhaust system memory. (closes issue ASTERISK-23340) Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 410381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	ef69b5176d	unqiueid: correct max uniqueid length test This patch adds null string test prior to checking for a max uniqueid value that was added in r410157. ........ Merged revisions 410368 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	3ff60b75b1	pjsip_cli: Create pjsip show channel and contact, and general cli code cleanup. Created the 'pjsip show channel' and 'pjsip show contact' commands. Refactored out the hated ast_hashtab. Replaced with ao2_container. Cleaned up function naming. Internal only, no public name changes. Cleaned up whitespace and brace formatting in cli code. Changed some NULL checking from "if"s to ast_asserts. Fixed some register/unregister ordering to reduce deadlock potential. Fixed ast_sip_location_add_contact where the 'name' buffer was too short. Fixed some self-assignment issues in res_pjsip_outbound_registration. (closes issue ASTERISK-23276) Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged revisions 410287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b2c4eaf06a	config_options: Display the see-also information for CLI config option help The config option help information has always parsed the <see-also> tags in the XML documentation. Unfortunately, it just never bothered displaying them on the CLI. With this patch, when you execute 'config show help [module] [obj] [option]', it will display what other options are useful to you. (closes issue ASTERISK-22008) Reported by: Richard Mudgett ........ Merged revisions 410209 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c162101d69	Make res_sorcery_realtime filter unknown retrieved results. When retrieving data from a database or other realtime backend, it's quite possible to retrieve variables that Asterisk does not care about but that are legitimate to exist. Asterisk does not need to throw a hissy fit when these variables are encountered but rather just filter them out. Review: https://reviewboard.asterisk.org/r/3305 ........ Merged revisions 410187 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	feae552139	pjsip: allow and disallow show same codecs In order to prevent confusion over the allow and disallow list of codecs being the same an option for registering a field as an alias is added. The alias field will be read from the configuration file, but afterwards is not listed as a known field. With disallow set as an alias, the CLI command pjsip show endpoint # will list the allow= field, but not the disallow field. (closes issue ASTERISK-23092) Review: https://reviewboard.asterisk.org/r/3193/ ........ Merged revisions 410190 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ad1245cb5	stasis cache: Enhance to keep track of an item from different entities. A stasis cache entry now contains more than a single message/snapshot. It contains messages/snapshots for the local entity as well as any remote entities that post to the cached item. In addition callbacks can be supplied when the cache is created to compute and post the aggregate message/snapshot representing all entities stored in the cache entry. * All stasis messages now have an eid to indicate what entity posted it. * The stasis cache enhancements allow device state to cache and aggregate the device states from local and remote entities in a single operation. The cached aggregate device state is available immediately after it is posted to the stasis bus. This improves performance by eliminating a cache dump and associated ao2 container traversals to calculate the aggregate state. (closes issue ASTERISK-23204) Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3281/ ........ Merged revisions 410184 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4fd50a9d81	sorcery.c: Fix off-nominal path ref and memory leak in ast_sorcery_objectset_json_create(). * Made exit a loop early on error in ast_sorcery_objectset_json_create(). * Removed some dead code in ast_sorcery_objectset_create2(). ........ Merged revisions 410089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	a4906e9f86	sorcery: Create AST_SORCERY dialplan function. This patch creates the AST_SORCERY dialplan function which allows someone to retrieve any value from a sorcery-based config file. It's similar to AST_CONFIG. The creation of the function itself was fairly straightforward but it required changes to the underlying sorcery infrastructure that rippled into individual sorcery objects. The changes stemmed from inconsistencies in how sorcery created ast_variable objectsets from sorcery objects and the inconsistency in how individual objects used that feature especially when it came to parameters that can be specified multiple times like contact in aor and match in identify. You can read more here... http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html So, what this patch does, besides actually creating the AST_SORCERY function, is the following... * Creates ast_variable_list_append which is a helper to append one ast_variable list to another. * Modifies the ast_sorcery_object_field_register functions to accept the already-defined sorcery_fields_handler callback. * Modifies ast_sorcery_objectset_create to accept a parameter indicating return type preference...a single ast_variable with all values concatenated or an ast_variable list with multiple entries. Also fixed a few bugs. * Modifies individual sorcery object implementations to use the new function definition of the ast_sorcery_object_field_register functions. * Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement sorcery_fields_handler handlers so they return multiple occurrences as an ast_variable_list. * Added a whole bunch of tests to test_sorcery. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f0b8590c14	pjsip configuration: Make transport TOS values consistent with endpoints Transport TOS values were interpreted as DSCP values without being documented as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS values have historically. This patch makes the transport TOS values behave as TOS values and makes all TOS values readable as string values (e.g. AF11). In addition, alembic scripts have been updated to use the proper field types for all TOS/COS values. (issue ASTERISK-23235) Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3304/ ........ Merged revisions 410028 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	b98c2b0e82	config: Fix inverted test The test of the result of the stat() call was inverted such that its output was only used if the call failed. This inverts the test so that the output of stat() is used correctly. This was causing full reloads on unchanged files. (closes issue ASTERISK-23383) Reported by: David Woolley ........ Merged revisions 409916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409918 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	38a619af97	Corrected cross-platform stat nanosecond code When nanosecond time resolution was added for identifying config file changes, it didn't cover all of the myriad of ways that one might obtain nanosecond time resolution off of struct stat. Rather than complicate the #if even further figuring out one system from the next, this patch directly tests for the three struct members I know about today, and #ifdef's accordingly. Review: https://reviewboard.asterisk.org/r/3273/ ........ Merged revisions 409833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409834 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409835 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6204ea6c1a	AO2: Add an assert for bad objects This adds an assert that will only be active if Asterisk is compiled with DO_CRASH and allows the testsuite to fail tests that would otherwise require log file parsing. ........ Merged revisions 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409567 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	43858c24ab	doxygen: Tweak the link back to ye olde Digium website ........ Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409363 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c95146269c	devicestate.c: Simplified some logic in _ast_device_state(). ........ Merged revisions 409274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	77625956dd	stasis_cache.c: Remove some unnecessary RAII_VAR() usage. ........ Merged revisions 409272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f9c031ec39	stasis.c: Misc code cleanups. * Remove some unnecessary RAII_VAR() usage. * Made the struct stasis_subscription ao2 object use the ao2 lock instead of a redundant join_lock in the struct for ast_cond_wait(). * Removed locks on some ao2 objects that don't need the lock. * Made the topic pool entries container use the ao2 template functions. * Add some missing allocation failure checks. * Add missing cleanup in off nominal path of dispatch_message(). ........ Merged revisions 409270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	31707b1d69	main: Initialize dialplan providing core components prior to module pre-load It is possible to pre-load pbx_config. As a result, pbx_config - which will load and parse the dialplan - will attempt to use various dialplan components, such as device state providers and presence state providers, prior to them being initialized by the core. This would lead to a crash, as the components had not created their Stasis cache entries. This patch moves a number of core component initializations before the module pre-load. This guarantees that if someone does pre-load pbx_config - or other pbx modules - that the Stasis caches for the various core components are created. (closes issue ASTERISK-23320) Reported by: xrobau (closes issue ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy, Rusty Newton ........ Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	e468e73b9e	Remove extra defines of AST_PBX_MAX_STACK. * Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ ........ Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408786 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408787 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	73709e22ef	rtp_engine: Dynamic payload change in rtp mapping not supported Asterisk didn't support the dynamic payload change in rtp mapping in the 200 OK response. Scenario: Asterisk sends the INVITE proposing alaw and telephone-event, it proposes rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with alaw and telephone-event also, but it proposes a different rtpmap number (rtpmap:103) for telephone-event. Expected Behaviour: Asterisk should honour the rtpmapping in the response and send DTMF packets using 103 as payload type for DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload type 101. With this patch asterisk now supports changes that can occur in the rtp mapping in the response. (closes issue ASTERISK-23279) Reported by: NITESH BANSAL Review: https://reviewboard.asterisk.org/r/3225/ Patches: dynamic_payload_change.patch uploaded by nbansal (license 6418) ........ Merged revisions 408729 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9e6407c07b	manager: Fix AMI Status action of a single channel. Fixed use of uninitialized ao2 container iterator in an off-nominal condition. Either a memory allocation error or the requested channel is an internal channel not exposed to the outside. ........ Merged revisions 408715 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d277f3ec3e	json: Fix off-nominal json ref counting issues. * 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	12 years ago
Richard Mudgett	eec8ccc10b	json: Fix json API wrapper code for json library versions earlier than 2.3.0. * Fixed json ref counting issue with json API wrapper code for ast_json_object_update_existing() and ast_json_object_update_missing() when the json library is earlier than version 2.3.0. ........ Merged revisions 408711 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	b88c818153	rtp_engine: Output mixup in ${CHANNEL(rtpqos,audio,all)} Fixed the output of CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter. (closes issue ASTERISK-23261) Reported by: rsw686 Patches: rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged revisions 408646 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408647 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408649 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	41a80d6a9f	channel.c: MOH is not working for transferee after attended transfer Updated the code to check to see if MOH is playing on the transferor and if so then start it on the channel that replaces it during a masquerade. Example scenario of the problem: Alice calls Bob and then Bob begins the attended transfer process into a queue. Upon going on hold Alice hears music and so does Bob once he is in the queue. Bob then transfers Alice into the queue and then music for Alice stops even though she should be hearing it since has now replaced Bob in the queue. The problem that was occurring is that once the channel was masqueraded the app (queues, confbridge, etc...) had no way of knowing that the channel had just been swapped out thus it did not start music for the present channel. Credit to Olle Johansson for pointing me in the right direction on this issue. (closes issue ASTERISK-19499) Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/3226/ ........ Merged revisions 408642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408643 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408644 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408645 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	31a18c14b8	pjsip_cli: Fix memory leak in ast_sip_cli_print_sorcery_objectset. Fixed memory leaks in ast_sip_cli_print_sorcery_objectset and ast_variable_list_sort. (closes issue ASTERISK-23266) Review: http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions 408520 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

1 2 3 4 5 ...

4486 Commits (2dae3d6ea30c3a5879049a510a81102e2fb97d3a)