3188 Commits (e7edb59db6dfb543300f43c8055adda4ab1fd1c9)

Author	SHA1	Message	Date
Matt Jordan	620054c527	Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13	10 years ago
Kevin Harwell	b1e8c0b9eb	audiohook.c: Difference in read/write rates caused continuous buffer resets ... Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f	10 years ago
Matt Jordan	31cc24aad6	res/res_http_websocket: Add a pre-session established callback ... This patch updates http_websocket and its corresponding implementation with a pre-session established callback. This callback allows for WebSocket server consumers to be notified when a WebSocket connection is attempted, but before we accept it. Consumers can choose to reject the connection, if their application specific logic allows for it. As a result, this patch pulls out the previously private websocket_protocol struct and makes it public, as ast_websocket_protocol. In order to preserve backwards compatibility with existing modules, the existing APIs were left as-is, and new APIs were added for the creation of the ast_websocket_protocol as well as for adding a sub-protocol to a WebSocket server. In particular, the following new API calls were added: * ast_websocket_add_protocol2 - add a protocol to the core WebSocket server * ast_websocket_server_add_protocol2 - add a protocol to a specific WebSocket server * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. Consumers can populate this with whatever callbacks they wish to support, then add it to the core server or a specified server. ASTERISK-24988 Reported by: Joshua Colp Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2	10 years ago
George Joseph	dd78ab42e4	res_pjsip_config_wizard/config: Fix template processing ... The config wizard was always pulling the first occurrence of a variable from an ast_variable list but this gets the template value from the list instead of any overridden value. This patch creates ast_variable_find_last_in_list() in config.c and updates res_pjsip_config_wizard to use it instead of ast_variable_find_in_list. Now the overridden values, where they exist, are used instead of template variables. Updated test_config to test the new API. ASTERISK-25089 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4	10 years ago
George Joseph	637c8f065e	sorcery: Add API to insert/remove a wizard to/from an object type's list ... Currently you can 'apply' a wizard to an object type but the wizard always goes at the end of the object type's wizard list. This patch adds a new ast_sorcery_insert_wizard_mapping function that allows you to insert a wizard anyplace in the list. I.E. You could add a caching wizard to an object type and place it before all wizards. ast_sorcery_get_wizard_mapping_count and ast_sorcery_get_wizard_mapping were added to allow examination of the mapping list. ast_sorcery_remove_mapping was added to remove a mapping by name. As part of this patch, the object type's wizard list was converted from an ao2_container to an AST_VECTOR_RW. A new test was added to test_sorcery for this capability. ASTERISK-25044 #close Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57	10 years ago
George Joseph	ea917fefaf	vector: Add REMOVE, ADD_SORTED and RESET macros ... Based on feedback from Corey Farrell and Y Ateya, a few new macros have been added... AST_VECTOR_REMOVE which takes a parameter to indicate if order should be preserved. AST_VECTOR_ADD_SORTED which adds an element to a sorted vector. AST_VECTOR_RESET which cleans all elements from the vector leaving the storage intact. Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14	10 years ago
Joshua Colp	1e44d1bef9	Merge "res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination" into 13	10 years ago
Joshua Colp	d649d682c4	res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination ... The res_pjsip_exten_state module currently has a race condition between processing the extension state callback from the PBX core and processing the subscription shutdown callback from res_pjsip_pubsub. There is currently no synchronization between the two. This can present a problem as while the SIP subscription will remain valid the tree it points to may not. This is in particular a problem as a task to send a NOTIFY may get queued which will try to use the tree that may no longer be valid. This change does the following to fix this problem: 1. All access to the subscription tree is done within the task that sends the NOTIFY to ensure that no other thread is modifying or destroying the tree. This task executes on the serializer for the subscriptions. 2. A reference to the subscription serializer is kept to ensure it remains valid for the lifetime of the extension state subscription. 3. The NOTIFY task has been changed so it will no longer attempt to send a NOTIFY if the subscription has already been terminated. ASTERISK-25057 #close Reported by: Matt Jordan Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643	10 years ago
George Joseph	5f9aea8e3c	vector: Additional enhancements and fixes ... After using the new vector stuff for real I found... A bug in AST_VECTOR_INSERT_AT that could cause a seg fault. The callbacks needed to be closer to ao2_callback in behavior WRT to CMP_MATCH and CMP_STOP behavior and the ability to return a vector of matched entries. A pre-existing issue with APPEND and REPLACE was also fixed. I also added a new macro to test.h that acts like ast_test_validate but also accepts a return code variable and a cleanup label. As well as printing the error, it sets the rc variable to AST_TEST_FAIL and does a goto to the specified label on error. I had a local version of this in test_vector so I just moved it. ASTERISK-25045 Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc	10 years ago
George Joseph	7a7e9733c2	vector: Traversal, retrieval, insert and locking enhancements ... Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really does replace not insert. The few users of AST_VECTOR_INSERT were refactored. Because these are macros, there should be no ABI compatibility issues. Added AST_VECTOR_INSERT_AT that actually inserts an element into the vector at a specific index pushing existing elements to the right. Added AST_VECTOR_GET_CMP that can retrieve from the vector based on a user-provided compare function. Added AST_VECTOR_CALLBACK function that will execute a function for each element in the vector. Similar to ao2_callback and ao2_callback_data functions although the vector callback can take a variable number of arguments. This should allow easy migration to a vector where a container might be too heavy. Added read/write locked vector and lock manipulation macros. Added unit tests. ASTERISK-25045 #close Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0	10 years ago
Matt Jordan	74799b3fe2	Merge "Remove unneeded uses of optional_api providers." into 13	10 years ago
Matt Jordan	78c02f8e88	Merge "Update configure.ac/Makefile for clang" into 13	10 years ago
Corey Farrell	ad6ea29697	Remove unneeded uses of optional_api providers. ... A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478	10 years ago
Rodrigo Ramírez Norambuena	525c8c8689	include/asterisk/channel.h: Fix typo ... Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3	10 years ago
Diederik de Groot	9c3ed42875	Update configure.ac/Makefile for clang ... Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which checks compiler requirements for RAII: gcc: -fnested-functions support clang: -fblocks (and if required -lBlocksRuntime) The original check was implemented in configure.ac and now has it's own file. This function also sets C_COMPILER_FAMILY to either gcc or clang for use by makefile Created autoconf/ast_check_strsep_array_bounds.m4 (contains AST_CHECK_STRSEP_ARRAY_BOUNDS): which checks if clang is able to handle the optimized strsep & strcmp functions (linux). If not, the standard libc implementation should be used instead. Clang + the optimized macro's work with: strsep(char *, char []), but not with strsepo(char *, char *). Instead of replacing all the occurences throughout the source code, not using the optimized macro version seemed easier See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h': llvm-comment: Normally, this array-bounds warning are suppressed for macros, so that unused paths like the one that accesses __s1[3] are not warned about. But if you preprocess manually, and feed the result to another instance of clang, it will warn about all the possible forks of this particular if statement. Instead of switching of this optimization, another solution would be to run the preproces- sing step with -frewrite-includes, which should preserve enough information so that clang should still be able to suppress the diag- nostic at the compile step later on. See also "https://llvm.org/bugs/show_bug.cgi?id=20144" See also "https://llvm.org/bugs/show_bug.cgi?id=11536" Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning suppressions: -Wno-unused-value -Wno-parentheses-equality In an earlier review (reviewboard: 4550 and 4554), they were deemed a nuisace and less than benefitial. configure.ac: Added AST_CHECK_RAII() see earlier Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier Removed moved content ASTERISK-24917 Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb	10 years ago
Kevin Harwell	3fb6daeb55	res_fax: allow 2400 transmission rate according to v.27ter standard ... A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits per second. This reverts all or some of those patches since according to the v.27ter standard a rate of 2400 bits per second is also supported. One of the original patches also added 9600 bits per second support for v.27. This patch also removes that since v.27ter only supports 2400/4800 bits per second. Also, since Asterisk specifically supports v.27ter the enum was renamed to better reflect this. ASTERISK-24955 #close Reported by: Matt Jordan Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733	10 years ago
Mark Michelson	e39bd6ba46	res_pjsip_outbound_registration: Don't fail on delayed processing: 13. ... This is the Asterisk 13 version of a change to master that allows for registration responses to be processed successfully potentially after the original transaction has timed out. The main difference between this and the master change is that the master version has API changes that are unacceptable for 13. For 13, this is worked around by adding a new API call that the outbound registration code uses instead. The following is the text from the master version of this commit: Odd behaviors have been observed during outbound registrations. The most common problem witnessed has been one where a request with authentication credentials cannot be created after receiving a 401 response. Other behaviors include apparently processing an incorrect SIP response. Inspecting the code led to an apparent issue with regards to how we handle transactions in outbound registration code. When a response to a REGISTER arrives, we save a pointer to the transaction and then push a task onto the registration serializer. Between the time that we save the pointer and push the task, it's possible for the transaction to be destroyed due to a timeout. It's also possible for the address to be reused by the transaction layer for a new transaction. To allow for authentication of a REGISTER request to be authenticated after the transaction has timed out, we now also hold a reference to the original REGISTER request instead of the transaction. The function for creating a request with authentication has been altered to take the original request instead of the transaction where the original request was sent. ASTERISK-25020 Reported by Mark Michelson Change-Id: If1ee5f601be839479a219424f0358a229f358f7c	10 years ago
Matt Jordan	5a3948a66f	Merge "Fix/Update clang-RAII macro implementation" into 13	10 years ago
Joshua Colp	96e18453f4	Merge "pjsip_options: Fix non-qualified contacts showing as unavailable" into 13	10 years ago
Diederik de Groot	2be9cc2643	Fix/Update clang-RAII macro implementation ... - When you need to refer to 'variable XXX' outside a block, it needs to be declared as '__block XXX', otherwise it will not be available with- in the block, making updating that variable hard to do, and ast_free lead to issues. - Removed the #error message because it creates complications when compiling external projects against asterisk For example when using a different compiler than the one used to compile asterisk. The warning/error should be generated during the configure process not the compilation process ASTERISK-24917 Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2	10 years ago
George Joseph	63169e00ff	pjsip_options: Fix non-qualified contacts showing as unavailable ... The "Add qualify_timeout processing and eventing" patch introduced an issue where contacts that had qualify_frequency set to 0 were showing Unavailable instead Unknown. This patch checks for qualify_frequency=0 and create an "Unknown" contact_status with an RTT = 0. Previously, the lack of contact_status implied Unknown but since we're now changing endpoint state based on contact_status, I've had to add new UNKNOWN status so that changes could trigger the appropriate contact_status observers. ASTERISK-24977: #close Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7	10 years ago
Corey Farrell	c59a800707	Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled. ... When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be called as a function. This causes a compile error with raw threadstorage as it uses NULL for cleanup. This fix uses a macro that provides NULL when DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);" with "{};" when DEBUG_THREADLOCALS is enabled. ASTERISK-24975 #close Reported by: Ashley Sanders Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402	10 years ago
Matt Jordan	e05b076827	Merge "Detect potential forwarding loops based on count." into 13	10 years ago
Mark Michelson	4f1a8dbe92	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
George Joseph	674b18bdf0	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matt Jordan	f1abf51b73	Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13	10 years ago
George Joseph	bf46799f0e	res_pjsip: Refactor endpt_send_request to include transaction timeout ... This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html Since we currently have no control over pjproject transaction timeout, this patch pulls the pjsip_endpt_send_request function out of pjproject and into res_pjsip/endpt_send_transaction in order to implement that capability. Now when the transaction is initiated, we also schedule our own pj_timer with our own desired timeout. If the transaction completes before either timeout, pjproject cancels its timer, and calls our tsx callback where we cancel our timer and run the app callback. If the pjproject timer times out first, pjproject calls our tsx callback where we cancel our timer and run the app callback. If our timer times out first, we terminate the transaction which causes pjproject to cancel its timer and call our tsx callback where we run the app callback. Regardless of the scenario, pjproject is calling the tsx callback inside the group_lock and there are checks in the callback to make sure it doesn't run twice. As part of this patch ast_sip_send_out_of_dialog_request was created to replace its similarly named private function. It takes a new timeout argument in milliseconds (<= 0 to disable the timeout). ASTERISK-24863 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747	10 years ago
George Joseph	1b6f6ff841	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matt Jordan	d1a6f1a9f9	git migration: Remove support for file versions ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Alter the "core show file version" CLI command such that it always reports the version of Asterisk. The file version is no longer available. * main/manager: The Version key now always reports the Asterisk version. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action. - Modification of the "core show file version" CLI command. Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28	10 years ago
George Joseph	555b5f5d30	Add .gitignore and .gitreview files ... Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696 Tested-by: George Joseph	10 years ago
Richard Mudgett	13cd99682d	chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. ... With this patch, chan_pjsip/res_pjsip now sets the native formats to the codecs negotiated by a call. * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native formats to include all the negotiated audio codecs instead of only the initial preferred audio codec and later the currently received audio codec. * The audio frame handling in channel.c:ast_read() is more streamlined and will automatically adjust to changes in received frame formats. The new policy is to remove translation and pass the new frame format to the receiver except if the translation was to a signed linear format. A more long winded version is commented in ast_read() along with some caveats. * The audio frame handling in channel.c:ast_write() is more streamlined and will automatically adjust any needed translation to changes in the frame formats sent. Frame formats sent can change for many reasons such as a recording is being played back or the bridged peer changed the format it sends. Since it is a normal expectation that sent formats can change, the codec mismatch warning message is demoted to a debug message. * Removed the short circuit check in channel.c:ast_channel_make_compatible_helper(). Two party bridges need to make channels compatible with each other. However, transfers and moving channels among bridges can result in otherwise compatible channels having sub-optimal translation paths if the make compatible check is short circuited. A result of forcing the reevaluation of channel compatibility is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc options take effect consistently now. It is unfortunate that these two options are enabled by default and negate some of the benefits to the changes in channel.c:ast_read() by forcing translation through signed linear on a two party bridge. * Improved the softmix bridge technology to better control the translation of frames to the bridge. All of the incoming translation is now normally handled by ast_read() instead of splitting any translation steps between ast_read() and the slin factory. If any frame comes in with an unexpected format then the translation path in ast_read() is updated for the next frame and the slin factory handles the current frame translation. This is the final patch in a series of patches aimed at improving translation path choices. The other patches are on the following reviews: https://reviewboard.asterisk.org/r/4600/ https://reviewboard.asterisk.org/r/4605/ ASTERISK-24841 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4609/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	88b0fa7755	res_pjsip: Add an 'auto' option for DTMF Mode ... This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	6ba6e3dffd	clang compiler warnings: Fix autological comparisons ... This fixes autological comparison warnings in the following: * chan_skinny: letohl may return a signed or unsigned value, depending on the macro chosen * func_curl: Provide a specific cast to CURLoption to prevent mismatch * cel: Fix enum comparisons where the enum can never be negative * enum: Fix comparison of return result of dn_expand, which returns a signed int value * event: Fix enum comparisons where the enum can never be negative * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be negative * presencestate: Use the actual enum value for INVALID state * security_events: Fix enum comparisons where the enum can never be negative * udptl: Don't bother to check if the return value from encode_length is less than 0, as it returns an unsigned int * translate: Since the parameters are unsigned int, don't bother checking to see if they are negative. The cast to unsigned int would already blow past the matrix bounds. * res_pjsip_exten_state: Use a temporary value to cache the return of ast_hint_presence_state * res_stasis_playback: Fix enum comparisons where the enum can never be negative * res_stasis_recording: Add an enum value for the case where the recording operation is in error; fix enum comparisons * resource_bridges: Use enum value as opposed to -1 * resource_channels: Use enum value as opposed to -1 Review: https://reviewboard.asterisk.org/r/4533 ASTERISK-24917 Reported by: dkdegroot patches: rb4533.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4441bb6a25	Bridging: Eliminate the unnecessary make channel compatible with bridge operation. ... When a channel enters the bridging system it is first made compatible with the bridge and then the bridge technology makes the channel compatible with the technology. For all but the DAHDI native and softmix bridge technologies the make channel compatible with the bridge step is an effective noop because the other technologies allow all audio formats. For the DAHDI native bridge technology it doesn't matter because it is not an initial bridge technology and chan_dahdi allows only one native format per channel. For the softmix bridge technology, it is a noop at best and harmful at worst because the wrong translation path could be setup if the channel's native formats allow more than one audio format. This is an intermediate patch for a series of patches aimed at improving translation path choices. * Removed code dealing with the unnecessary step of making the channel compatible with the bridge. ASTERISK-24841 Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	95de71f247	build: Fixes for gcc 5 compilation ... These are fixes for compilation under gcc 5.0... chan_sip.c: In parse_request needed to make 'lim' unsigned. inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 inline semantics (same as clang). ccss.c: In ast_cc_set_parm, needed to fix weird comparison. dsp.c: Needed to work around a possible compiler bug. It was throwing an array-bounds error but neither sgriepentrog, rmudgett nor I could figure out why. manager.c: In action_atxfer, needed to correct an array allocation. This patch will go to 11, 13, trunk. Review: https://reviewboard.asterisk.org/r/4581/ Reported-by: Jeffrey Ollie Tested-by: George Joseph ASTERISK-24932 #close ........ Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	169e57d2e0	pjsip: resolve compatibility problem with ast_sip_session ... A change in r430179 inserted a variable near the top of a structure caused a problem when running DPMA in a version of Asterisk compiled across the change. This patch moves the new variable to the end of the structure, eliminating the problem. Review: https://reviewboard.asterisk.org/r/4574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	2d39bc5528	Fix an ABI compatibility issue with ast_log_safe for modules. ... Binary modules are sometimes built against the latest release of Asterisk in each branch, and need to be compatible with all releases of that branch. This change ensures that utils.h only uses ast_log_safe from the core. For modules and utilities ast_log is used instead. Review: https://reviewboard.asterisk.org/r/4548/ ........ Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	09b681e344	clang compiler warnings: Fix invalid enum conversion ... This patch fixes some invalid enum conversion warnings caught by clang. In particular: * chan_sip: Several functions mixed usage of the st_refresher_param enum and st_refresher enum. This patch corrects the functions to use the right enum. * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state. * strings: Fixed incorrect usage of AO2 flags with strings container. * res_stasis: Change a return enumeration to stasis_app_user_event_res. Review: https://reviewboard.asterisk.org/r/4535 ASTERISK-24917 Reported by: dkdegroot patches: rb4535.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	85feac857c	Add stateful PJSIP response API call, and use it for out-of-dialog responses. ... Asterisk had an issue where retransmissions of MESSAGE requests resulted in Asterisk processing the retransmission as if it were a new MESSAGE request. This patch fixes the issue by creating a transaction in PJSIP on the incoming request. This way, if a retransmission arrives, the PJSIP transaction layer will resend the response and Asterisk will not ever see the retransmission. ASTERISK-24920 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4532/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	b0df413fb2	Fix link error for utils/aelparse. ... Use the standard ast_log instead of ast_log_safe for STANDALONE programs. Review: https://reviewboard.asterisk.org/r/4538/ ........ Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	d01706ce1e	Improved and portable ast_log recursion avoidance ... This introduces a new logger routine ast_log_safe. This routine should be used for all error messages in code that can be run as a result of ast_log. ast_log_safe does nothing if run recursively. All error logging in astobj2.c, strings.c and utils.h have been switched to ast_log_safe. This required adding support for raw threadstorage. This provides direct access to the void* pointer in threadstorage. In ast_log_safe, NULL is used to signify that this thread is not already running ast_log_safe, (void*)1 when it is already running. This was done since it's critical that ast_log_safe do nothing that could log during recursion checking. ASTERISK-24155 #close Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/4502/ ........ Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	6adf26f14d	Replace most uses of ast_register_atexit with ast_register_cleanup. ... Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	fd434a210f	app_confbridge: file playback blocks dtmf ... Attempting to execute DTMF in a confbridge while file playback (prompt, announcement, etc) is occurring is not allowed. You have to wait until the sound file has completed before entering DTMF. This patch fixes it so that app_confbridge now monitors for dtmf key presses during menu driven file playback. If a key is pressed playback stops and it executes the matched menu option. ASTERISK-24864 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4510/ ........ Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	b1e9552b08	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. ... Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	2c7b945149	Various: bugfixes found via chaos ... Using DEBUG_CHAOS several instances of a null pointer crash, and one uninitialized variable were uncovered and fixed. Also added details on why Asterisk failed to initialize. Review: https://reviewboard.asterisk.org/r/4468/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	1fb1c81923	core: Introduce chaos into memory allocations ... Locate potential crashes by exercising seldom used code paths. This patch introduces a new define DEBUG_CHAOS, and mechanism to randomly return an error condition from functions that will seldom do so. Functions that handle the allocation of memory get the first treatment. Review: https://reviewboard.asterisk.org/r/4463/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	94fe4a9178	res_pjsip: Allow configuration of endpoint identifier query order ... Updated some documentation stating that endpoint identifiers registered without a name are place at the front of the lookup list. Also renamed register method 'ast_sip_register_endpoint_identifier_by_name' to 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	1f428f25f0	res_pjsip: Allow configuration of endpoint identifier query order ... This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	0497b7b155	Revert - res_pjsip: Allow configuration of endpoint identifier query order ... Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	f5bc032567	Add support for the clang compiler; update RAII_VAR to use BlocksRuntime ... RAII_VAR, which is used extensively in Asterisk to manage reference counted resources, uses a GCC extension to automatically invoke a cleanup function when a variable loses scope. While this functionality is incredibly useful and has prevented a large number of memory leaks, it also prevents Asterisk from being compiled with clang. This patch updates the RAII_VAR macro such that it can be compiled with clang. It makes use of the BlocksRuntime, which allows for a closure to be created that performs the actual cleanup. Note that this does not attempt to address the numerous warnings that the clang compiler catches in Asterisk. Much thanks for this patch goes to: * The folks on StackOverflow who asked this question and Leushenko for providing the answer that formed the basis of this code: http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang * Diederik de Groot, who has been extremely patient in working on getting this patch into Asterisk. Review: https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133 ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by: Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik de Groot (License 6600) ........ Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago