820 Commits (15.7)

Author	SHA1	Message	Date
Matt Jordan	3b9cba4294	funcs/func_cdr: Correctly report high precision values for duration and billsec ... When CDRs were refactored, func_cdr's ability to report high precision values for duration and billsec (the 'f' option) was broken. This was due to func_cdr incorrectly interpreting the duration/billsec values provided by the CDR engine in milliseconds, as opposed to seconds. Since the CDR engine only provides duration and billsec in seconds, and does not expose either attribute with sufficient precision to merely pass back the underlying value, this patch fixes the bug by re-calculating duration and billsec with microsecond precision based on the start/answer/end times on the CDR. ASTERISK-25179 #close Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841	9 years ago
Rusty Newton	68cad96ffd	func_channel: Add help text for undocumented CHANNEL function arguments ... Adding help text documentation for: * hangupsource * appname * appdata * exten * context * channame * uniqueid * linkedid ASTERISK-24097 #close Reported by: Steven T. Wheeler Tested by: Rusty Newton Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d	9 years ago
Joshua Colp	0be147f713	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased."	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Walter Doekes	7dd8f89a50	func_callerid: Document that CALLERID(pres) is available. ... CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a	10 years ago
Matt Jordan	7be6194d6f	funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function ... When ab803ec342 was committed, it accidentally forgot to actually *add* the HOLD_INTERCEPT function. This highlights two interesting points: * Gerrit forces you to put the patch as it is going to into the repo up for review, which Review Board did not. Yay Gerrit. * No one apparently bothered to use this feature, or else they don't know about it. I'm going to go with the latter explanation. ASTERISK-24922 Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396	10 years ago
Ivan Poddubny	c944263e36	func_presencestate: Return "not_set" when no data is set in AstDB ... Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not exist, i.e. when a new CustomPresence is added in the dialplan. ASTERISK-25400 #close Reported by: Andrew Nagy Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a	10 years ago
Rusty Newton	d02196448b	Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c ... * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 years ago
Corey Farrell	ea9d5f155e	func_pjsip_aor: Fix leaked contact from iterator. ... ASTERISK-25162 #close Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e	10 years ago
Corey Farrell	0a46d43b9c	Fix potential crash after unload of func_periodic_hook or test_message. ... These modules save a pointer to the context they create on load, and use that pointer to destroy the context at unload. It is not safe to save this pointer, it is replaced during load of pbx_config, pbx_lua or pbx_ael. This change causes the modules to pass NULL to ast_context_destroy, a safer way to perform the unregistration since it does not use a pointer that could become invalid. ASTERISK-25085 #close Reported by: Corey Farrell Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835	10 years ago
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. ... Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	10 years ago
George Joseph	298faf7c50	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
Mark Michelson	aae45acbda	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
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... 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: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Matthew Jordan	ea0098724e	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 ........ Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	af4d802773	clang compiler warnings: Fix sometimes-initialized warning in func_math ... This patch fixes a bug in a unit test in func_math where a variable could be passed to ast_free that wasn't allocated. This patch corrects the issue and ensures that we only attempt to free a variable if we previously allocated it. Review: https://reviewboard.asterisk.org/r/4552 ASTERISK-24917 Reported by: dkdegroot patches: rb4552.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434191 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	d2776d4d45	clang compiler warnings: Fix a variety of "unused" warnings ... This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable errors caught by clang. Specifically: * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], qsmp_cmd_usage[] * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" * funcs/func_env.c:729: Fixed ast_str_append_substr. * main/editline/np/strlcat.c: removed unused rcsid variable * main/editline/np/strlcpy.c: removed unused rcsid variable * main/security_events.c: removed unused TIMESTAMP_STR_LEN * utils/conf2ael.c: removed unused cfextension_states * utils/extconf.c: removed unused cfextension_states Review: https://reviewboard.asterisk.org/r/4526 ASTERISK-24917 Reported by: dkdegroot patches: rb4526.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433694 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	60f01520e7	Fix compilations errors on 64-bit OpenBSD systems ... In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to (long) when printing members of certain time structs. Review: https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close Reported by: snuffy Tested by: snuffy patches: openbsd-time64.diff uploaded by snuffy (License 5024) ........ Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433269 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	627cc16a8d	funcs/func_env: Fix regression caused in FILE read operation ... When r432935 was merged, it did correctly fix a situation where a FILE read operation on the middle of a file buffer would not read the requested length in the parameters passed to the FILE function. Unfortunately, it would also allow the FILE function to append more bytes than what was available in the buffer if the length exceeded the end of the buffer length. This patch takes the minimum of the remaining bytes in the buffer along with the calculated length to append provided by the original patch, and uses that as the length to append in the return result. This patch also updates the unit tests with the scenarios that were originally pointed out in ASTERISK-21765 that the original implementation treated incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433174 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	0d52907d2b	func_curl: Don't hold exclusive lock when performing HTTP request. ... This code originally kept a lock held when performing the HTTP request to ensure that the options provided to curl remain valid. This doesn't seem to be necessary these days and holding the lock caused requests to happen sequentially instead of in parallel. ASTERISK-18708 #close Reported by: Dave Cabot ........ Merged revisions 432948 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432949 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	b4cc056067	FILE: fix retrieval of file contents when offset is specified ... The loop that reads in a file was not correctly using the offset when determining what bytes to append to the output. This patch corrects the logic such that the correct portion of the file is extracted when an offset is specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by Di-Shi Sun (License 5076) ........ Merged revisions 432935 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432938 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	3cccfac399	Multiple revisions 431297-431298 ... ........ r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines Mitigate possible HTTP injection attacks using CURL() function in Asterisk. CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection can be performed given properly-crafted URLs. Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may get cURL URLs from user input or remote sources, we have made a patch to Asterisk to prevent such HTTP injection attacks from originating from Asterisk. ASTERISK-24676 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4364 AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from previous patch. ........ Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431299 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 431301 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
David M. Lee	965777ccfc	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/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	e4738a59eb	CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. ... Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	b38acbce6e	funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed ... When the channel datastore associated with the usage of CURLOPT on a specific channel is freed, the underlying structure holding the list of options is not disposed of. This patch properly frees the structure in the datastore .destroy callback. ASTERISK-24672 #close Reported by: Kristian Hogh patches: func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639) ........ Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	685f7ef924	func_config: Add ability to retrieve specific occurrence of a variable ... I guess nobody uses templates with AST_CONFIG because today if you have a context that inherits from a template and you call AST_CONFIG on the context, you'll get the value from the template even if you've overridden it in the context. This is because AST_CONFIG only gets the first occurrence which is always from the template. This patch adds an optional 'index' parameter to AST_CONFIG which lets you specify the exact occurrence to retrieve, or '-1' to retrieve the last. The default behavior is the current behavior. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4313/ ........ Merged revisions 430315 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	f7cf988a82	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/ ........ Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	2486b48cec	AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI. ... The DB dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Asterisk now inhibits the DB function from being executed from an external interface if the live_dangerously option is set to no. ASTERISK-24534 Reported by: Gareth Palmer patches: submitted by Gareth Palmer (license 5169) ........ Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 428363 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428409 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428413 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	97e1c7f3a9	func_talkdetect: Fix stasis message leak in audiohook callback. ... ASTERISK-24482 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4142/ ........ Merged revisions 427203 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427204 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ab16f46139	func_cdr: Fix CDR_PROP payload leak ... Remove duplicate allocation of payload, preventing leak. ASTERISK-24455 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4113/ ........ Merged revisions 426252 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	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 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). ... This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9183416fe2	func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel. ... It would be useful to get the current hold status of a channel. Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for the hold status of a channel. ASTERISK-24038 Reported by: Matt Jordan AFS-113 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	025bd1bf3f	func_channel.c: Add missing locking to some CHANNEL() requests. ... * The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and audiowriteformat now need locking since the media format rework when accessing the channel's format pointers. * Increased the buffer size for CHANNEL() audionativeformat and videonativeformat output strings since the allow=all can be a lengthy list. * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling, secure_bridge_media, and state. * Ensured the output buffer is initialized for secure_bridge_signaling and secure_bridge_media. * Made use the locked_copy_string() macro instead of inlining it for trace and checkhangup. ........ Merged revisions 422700 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	1de8b8035e	func_config: Change 'Not Found' message from ERROR to DEBUG ... When you call the CONFIG dialplan function with the name of a variable that doesn't exist in the target context you get an ERROR. This does nothing but clutter up the logs with messages that may be perfectly acceptable. Just because a variable wasn't in the context doesn't mean it's an error. Maybei t's optional or just needs to be defaulted or ignored. This patch changes the log level from ERROR to DEBUG. If a dialplan developer wants to debug their dialplan they still canby setting the console debug level as needed. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........ Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6650704414	funcs/func_jitterbuffer: Tweak documentation ... This patch merely reformats and cleans up a bit of the jitterbuffer documentation for the wiki. ........ Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8610daec3b	Add documentation to the ability to retrieve the source port of a SIP call. ... (belongs with r419970) ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by dtryba Review: https://reviewboard.asterisk.org/r/3781/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked ... This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	922e3203a9	xmldocs: Add support for an <example> tag in the Asterisk XML Documentation ... This patch adds support for an <example /> tag in the XML documentation schema. For CLI help, this doesn't change the formatting too much: - Preceeding white space is removed - Unlike with para elements, new lines are preserved However, having an <example /> tag in the XML schema allows for the wiki documentation generation script to surround the documentation with {code} or {noformat} tags, generating much better content for the wiki - and allowing us to put dialplan examples (and other code snippets, if desired) into the documentation for an application/function/AMI command/etc. Review: https://reviewboard.asterisk.org/r/3807/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2758cc76e5	datastores: Audit ast_channel_datastore_remove usage. ... Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in app_speech_utils and func_frame_trace. * Fixed app_speech_utils not locking the channel when accessing the channel datastore list. Review: https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leak in func_jitterbuffer. (Was not in v12) Review: https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in abstract_jb. * Fixed leak in ast_channel_unsuppress(). Used by ARI mute control and res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used by ARI mute control and res_mutestream. Review: https://reviewboard.asterisk.org/r/3861/ ........ Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419685 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419686 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	media formats: re-architect handling of media for performance improvements ... In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b71be2112e	func_audiohookinherit.c: Fixup some XML documentation wording. ... ........ Merged revisions 418937 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	af4cd65143	Channels: Masquerades to automatically move frame/audio hooks ... Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	4420ce6e5c	func_uri: URIENCODE/URIDECODE - allow empty strings as argument ... Previously these two dialplan functions would issue warnings and return failure when an empty string is used as the argument. Now they will not issue a warning and will successfully return an empty string. ASTERISK-23911 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3745/ ........ Merged revisions 418641 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 418649 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 418650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	fc52fd8884	Allow the PUSH and UNSHIFT functions to set inheritable channel variables. ... ........ Merged revisions 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416501 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416502 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	795af210a3	func_odbc: Fix fixed size buffers fix (r414968). ... The change that removed the fixed size buffers in odbc-related code -- removing arbitrary column width limits -- was incomplete. This change adds: no segfault on writesql without insertsql and return value checks after strdup. While I was in the vicinity I cleaned up the linefeeds in the odbc function descriptions, moved some code for clarity, removed some blobs and noted (but didn't fix) that the 'odbc write ... exec' CLI command doesn't behave as the dialplan equivalent when insertsql= is used. ASTERISK-23582 #close Review: https://reviewboard.asterisk.org/r/3579/ ........ Merged revisions 414997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414998 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	53968c00b3	TALK_DETECT: A channel function that raises events when talking is detected ... This patch adds a new channel function TALK_DETECT that, when set on a channel, causes events indicating the start/stop of talking on a channel to be emitted to both AMI and ARI clients. The function allows setting both the silence threshold (the length of silence after which we decide no one is talking) as well as the talking threshold (the amount of energy that counts as talking). Parameters can be updated on a channel after talk detection has been enabled, and talk detection can be removed at any time. The events raised by the function use a nomenclature similar to existing AMI/ARI events. For AMI: ChannelTalkingStart/ChannelTalkingStop For ARI: ChannelTalkingStarted/ChannelTalkingFinished Review: https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close Reported by: Matt Jordan ........ Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	dcfae78574	res_config_odbc: Use dynamically sized buffers to store row data so values do not get truncated. ... ASTERISK-23582 #close ASTERISk-23582 #comment Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/3557/ ........ Merged revisions 414693 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414694 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414695 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414696 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	43bd3580e2	pbx.c: prevent potential crash from recursive replace() ... Recurisve usage of replace() resulted in corruption of the temporary string storage and potential crash. By changing the string to be allocated separtely per instance, this is eliminated. ASTERISK-23650 #comment Reported by: Roel van Meer ASTERISK-23650 #close Review: https://reviewboard.asterisk.org/r/3539/ ........ Merged revisions 414214 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414215 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	21949def8e	Fix 32bit build for func_env ... ........ Merged revisions 413592 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413597 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	68066907e1	Ensure that presence state is decoded properly on Asterisk startup. ... The CustomPresence provider callback will automatically base64 decode stored data if the 'e' option was present when the state was set. However, since the provider callback was bypassed on Asterisk startup, encoded presence subtypes and messages were being sent instead. This fix makes it so the provider callback is always used when providing presence state updates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	120ac66df9	Make behavior of the PRESENCE_STATE 'e' option more consistent. ... When writing presence state, if 'e' is specified, then the presence state will be stored in the astdb encoded. However, consumers of presence state events or those that query for the presence state will be given decoded information. If base64 encoding is desired for consumers, then the information can be base64-encoded manually and the 'e' option can be omitted. closes issue ASTERISK-23671 Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3482 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	4b9b4790d9	Fix error loading res_monitor. ... For some odd reason, loading app_mixmonitor was fine, but res_monitor was not. This patch fixes a set of issues related to func_periodic_hook exporting the beep functions that gets res_monitor working again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	5b7a769fd8	(mix)monitor: Add options to enable a periodic beep ... Add an option to enable a periodic beep to be played into a call if it is being recorded. If enabled, it uses the PERIODIC_HOOK() function internally to play the 'beep' prompt into the call at a specified interval. This option is provided for both Monitor() and MixMonitor(). Review: https://reviewboard.asterisk.org/r/3424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	39ba6a16f5	func_periodic_hook: move module ref ... The previous code left one error path where the module would be unref'd twice instead of once. It was done once in the error handling block, and again inside of datastore destruction. Now the module ref is only released in the datastore destructor and only acquired when the datastore has been successfully allocated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	23329d07cf	func_periodic_hook: add module ref counting ... This module lacked necessary module ref count incrementing and decrementing when used. This patch adds it. There's already a datastore used, so doing the ref counting along with the lifetime of the datastore provides a convenient place to do it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	607c6e9772	func_periodic_hook: List more modules as dependencies ... This module makes use of some existing Asterisk components. app_chanspy was already listed as a dependency. There are a few function modules used, as well, so list them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	ea290b2c3e	func_periodic_hook: New function for periodic hooks. ... This commit introduces a new dialplan function, PERIODIC_HOOK(). It allows you run to a dialplan hook on a channel periodically. The original use case that inspired this was the ability to play a beep periodically into a call being recorded. The implementation is much more generic though and could be used for many other things. The implementation makes heavy use of existing Asterisk components. It uses a combination of Local channels and ChanSpy() to run some custom dialplan and inject any audio it generates into an active call. The other important bit of the implementation is how it figures out when to trigger the beep playback. This implementation uses the audiohook API, even though it's not actually touching the audio in any way. It's a convenient way to get a callback and check if it's time to kick off another beep. It would be nice if this was timer event based instead of polling based, but unfortunately I don't see a way to do it that won't interfere with other things. Review: https://reviewboard.asterisk.org/r/3362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	d44aefeef4	Fix stuck channel in ARI through the introduction of synchronous bridge actions. ... Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a4906e9f86	sorcery: Create AST_SORCERY dialplan function. ... This patch creates the AST_SORCERY dialplan function which allows someone to retrieve any value from a sorcery-based config file. It's similar to AST_CONFIG. The creation of the function itself was fairly straightforward but it required changes to the underlying sorcery infrastructure that rippled into individual sorcery objects. The changes stemmed from inconsistencies in how sorcery created ast_variable objectsets from sorcery objects and the inconsistency in how individual objects used that feature especially when it came to parameters that can be specified multiple times like contact in aor and match in identify. You can read more here... http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html So, what this patch does, besides actually creating the AST_SORCERY function, is the following... * Creates ast_variable_list_append which is a helper to append one ast_variable list to another. * Modifies the ast_sorcery_object_field_register functions to accept the already-defined sorcery_fields_handler callback. * Modifies ast_sorcery_objectset_create to accept a parameter indicating return type preference...a single ast_variable with all values concatenated or an ast_variable list with multiple entries. Also fixed a few bugs. * Modifies individual sorcery object implementations to use the new function definition of the ast_sorcery_object_field_register functions. * Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement sorcery_fields_handler handlers so they return multiple occurrences as an ast_variable_list. * Added a whole bunch of tests to test_sorcery. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	ad39358e19	Fix documentation for PRESENCE_STATE to properly illustrate how to create a presence hint. ... There was a missing comma. This was discovered by Dan Kaplan. ........ Merged revisions 409886 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409887 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	959c3a7c15	func_audiohookinheritance: Check If A Channel Was Specified ... This patch prevents a crash when using the function audiohookinheritance without setting the channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by: Joel Vandal Patches: asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3272/ ........ Merged revisions 409623 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409625 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409626 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5fd63e2d0b	funcs/func_cdr: Handle empty time values when extracting parsed values ... When extracting timestamps that are parsed, time stamp values that are not set (time values of 0.000000) should not actually result in a parsed string. The value should be skipped, and the result of the CDR function should be an empty string. Prior to this patch, the result was fed to the time formatting, which would result in an output of a date/time in 1969. ........ Merged revisions 407747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9e7a10d894	funcs/func_cdr: Fix non-epoch timestamps broken by improper char array deref ... Thanks to snuffy for pointing this issue out and fixing it. (closes issue ASTERISK-23250) Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy (License 5024) ........ Merged revisions 407259 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	828f339a9c	verbosity: Fix performance of console verbose messages. ... The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	373965dbff	CDRs: Synchronize dialplan applications that manipulate CDRs with the engine ... In https://reviewboard.asterisk.org/r/3057/, applications and functions that manipulate CDRs were made to interact over Stasis. This was done to synchronize manipulations of CDRs from the dialplan with the updates the engine itself receives over the message bus. This change rested on a faulty premise: that messages published to the CDR topic or to a topic that forwards to the CDR topic are synchronized with the messages handled by the CDR topic subscription in the CDR engine. This is not the case. There is no ordering guaranteed for two messages published to the same topic; ordering is only guaranteed if a message is published to the same subscriber. Stasis was modified in r405311 to allow a publisher to synchronize on the subscriber. This patch uses that API to synchronize the CDR publishers with the CDR engine message router, which maintains the overall topic subscription. (closes issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	ff02c8e254	func_strings: use memmove to prevent overlapping memory on strcpy ... When calling REPLACE() with an empty replace-char argument, strcpy is used to overwrite the the matching <find-char>. However as the src and dest arguments to strcpy must not overlap, it causes other parts of the string to be overwritten with adjacent characters and the result is mangled. Patch replaces call to strcpy with memmove and adds a test suite case for REPLACE. (closes issue ASTERISK-22910) Reported by: Gareth Palmer Review: https://reviewboard.asterisk.org/r/3083/ Patches: func_strings.patch uploaded by Gareth Palmer (license 5169) ........ Merged revisions 404674 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404675 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404676 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e4803bbd9e	Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. ... This change is in preparation for external MWI support. Removed code from the system for normal mailbox handling that appends @default to the mailbox identifier if it does not have a context. The only exception is the legacy hasvoicemail users.conf option. The legacy option will only work for app_voicemail mailboxes. The system cannot make any assumptions about the format of the mailbox identifer used by app_voicemail. chan_sip and chan_dahdi/sig_pri had the most changes because they both tried to interpret the mailbox identifier. chan_sip just stored and compared the two components. chan_dahdi actually used the box information. The ISDN MWI support configuration options had to be reworked because chan_dahdi was parsing the box@context format to get the box number. As a result the mwi_vm_boxes chan_dahdi.conf option was added and is documented in the chan_dahdi.conf.sample file. Review: https://reviewboard.asterisk.org/r/3072/ ........ Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7e9febbf86	app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state ... When doing the rework of the CDR engine that pushed all of the logic into cdr.c and made it respond to changes in channel state over Stasis, we knew that accessing the CDR engine from the dialplan would be "slightly" non-deterministic. Dialplan threads would be accessing CDRs while Stasis threads would be updating the state of said CDRs - whereas in the past, everything happened on the dialplan threads. Tests have shown that "slightly" is in reality "very". This patch synchronizes things by making the dialplan applications/functions that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to send their requests over to the CDR engine, and synchronize on the channel Stasis topic via a subscription so that they return their values/control to the dialplan at the appropriate time. While going through this, the following changes were also made: * DISA, which can reset the CDR when a user successfully authenticates, now just uses the ResetCDR app to do this. This prevents having to duplicate the same Stasis synchronization logic in that application. * Answer no longer disables CDRs. It actually didn't work anyway - calling DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer time - it just kills all CDRs on that channel, which isn't what the caller would intend. (closes issue ASTERISK-22884) (closes issue ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/ ........ Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	28c0cb28d0	channel locking: Add locking for channel snapshot creation ... Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	2a6df87fcd	func_strings: Documentation fix for QUOTE() ... Example output was inaccurate. (issue ASTERISK-22970) (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches: func_strings.patch uploaded by Gareth Palmer (license 5169) ........ Merged revisions 404081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404087 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404099 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	744556c01d	security: Inhibit execution of privilege escalating functions ... This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ce423d2ea4	func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip ... This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan to use the CHANNEL function on a chan_pjsip channel to obtain run-time information about the channel from the PJSIP channel driver and the PJSIP stack. This includes: * RTP information, including source/destination media addresses, whether or not the media is secure, held, and other properties. * RTCP information. This includes sets of parseable information, as well as individual statistic attriutes. * PJSIP information. This includes URIs, local/remote signalling addresses, whether or not the signalling is secure, and other properties. * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT function to obtain more detailed endpoint information. Review: https://reviewboard.asterisk.org/r/3038/ ........ Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f46b30bd36	func_pjsip_endpoint: Add PJSIP_ENDPOINT function for querying endpoint details ... This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint, any property configured on an endpoint. This function is a companion to the CHANNEL function, which can be used to extract the endpoint name for a channel. Review: https://reviewboard.asterisk.org/r/3035 ........ Merged revisions 403616 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1212906351	Reverting r403311. It's causing ARI tests to hang. ... ........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. ... This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	beb5cdbef5	memory leaks: Memory leak cleanup patch by Corey Farrell (first set) ... (issue ASTERSIK-22467) Reported by: Corey Farrell Patches: chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909) cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909) func_math.patch uploaded by corefarrell (license 5909) main-test-cleanup.patch uploaded by coreyfarrell (license 5909) test_dlinklists.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 401660 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401661 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401662 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4e72ad1b4f	Fix func_config list entry allocation ... The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott ........ Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400697 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400701 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	de07050d0f	Parse arguments passed to the CDR_PROP function correctly ... I can only blame this on a bad merge, because this in no way worked properly the way it was written. Mea culpa. The function should now parse its arguments correctly and function properly. (Note that the API used by the CDR_PROP function has working unit tests... this was merely bad coding of the actual registered function) (closes issue ASTERISK-22613) Reported by: Private Name ........ Merged revisions 400196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2de42c2a25	Multiple revisions 399887,400138,400178,400180-400181 ... ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	83bf017db9	Fix incorrect usages of ast_realloc(). ... There are several locations in the code base where this is done: buf = ast_realloc(buf, new_size); This is going to leak the original buf contents if the realloc fails. Review: https://reviewboard.asterisk.org/r/2832/ ........ Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398758 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398759 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	186db8fdaf	Fix some uninitialized buffers for CDR handling valgrind found. ... * Made ast_strftime_locale() ensure that the output buffer is initialized. The std library strftime() returns 0 and does not touch the buffer if it has an error. However, the function can also return 0 without an error. (closes issue ASTERISK-22412) Reported by: rmudgett ........ Merged revisions 397902 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5482fd21c8	Some CDR code optimization. ... ........ Merged revisions 397892 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4ea0acbb44	Whitespace and curly braces. ... ........ Merged revisions 397885 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	59753b1ea1	Strip down the old event system ... This removes unused code, event types, IE pltypes, and event IE types where possible and makes several functions private that were once public. This includes a renumbering of the remaining event and IE types which breaks binary compatibility with previous versions. The last remaining consumers of the old event system (or parts thereof) are main/security_events.c, res/res_security_log.c, tests/test_cel.c, tests/test_event.c, main/cel.c, and the CEL backends. Review: https://reviewboard.asterisk.org/r/2703/ (closes issue ASTERISK-22139) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c790848794	ARI: Add recording controls ... This patch implements the controls from ARI recordings. The controls are: * DELETE /recordings/live/{recordingName} - stop recording and discard it * POST /recordings/live/{recordingName}/stop - stop recording * POST /recordings/live/{recordingName}/pause - pause recording * POST /recordings/live/{recordingName}/unpause - resume recording * POST /recordings/live/{recordingName}/mute - mute recording (record silence to the file) * POST /recordings/live/{recordingName}/unmute - unmute recording. Since this underlying functionality did not already exist, is was added to app.c by a set of control frames, similar to how playback control works. The pause/mute control frames are toggles, even though the ARI controls are idempotent, to be consistent with the playback control frames. (closes issue ASTERISK-22181) Review: https://reviewboard.asterisk.org/r/2697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	07d3705b42	Check result of ast_var_assign() calls for memory allocation failure (2). ... Missed a spot in the previous commit. ........ Merged revisions 396310 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	ccdfe67bf2	Check result of ast_var_assign() calls for memory allocation failure. ... We try to keep the system running even when all available memory is spent. Review: https://reviewboard.asterisk.org/r/2734/ ........ Merged revisions 396279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f8622e7c5c	Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels. ... This commit is smaller than the initial review placed on review board. This is because a change to allow for channel drivers to access parking functionality externally was committed and invalidated quite a few of the changes initially made. (closes issue ASTERISK-22039) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2717 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	03090a88ba	Fix documentation replication issues ... This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9d8a5ceb02	Move after bridge callbacks into their own file ... One more major refactoring to go. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f3fcf0aa2e	func_channel: dtmf_features setting ... Allows reading andsetting dtmf features via a channel function CHANNEL(dtmf_features) (closes issue ASTERISK-21876) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2648/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	88e344d2df	Clean up documentation ... This patch cleans up documentation in func_channel for the following items: * rtpsource * secure_signaling * secure_media * various OOH323 parameters (closes issue ASTERISK-20969) Reported by: snuffy patches: func_chan-update.diff uploaded by snuffy (License 5024) ........ Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394981 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a3592db1fb	Clarify documentation for function PASSTHRU ... It is not apparent to the average user that the PASSTHRU function should not be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions which take a variable name and not its contents. This patch clarifies the behavior in the documentation and provides an example. (closes issue ASTERISK-21717) Reported by: Richard Miller patches: func_strings.diff uploaded by Richard Miller (license 5685) ........ Merged revisions 394302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394303 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d014ad2261	func_channel: Read/Write after_bridge_goto option ... Allows reading and setting of a channel's after_bridge_goto datastore (closes issue ASTERISK-21875) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago