3225 Commits (20ee33e22e3724c35ab3078c5032b87af5b4920c)

Author	SHA1	Message	Date
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
Richard Mudgett	bd029688cd	res_pjsip: Move internal init/destroy prototypes to private header file. ... Done as a separate commit from a finding in https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	c24a294f0b	res_pjsip: Fix pjsip.conf type=global object default value handling. ... When a type=global section is not defined in pjsip.conf the global defaults are not applied. As a result the mandatory Max-Forwards header is not added to SIP messages for res_pjsip/chan_pjsip. The handling of pjsip.conf type=global objects has several problems: 1) If the global object is missing the defaults are not applied. 2) If the global object is missing the default_outbound_endpoint's default value is not returned by ast_sip_global_default_outbound_endpoint(). 3) Defines are needed so default values only need to be changed in one place. * Added a sorcery instance observer callback to check if there were any type=global sections loaded. If there were more than one then issue an error message. If there were none then apply the global defaults. * Fixed ast_sip_global_default_outbound_endpoint() to return the documented default when no type=global object is defined. * Made defines for the global default values. * Increased the default_useragent[] size because SVN version strings can get lengthy and 128 characters may not be enough. * Fixed an off-nominal code path ref leak in global_alloc() if the string fields fail to initialize. * Eliminated RAII_VAR in get_global_cfg() and ast_sip_global_default_outbound_endpoint(). ASTERISK-24807 #close Reported by: Anatoli Review: https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	cae712d986	res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER. ... A race condition happened between initiating a transfer and requesting that a dialog termination be delayed. Occasionally, the transferrer channels would exit the bridge and hangup before the dialog termination delay was requested. * Made request dialog termination delay before initiating the transfer action. If the transfer fails then cancel the delayed dialog termination request. ASTERISK-24755 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4460/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	110b99646c	res_pjsip: Allow configuration of endpoint identifier query order ... It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Jonathan Rose	772793f18e	app: Add functions to swap voicemail function table for testing purposes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	9d85e855de	ARI: Fix crash if integer values used in JSON payload 'variables' object. ... Sending the following ARI commands caused Asterisk to crash if the JSON body 'variables' object passes values of types other than strings. POST /ari/channels POST /ari/channels/{channelid} PUT /ari/endpoints/sendMessage PUT /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(), ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751 #close Reported by: jeffrey putnam Review: https://reviewboard.asterisk.org/r/4447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	c33c5183a5	Dial API: add self destruct option when complete ... This patch adds a self-destruction option to the dial api. The usefulness of this is mostly when using async mode to spawn a separate thread used to handle the new call, while the calling thread is allowed to go on about other business. The only alternative to this option would be the calling thread spawning a new thread, or hanging around itself waiting to destroy the dial struct after completion. Example of use (minus error checking): struct ast_dial *dial = ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo"); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1); The dial_run call will return almost immediately after spawning the new thread to run and monitor the dial. If the call is answered, it is placed into the echo app. When completed, it will call ast_dial_destroy() on the dial structure. Note that any allocations made to pass values to ast_dial_set_user_data() or dial options must be free'd in a state callback function on any of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED, AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review: https://reviewboard.asterisk.org/r/4443/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	54a699fb64	asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers. ... Add a couple of missing closing brackets / parenthesis. ASTERISK-24814 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4436/ ........ Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	eb9448a1ae	Create work around for scheduler leaks during shutdown. ... * Added ast_sched_clean_by_callback for cleanup of scheduled events that have not yet fired. * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. Cleanup of replace_callno events is only run 11, since it no longer releases any references or allocations in 13+. ASTERISK-24451 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........ Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	1995baad71	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app ... This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	feddab7944	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	2531f75057	Enable REF_DEBUG for ast_module_ref / ast_module_unref. ... Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Ashley Sanders	112d23c73e	HTTP: For httpd server, need option to define server name for security purposes ... Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	f7d23dfcc6	stasis transfer: fix stasis bridge push race part two ... When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
David M. Lee	702d79de2a	Various fixes for OS X ... This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	630eea087d	Investigate and fix memory leaks in Asterisk ... Fixed memory leaks that were found in Asterisk. ASTERISK-24693 #close Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4347/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	e23f07beb8	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	355eb9d22f	Bridge core: Pass a ref with the swap channel when joining a bridge. ... When code imparts a channel into a bridge to swap with another channel, a ref needs to be held on the swap channel to ensure that it cannot dissapear before finding it in the bridge. * The ast_bridge_join() swap channel parameter now always steals a ref for the swap channel. This is the only change to the bridge framework's public API semantics. * bridge_channel_internal_join() now requires the bridge_channel->swap channel to pass in a ref. ASTERISK-24649 Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4354/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	8bc4a89e1f	Add support for the ca_list_path option for PJSIP transports. ... This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	056f11ac65	chan_pjsip: Add configure check for 'pjsip_get_dest_info' function. ... The 'pjsip_get_dest_info' function is used to determine if the signaling transport of the dialog is secure or not. This function was added in PJSIP 2.3 and does not exist in earlier versions. This configure check allows Asterisk to build and run with older versions at the loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of this argument will require upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4329/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	368ecf13bf	AMI: Revert non-backwards compatible changes from earlier commit. ... * Reverted the change to astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. Unfortunately changing the case of a returned value is not a backward compatible change so for now FAXSessions is going to have to remain inconsistent with all of the other AMI list actions. * Reverted the minor protocol error fix in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". Caught by the testsuite. ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4e2be8fb8f	main/syslog: Allow dynamic logs, such as security events, to log to the syslog ... The security event log uses a dynamic log level (SECURITY) that is registered with the Asterisk logging core. Unfortunately, the syslog would ignore log statements that had a dynamic log level associated with them. Because the syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic log entries sent to the syslog as logs with a level of NOTICE. ASTERISK-20744 #close Reported by: Michael Keuter Tested by: Michael L. Young, Jacek Konieczny patches: asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	eb9ce791d8	res_fax: Add T.38 negotiation timeout option ... This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	75cd302b0a	config: Add option to NOT preserve effective context when changing a template ... Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a7c38428af	pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions. ... The PJSIP_AOR dialplan function allows inspection of configured AORs including what contacts are currently bound to them. The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. These can include both externally added (by way of registration) or permanent ones. ASTERISK-24341 Reported by: xrobau Review: https://reviewboard.asterisk.org/r/4308/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	915bb88d3e	res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. ... Note that this is backport from trunk of r425825. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ ASTERISK-24644 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	006ffdcfb2	res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable. ... Note that this is a backport of r425804 from trunk. This change adds a configuration option which adds a 'user=phone' parameter if the user portion of the request URI or the From URI is determined to be a number. Review: https://reviewboard.asterisk.org/r/4073/ ASTERISK-24643 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fca0be57d9	queue_log: Post QUEUESTART entry when Asterisk fully boots. ... The QUEUESTART log entry has historically acted like a fully booted event for the queue_log file. When the QUEUESTART entry was posted to the log was broken by the change made by ASTERISK-15863. * Made post the QUEUESTART queue_log entry when Asterisk fully boots. This restores the intent of that log entry and happens after realtime has had a chance to load. AST-1444 #close Reported by: Denis Martinez Review: https://reviewboard.asterisk.org/r/4282/ ........ Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	2b8c441096	loader: Move definition of ast_module_reload from _private.h to module.h ... No functionality change. Just move the definition of ast_module_reload from _private.h to module.h so it can be public. Also removed the include of _private.h from manager.c since ast_module_load was the only reason for including it. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8f12ded887	DEBUG_THREADS: Fix regression and lock tracking initialization problems. ... This patch started with David Lee's patch at https://reviewboard.asterisk.org/r/2826/ and includes a regression fix introduced by the ASTERISK-22455 patch. The initialization of a mutex's lock tracking structure was not protected in a critical section. This is fine for any mutex that is explicitly initialized, but a static mutex may have its lock tracking double initialized if multiple threads attempt the first lock simultaneously. * Added a global mutex to properly serialize initialization of the lock tracking structure. The painful global lock can be mitigated by adding a double checked lock flag as discussed on the original review request. * Defer lock tracking initialization until first use. * Don't be "helpful" and initialize an uninitialized lock when DEBUG_THREADS is enabled. Debug code is not supposed to fix or change normal code behavior. We don't need a lock initialization race that would force a re-setup of lock tracking. Lock tracking already handles initialization on first use. * Properly handle allocation failures of the lock tracking structure. * No need to initialize tracking data in __ast_pthread_mutex_destroy() just to turn around and destroy it. The regression introduced by ASTERISK-22455 is the result of manipulating a pthread_mutex_t struct outside of the pthread library code. The pthread_mutex_t struct seems to have a global linked list pointer member that can get changed by other threads. Therefore, saving and restoring the contents of a pthread_mutex_t struct is a bad thing. Thanks to Thomas Airmont for finding this obscure regression. * Don't overwrite the struct ast_lock_track.reentr_mutex member to restore tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The pthread_mutex_t struct must be treated as a read-only opaque variable. Miscellaneous other items fixed by this patch: * Match ast_suspend_lock_info() with ast_restore_lock_info() in __ast_cond_timedwait(). * Made some uninitialized lock sanity checks return EINVAL and try a DO_THREAD_CRASH. * Fix bad canlog initialization expressions. ASTERISK-24614 #close Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/4247/ Review: https://reviewboard.asterisk.org/r/2826/ ........ Merged revisions 429539 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	439e6e1c5d	media: Fix crash when determining sample count of a frame during shutdown. ... When shutting down Asterisk the codecs are cleaned up. As a result anything attempting to get a codec based on ID or details will find that no codec exists. This currently occurs when determining the sample count of a frame. This code did not take this situation into account. This change fixes this by getting the codec directly from the format and eliminates the lookup. This is both faster and also provides a guarantee that the codec will exist and will be valid. ASTERISK-24604 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4260/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0c9fbb449f	res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress. ... Given the scenario where a PJSIP channel is in a native RTP bridge with direct media and the channel is then hung up the code will currently re-INVITE the channel back to Asterisk and send a BYE at the same time. Many SIP implementations dislike this greatly. This change makes it so that if a re-INVITE transaction is in progress the BYE is queued to occur after the completion of the transaction (be it through normal means or a timeout). Review: https://reviewboard.asterisk.org/r/4248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	525c823b4b	Direct Media calls within private network sometimes get one way audio ... When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media. ASTERISK-24563 Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ ........ Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cbc91f50c1	AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features ... AMI/ARI are getting a few enhancements in the next release of Asterisk 13. Per semantic versioning, that warrants a bump in the minor version number, as it reflects a backwards compatible change. Hence, this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	93b1df3bf6	Add new AMI and ARI events for connected line changes on a channel. ... The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	55c9a46abd	Stasis: Fix StasisStart/End order and missing events ... This corrects several bugs that currently exist in the stasis application code. * After a masquerade, the resulting channels have channel topics that do not match their uniqueids ** Masquerades now swap channel topics appropriately * StasisStart and StasisEnd messages are leaked to observer applications due to being published on channel topics ** StasisStart and StasisEnd publishing is now properly restricted to controlling apps via app topics * Race conditions exist where StasisStart and StasisEnd messages due to a masquerade may be received out of order due to being published on different topics ** These messages are now published directly on the app topic so this is now a non-issue * StasisEnds are sometimes missing when sent due to masquerades and bridge swaps into and out of Stasis() ** This was due to StasisEnd processing adjusting message-sent flags after Stasis() had already exited and Stasis() had been re-entered ** This was corrected by adjusting these flags prior to sending the message while the initial Stasis() application was still shutting down Review: https://reviewboard.asterisk.org/r/4213/ ASTERISK-24537 #close Reported by: Matt DiMeo ........ Merged revisions 429061 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	67372d1f0a	sorcery: Add additional observer capabilities. ... Add new global, instance and wizard observers. instance_created wizard_registered wizard_unregistered instance_destroying instance_loading instance_loaded wizard_mapped object_type_registered object_type_loading object_type_loaded wizard_loading wizard_loaded Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4215/ ........ Merged revisions 428999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a53efbb86c	config: Create ast_variable_find_in_list() ... Add const char *ast_variable_find_in_list(const struct ast_variable *list, const char *variable); ast_variable_find() requires a config category to search whereas ast_variable_find_in_list() just needs the root list element which is useful if you don't have a category. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4217/ ........ Merged revisions 428733 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8fb20cb05d	stringfields: Fix bug in ast_string_fields_copy. ... ast_string_fields_copy relies on the fact that __ast_string_field_release_active never previously zeroed pool->used, so keeping the existing pointer was "ok". Now that existing pools can be reset to 'empty', it is important to set each field to __ast_string_field_empty after releasing the memory. ASTERISK-24535 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4186/ ........ Merged revisions 428272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a0d9eab389	res_pjsip_sdp_rtp: Add support for optimistic SRTP. ... Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a1f1cdbd87	Allow for transferer to retry when dialing an invalid extension. ... This allows for a configurable number of attempts for a transferer to dial an extension to transfer the call to. For Asterisk 13, the default values are such that upgrading between versions will not cause a behaivour change. For trunk, though, the defaults will be changed to be more user-friendly. Review: https://reviewboard.asterisk.org/r/4167 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1536b0ecb6	Fix race condition that could result in ARI transfer messages not being sent. ... From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	77c57f2b1a	Fix race condition where duplicated requests may be handled by multiple threads. ... This is the Asterisk 13 version of the patch. The main difference is in the pubsub code since it was completely refactored between Asterisk 12 and 13. Review: https://reviewboard.asterisk.org/r/4175 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5013659972	Stasis: Fix StasisEnd message ordering ... This change corrects message ordering in cases where a channel-related message can be received after a Stasis/ARI application has received the StasisEnd message. The StasisEnd message was being passed to applications directly without waiting for the channel topic to empty. As a result of this fix, other bugs were also identified and fixed: * StasisStart messages were also being sent directly to apps and are now routed through the stasis message bus properly * Masquerade monitor datastores were being removed at the incorrect time in some cases and were causing StasisEnd messages to not be sent * General refactoring where necessary for the above * Unsubscription on StasisEnd timing changes to prevent additional messages from following the StasisEnd when they shouldn't A channel sanitization function pointer was added to reduce processing and AO2 lookups. Review: https://reviewboard.asterisk.org/r/4163/ ASTERISK-24501 #close Reported by: Matt Jordan ........ Merged revisions 427788 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2878554bcc	Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits. ... * Made collecting DTMF digits for the DTMF feature hooks pass frames from the bridge. * Made collecting DTMF digits possible by other bridge hooks if there is a need. ASTERISK-24447 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	433366ab90	Fix unintential memory retention in stringfields. ... * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	9d1b3ec22a	config: Make text_file_save and 'dialplan save' escape semicolons in values. ... When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ ........ Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	88d9d3f1fc	configure: Add autoconf check for libopus. ... Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. Review: https://reviewboard.asterisk.org/r/4106/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cb31a8aa7a	build: Force -fsigned-char on platforms where the default for char is unsigned ... gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and SPARC default to 'signed char'. This is only an issue in the rare cases where negative values are assigned to a 'char' but this this patch insures compatibility by detecting platforms that default to 'unsigned' and adding an '-fsigned-char' flag to _ASTCFLAGS. If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh and ./configure to regenerate the build files. You shouldn't have to do this for Intel or SPARC. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4091/ ........ Merged revisions 425964 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b8f505294a	config: Fix inf loop using ast_category_browse and ast_variable_retrieve ... Fix infinite loop when calling ast_variable_retrieve inside an ast_category_browse loop when there is more than 1 category with the same name. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4089/ ........ Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	289830cdc6	PJSIP: Enforce module load dependencies ... This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	54a5b8798f	config: Fix SEGV in unit test with MALLOC_DEBUG ... With MALLOC_DEBUG the /main/config config_basic_ops test was causing a SEGV while doing an ast_category_delete in an ast_category_browse loop. Apparently this never worked but was also never tested. I removed the test, added 2 notes to config.h indicating that it's not supported and added a few lines of code to ast_category_delete to prevent the SEGV should someone attempt it in the future. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4078/ ........ Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	26724c65f7	res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup ... Based on feedback from Richard, I created an accessor for res_phoneprov/ast_phoneprov_std_variable_lookup and added load priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ ........ Merged revisions 425480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8bb06d592a	manager/config: Support templates and non-unique category names via AMI ... This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d1c9621852	res_phoneprov: Refactor phoneprov to allow pluggable config providers ... This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6a844be566	chan_pjsip: Fix deadlock when masquerading PJSIP channels. ... Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	fade256307	PJSIP: Prevent T38 framehook being put on wrong channel ... This change gives framehooks a reverse-direction masquerade callback in addition to chan_fixup_cb similar to the callback added to datastores to handle the same situation. The new callback provides the same parameters as the fixup callback, but is called on the new channel's framehooks before moving framehooks from the old channel to the new channel. This gives the framehooks an oppurtunity to decide whether they should remain on the new channel or be removed. This new callback is used to prevent the PJSIP T.38 framehook from remaining on a masqueraded channel if the new channel is not also a PJSIP channel. This was causing a crash when a local channel was masqueraded into a PJSIP channel and the framehook was executed on the local channel since the channel's tech private data was not structured as expected. Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7ae706b2d	utils: Create ast_strsep function that ignores separators inside quotes ... This function acts like strsep with three exceptions... * The separator is a single character instead of a string. * Separators inside quotes are treated literally instead of like separators. * You can elect to have leading and trailing whitespace and quotes stripped from the result and have '\' sequences unescaped. Like strsep, ast_strsep maintains no internal state and you can call it recursively using different separators on the same storage. Also like strsep, for consistent results, consecutive separators are not collapsed so you may get an empty string as a valid result. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	23a375be5f	Add API call to determine if format capability structure is "empty". ... Empty here means that there are no formats in the format_cap structure or the only format in it is the "none" format. I've added calls to check the emptiness of a format_cap in a few places in order to short-circuit operations that would otherwise be pointless as well as to prevent some assertions from being triggered in cases where channels with no formats are used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7a35800551	res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies. ... res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE arrives. * It checks that there is a subscription handler for the Event * It checks that there are body generators for the types in the Accept header The problem is, there's nothing that ensures that these two things will actually mesh with each other. For instance, Asterisk will accept a subscription to MWI that accepts pidf+xml bodies. That doesn't make sense. With this commit, we add some type information to the mix. Subscription handlers state they generate data of type X, and body generators state that they consume data of type X. This way, Asterisk doesn't end up in some hilariously mismatched situation like the one in the previous paragraph. ASTERISK-24136 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3877 Review: https://reviewboard.asterisk.org/r/3878 ........ Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a2482acdce	res_pjsip: ami: Fix error in AMI output when an endpoint has no transport ... When no transport is associated to an endpoint, the AMI output for PJSIPShowEndpoint indicates an error instead of silently ignoring the missing transport. This patch causes the error to appear only if a transport was specified on the endpoint and the transport doesn't exist. It also fixes an issue with counting the objects that were actually found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3998/ ........ Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago