23767 Commits (72828808c88cd95cbc5014da931275bccda4208d)

 

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Author	SHA1	Message	Date
Richard Mudgett	72828808c8	confbridge: Make search the conference bridges container using OBJ_KEY. ... * Make confbridge config parsing user profile, bridge profile, and menu container hash/cmp functions correctly check the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags. * Made confbridge load_module()/unload_module() free all resources on failure conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e6d77dcdb5	Fix some bad whitespace ... This crept in with the RESTful HTTP interface merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c137d12111	Fix crash when AMI redirect action redirects two channels out of a bridge. ... The two party bridging loops were changing the bridge peer pointers without the channel locks held. Thus when ast_channel_massquerade() tested and used the pointer there is a small window of opportunity for the pointers to become NULL even though the masquerade code has the channels locked. (closes issue ASTERISK-21356) Reported by: William luke Patches: jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett Tested by: William luke ........ Merged revisions 386256 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386286 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Andrew Latham	e05870ccae	Doxygen - Markup Guidelines ... Expand on a commit by OEJ to use the Coding-Guidelines (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1c21b8575b	This patch adds a RESTful HTTP interface to Asterisk. ... The API itself is documented using Swagger, a lightweight mechanism for documenting RESTful API's using JSON. This allows us to use swagger-ui to provide executable documentation for the API, generate client bindings in different languages, and generate a lot of the boilerplate code for implementing the RESTful bindings. The API docs live in the rest-api/ directory. The RESTful bindings are generated from the Swagger API docs using a set of Mustache templates. The code generator is written in Python, and uses Pystache. Pystache has no dependencies, and be installed easily using pip. Code generation code lives in rest-api-templates/. The generated code reduces a lot of boilerplate when it comes to handling HTTP requests. It also helps us have greater consistency in the REST API. (closes issue ASTERISK-20891) Review: https://reviewboard.asterisk.org/r/2376/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Olle Johansson	1871017cc6	Fix mistake in Doxygen. ... Doxygen is only *ONE* comment that applies to the NEXT piece of code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	1cb52c6026	sla: remove redundant locking. ... sla.lock was already locked in the only place that sla_check_reload() was called. Remove the redundant locking of sla.lock done in this function. Less recursive locking is A Good Thing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	511cf82367	Prevent res_timing_pthread from blocking callers ... There were several reports of deadlock when using res_timing_pthread. Backtraces indicated that one thread was blocked waiting for the write to the pipe to complete and this thread held the container lock for the timers. Therefore any thread that wanted to create a new timer or read an existing timer would block waiting for either the timer lock or the container lock and deadlock ensued. This patch changes the way the pipe is used to eliminate this source of deadlocks: 1) The pipe is placed in non-blocking mode so that it would never block even if the following changes someone fail... 2) Instead of writing bytes into the pipe for each "tick" that's fired the pipe now has two states--signaled and unsignaled. If signaled, the pipe is hot and any pollers of the read side filedescriptor will be woken up. If unsigned the pipe is idle. This eliminates even the chance of filling up the pipe and reduces the potential overhead of calling unnecessary writes. 3) Since we're tracking the signaled / unsignaled state, we can eliminate the exta poll system call for every firing because we know that there is data to be read. (closes issue ASTERISK-21389) Reported by: Matt Jordan Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis patches: 0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417) (closes issue ASTERISK-19754) Reported by: Nikola Ciprich (closes issue ASTERISK-20577) Reported by: Kien Kennedy (closes issue ASTERISK-17436) Reported by: Henry Fernandes (closes issue ASTERISK-17467) Reported by: isrl (closes issue ASTERISK-17458) Reported by: isrl Review: https://reviewboard.asterisk.org/r/2441/ ........ Merged revisions 386109 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386159 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e61cc22404	cli.c: Properly initialize debug_modules and verbose_modules. ... This avoids some lock errors on the core set {debug,verbose} commands. ........ Merged revisions 386049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386051 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9c696e665f	Allow WebSocket connections on more URL's ... This patch adds the concept of ast_websocket_server to res_http_websocket, allowing WebSocket connections on URL's more more than /ws. The existing funcitons for managing the WebSocket subprotocols on /ws still work, so this patch should be completely backward compatible. (closes issue ASTERISK-21279) Review: https://reviewboard.asterisk.org/r/2453/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	aff127a737	Fix lock errors on startup. ... In messages.c, there are several places in the code where we create a tmp_tech_holder and pass that into an ao2_find call. Unfortunately, we weren't initializing the rwlock on the tmp_tech_holder, which the hash function was locking. It's apparently harmless, but still not the best code. This patch extracts all that copy/pasted code into two functions, msg_find_by_tech and msg_find_by_tech_name, which properly initialize and destroy the rwlock on the tmp_tech_holder. Review: https://reviewboard.asterisk.org/r/2454/ ........ Merged revisions 386006 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	fed7249ad1	res_xmpp and res_jabber need to search 'cachable' in the attrib section of the received IE, not data. ... (issue ASTERISK-20175) (closes issue ASTERISK-21429) (closes issue ASTERISK-21069) (closes issue ASTERISK-21164) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2452/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1f71abdfae	Allow res_corosync to build ... ast_enable_distributed_devstate is no longer applicable to how the distributed device state system works and is no longer necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	71a01725b8	Move presence state distribution to Stasis-core ... Convert presence state events to Stasis-core messages and remove redundant serializers where possible. Review: https://reviewboard.asterisk.org/r/2410/ (closes issue ASTERISK-21102) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	191cf99ae1	Move device state distribution to Stasis-core ... In the move from Asterisk's event system to Stasis, this makes distributed device state aggregation always-on, removes unnecessary task processors where possible, and collapses aggregate and non-aggregate states into a single cache for ease of retrieval. This also removes an intermediary step in device state aggregation. Review: https://reviewboard.asterisk.org/r/2389/ (closes issue ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c1ae5dc49b	Fixed a typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	f4870aa71c	Don't unnecessarily rebuild things on every run of 'make'. ... Review: https://reviewboard.asterisk.org/r/2449/ ........ Merged revisions 385745 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385768 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4b9e9fd2c0	Avoid unused variable warning when not in devmode ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c599aca553	Moved core logic from app_stasis to res_stasis ... After some discussion on asterisk-dev, it was decided that the bulk of the logic in app_stasis actually belongs in a resource module instead of the application module. This patch does that, leaves the app specific stuff in app_stasis, and fixes up everything else to be consistent with that change. * Renamed test_app_stasis to test_res_stasis * Renamed app_stasis.h to stasis_app.h * This is still stasis application support, even though it's no longer in an app_ module. The name should never have been tied to the type of module, anyways. * Now that json isn't a resource module anymore, moved the ast_channel_snapshot_to_json function to main/stasis_channels.c, where it makes more sense. Review: https://reviewboard.asterisk.org/r/2430/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2450722f52	DTMF events are now published on a channel's stasis_topic. AMI was ... refactored to use these events rather than producing the events directly in channel.c. Finally, the code was added to app_stasis to produce DTMF events on the WebSocket. The AMI events are completely backward compatible, including sending events on transmitted DTMF, and sending DTMF start events. The Stasis-HTTP events are somewhat simplified. Since DTMF start and DTMF send events are generally less useful, Stasis-HTTP will only send events on received DTMF end. (closes issue ASTERISK-21282) (closes issue ASTERISK-21359) Review: https://reviewboard.asterisk.org/r/2439 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c6cf12408e	Fix the svn:keywords property on several files. ... Normally I think keyword expansion is silly, but the one time it would have been good, it didn't work because the property had quotes in it. This patch fixes obviously busted svn:keywords properties. ........ Merged revisions 385683 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385689 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	2dd40e9707	Calculate the timestamp for outbound RTP if we don't have timing information ... This patch calculates the timestamp for outbound RTP when we don't have timing information. This uses the same approach in res_rtp_asterisk. Thanks to both Pietro and Tzafrir for providing patches. (closes issue ASTERISK-19883) Reported by: Giacomo Trovato Tested by: Pietro Bertera, Tzafrir Cohen patches: rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035) rtp-timestamp.patch uploaded by pbertera (License 5943) ........ Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385637 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5111744214	Don't attempt to create a voice frame on a read error ... Prior to this patch, a read error in snd_pcm_readi would still be treated as a nominal result when constructing a voice frame from the expected data. Since the value returned is negative, as opposed to the number of samples read, this could result in a crash. With this patch, we now return a null frame when a read error is detected. Note that the patch on ASTERISK-21329 was modified slightly for this commit, in that we bail immediately on detecting the read error, rather than bypassing the construction of the voice frame. (closes issue ASTERISK-21329) Reported by: Keiichiro Kawasaki patches: chan_alsa.diff uploaded by kawasaki (License 6489) ........ Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385634 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	a7c5183d67	Fix Manager Segfault When app_queue Is Unloaded ... When app_queue is unloaded, some manager commands are not being unregistered which result in a segfault. This patch corrects this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann, Corey Farrell Tested by: Corey Farrell Patches: asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L. Young (license 5026) asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2444/ ........ Merged revisions 385593 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385594 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	510b3b3594	Allow codec_resample to be unloaded ... Ensure that trans_size is correct to prevent uninitialized entries from preventing reload. (closes issue ASTERISK-21401) Reported by: Corey Farrell Tested by: Corey Farrell Patches: codec_resample-unload.patch uploaded by Corey Farrell ........ Merged revisions 385582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	1a09839e6b	Fix app_voicemail Segfault And A Few Memory Leaks ... The original report was that app_voicemail would crash. This was caused by ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being performed for that return status. After adding the initial patch to fix this issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered. During review, Walter Doekes (wdoekes) suggested adding a helper function in order to determine if we had a valid configuration or not. This patch does the following: * Creates a helper function to check if the configuration is valid * Adds calls to the new helper function where appropiate * Fixes memory leaks where the code returned without running ast_config_destroy() on the configuration that was loaded (closes issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco Kroon, Michael L. Young Patches: asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2443/ ........ Merged revisions 385551 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385557 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	ab6c0e74f1	Fix documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7f885dc31d	Expose channel snapshot manager blob generation ... These functions are already used in one branch (jrose's parking branch) and will soon be used in other branches as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	fcbb9f0c8d	Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX ... When we reload Asterisk or chan_sip, the flags force_rport and comedia that are turned on and off when using the auto_force_rport and auto_comedia nat settings go back to the default setting off. These flags are turned on when needed or off when not needed at the time that a peer registers, re-registers or initiates a call. This would apply even when only the default global setting "nat=auto_force_rport" is being used, which in this case would only affect the force_rport flag. Everything is good except for the following: The nat setting is set to auto_force_rport and auto_comedia. We reload Asterisk and the peer's registration has not expired. We load in the settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, those flags remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not. This patch does the following: * Moves the checking of whether a peer is behind NAT into its own function * Create a function to set the peer's NAT flags if they are using the auto_* NAT settings * Adds calls in sip_request_call() to these new functions in order to setup the dialog according to the peer's settings (closes issue ASTERISK-21374) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2421/ ........ Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	e5b0de5535	IAX2 defer_full_frames fail to get sent ... Ensure iax2_process_thread is signalled when a deferred frame is queued to it. (closes issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2426/ ........ Merged revisions 385429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385430 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	3959535615	IAX2, prevent network thread starting before all helper threads are ready ... On startup, it's possible for a frame to arrive before the processing threads were ready. In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive before we are at ast_cond_wait, the signal will be ignored. The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued. Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled after each thread is started. (issue ASTERISK-18827) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2427/ ........ Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385403 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	67cc3b53c9	Blocked revisions 385356 ... ........ Add dependency on libuuid, for res_rtp_asterisk pjproject is what actually requires libuuid. (closes issue ASTERISK-21125) reported by Private Name (Ed. note: Really? Private Name? I am rolling my eyes so hard right now.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	13e2aae2ef	Fix 'pri intense debug span' alias. ... ........ Merged revisions 385313 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3afeac5e3b	Eliminated dial_features_destroy() since it is equivalent to ast_free_ptr() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	eb2d144195	* Fix unlocked accesses to feature_list. The feature_list is now also ... protected by the features_lock. * Made all calls to ast_find_call_feature() have the features_lock held. * Fixed set_config_flags() to actually use find_group() to look for feature groups in DYNAMIC_FEATURES. The code originally assumed all feature groups were listed in DYNAMIC_FEATURES. * Make everyone use ast_rdlock_call_features(), ast_unlock_call_features(), and new ast_wrlock_call_features() instead of directly calling the rwlock API on features_lock. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	ff7ecd3dbf	Fixed manager channelvars support. ... For the events that have been ported to Stasis, this was broken in r384910, when a couple of lines of code was lost in a merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1f3fff7b91	Use LDAP memory management functions instead of Asterisk's ... When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk: invalid pointer errors) can occur as the memory is being allocated with Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the LDAP library's wrappers. This patch uses the LDAP library's wrappers where appropriate, so that compiling with MALLOC_DEBUG doesn't cause more problems than it solves. Note that the patch listed below was modified slightly for this commit to account for some additional memory allocation/deallocations. (closes issue ASTERISK-17386) Reported by: John Covert Tested by: Andrew Latham patches: issue18789-1.8-r316873.patch uploaded by seanbright (License 5060) ........ Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385199 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	caf4a5f605	Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE ... When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d09eeaa8eb	Rename struct feature_ds to struct feature_datastore. ... Because "struct feature_ds *feature_ds" is not a good thing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	0cc9528f9d	Backported app_stasis fix from stasis-http branch. ... The hash and compare functions for the control container was reusing the wrong ones, causing some problems. I fixed it, but in the wrong branch. Oh well, it happens. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	ee05bdec92	Add inheritance support to FEATURE()/FEATUREMAP(). ... The settings saved on the channel for FEATURE()/FEATUREMAP() were only for that channel. This patch adds the ability to have these settings inherited to child channels if you set FEATURE(inherit)=yes. Closes issue ASTERISK-21306. Review: https://reviewboard.asterisk.org/r/2415/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	98f2318559	Modified the list of keys for the driver backends for sake of sample clarity ... Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends. ........ Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 385048 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	5bf283d406	Clean up Makefile "warning" clutter when makeopts doesn't exist. ... Review: https://reviewboard.asterisk.org/r/2304 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a5df2542c3	Don't attempt a websocket protocol removal if res_http_websocket isn't there ... This patch sets the protocols container provided by res_http_websocket to NULL when the module gets unloaded and adds the necessary checks when adding/ removing a websocket protocol. This prevents some FRACKing on an invalid pointer to the disposed container if a module that uses res_http_websocket is unloaded after it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b8d4e573f1	Add multi-channel Stasis messages; refactor Dial AMI events to Stasis ... This patch does the following: * A new Stasis payload has been defined for multi-channel messages. This payload can store multiple ast_channel_snapshot objects along with a single JSON blob. The payload object itself is opaque; the snapshots are stored in a container keyed by roles. APIs have been provided to query for and retrieve the snapshots from the payload object. * The Dial AMI events have been refactored onto Stasis. This includes dial messages in app_dial, as well as the core dialing framework. The AMI events have been modified to send out a DialBegin/DialEnd events, as opposed to the subevent type that was previously used. * Stasis messages, types, and other objects related to channels have been placed in their own file, stasis_channels. Unit tests for some of these objects/messages have also been written. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a2a53cc306	Stasis application WebSocket support ... This is the API that binds the Stasis dialplan application to external Stasis applications. It also adds the beginnings of WebSocket application support. This module registers a dialplan function named Stasis, which is used to put a channel into the named Stasis app. As a channel enters and leaves the Stasis diaplan application, the Stasis app receives a 'stasis-start' and 'stasis-end' events. Stasis apps register themselves using the stasis_app_register and stasis_app_unregister functions. Messages are sent to an application using stasis_app_send. Finally, Stasis apps control channels through the use of the stasis_app_control object, and the family of stasis_app_control_* functions. Other changes along for the ride are: * An ast_frame_dtor function that's RAII_VAR safe * Some common JSON encoders for name/number, timeval, and context/extension/priority Review: https://reviewboard.asterisk.org/r/2361/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	426095bc55	Add a res_sorcery_astdb module which uses the astdb to persist objects. ... Review: https://reviewboard.asterisk.org/r/2420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	03286cf23f	Fix For Not Overriding The Default Settings In chan_sip ... The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. This issue is also present in Asterisk 1.8 and a separate patch will be applied to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2385/ ........ Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b8e5189456	Separate some event struct definitions from instantiation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a25d49296	chan_dahdi: Change inband_on_proceeding option default to no/disabled. ... (issue ASTERISK-21151) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago