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Author	SHA1	Message	Date
Rodrigo Ramírez Norambuena	a24ce38e5e	cdr_adaptive_odbc: Add ability to set character for quoted identifiers. ... Added the ability to set the character to quote identifiers. This allows adding the character at the start and end of table and column names. This setting is configurable for cdr_adaptive_odbc via the quoted_identifiers in configuration file cdr_adaptive_odbc.conf. ASTERISK-25006 Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8	10 years ago
Matt Jordan	46950d6901	Merge "cdr/cdr_adaptive_odbc.c: Refactor concatenate columns name."	10 years ago
Joshua Colp	84a30a4ad3	Merge "Check for ao2_alloc failure in __ast_channel_internal_alloc."	10 years ago
Joshua Colp	bfdc766bf6	Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs."	10 years ago
Corey Farrell	5757d2d30d	Check for ao2_alloc failure in __ast_channel_internal_alloc. ... Fix a crash that could occur in __ast_channel_internal_alloc if ao2_alloc fails. ASTERISK-24991 #close Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90	10 years ago
Mark Michelson	6331be0638	res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. ... When SUBSCRIBE dialogs were established, we never associated the endpoint that created the subscription with the dialog we end up creating. In most cases, this ended up not causing any problems. The actual bug that was observed was that when a device that was behind NAT established a subscription with Asterisk, Asterisk would end up sending in-dialog NOTIFY requests to the device's private IP addres instead of the public address of the NAT router. When Asterisk receives the initial SUBSCRIBE from the device, res_pjsip_nat rewrites the contact to the public address on which the SUBSCRIBE was received. This allows for the dialog to have its target address set to the proper public address. Asterisk then would send a 200 OK response to the SUBSCRIBE, then a NOTIFY with the initial subscription state. The device would then send a 200 OK response to Asterisk's NOTIFY. Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat did not rewrite the address in the Contact header. Then, when the PJSIP dialog layer processed the 200 OK, PJSIP would perform a comparison between the IP address in the Contact header and its saved target address for the dialog. Since they differed, PJSIP would update the target dialog address to be the address in the Contact header. From this point, if Asterisk needed to send a NOTIFY to the device, the result was that the NOTIFY would be sent to the private address that the device placed in the Contact header. The reason why res_pjsip_nat did not rewrite the address when it received the 200 OK response was that it could not associate the incoming response with a configured endpoint. This is because on a response, the only way to associate the response to an endpoint is by finding the dialog that the response is associated with and then finding the endpoint that is associated with that dialog. We do not perform endpoint lookups on responses. res_pjsip_pubsub skipped the step of associating the endpoint with the dialog we created, so res_pjsip_nat could not find the associated endpoint and therefore couldn't rewrite the contact. This commit message is like 50x longer than the actual fix. ASTERISK 24981 #close Reported by Mark Michelson Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd	10 years ago
Richard Mudgett	614f506690	chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. ... The chan_dahdi channel driver is a very old driver. The ability for it to support ISDN was added well after the initial analog support. Setting the softhangup flags is a carry over from the original analog code. The driver was not updated to call ast_queue_hangup() which will post the AMI HangupRequest event. * Changed sig_pri.c to call ast_queue_hangup() instead of setting the softhangup flag when the remote party initiates a hangup. ASTERISK-24895 #close Reported by: Andrew Zherdin Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325	10 years ago
Joshua Colp	b1deedf0dc	Merge "pjsip_options: Fix non-qualified contacts showing as unavailable"	10 years ago
Rodrigo Ramírez Norambuena	bff3064578	cdr/cdr_adaptive_odbc.c: Refactor concatenate columns name. ... The concatenate for columns name to INSERT INTO is always the same. It is possible to do it on one line. ASTERISK-24980 Change-Id: Ib8bb53c42535378581d4ef729cc5ebbb22b067ac	10 years ago
George Joseph	06ba1e59cb	pjsip_options: Fix format specifier for int64_t rtt. ... Contact status rtt is an int64_t and needs the PRId64 macro to properly create the format specifier on 32-bit systems. Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7	10 years ago
Matt Jordan	b94d70f42b	Merge "main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple"	10 years ago
Joshua Colp	1d5854f5f4	Merge "Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled."	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
Matt Jordan	8e903b17ea	main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple ... When a PBX registrar is unloaded, it will fail to remove its extension from the context root_table if a dialplan application used by that extension is still loaded. This can be the case for AGI, which can be unloaded after several of the standard PBX providers. Often, this is harmless; however, if the extension's priorities are removed during the failed unloading *and* the dialplan application later unregisters, it leaves a ticking timebomb for the next PBX provider that attempts to iterate over the extensions. When that occurs, the peer_table pointer on the extension will already be set to NULL. The current code does not check to see if the pointer is NULL before passing it to a hashtab function this is not NULL tolerant. Since it is possible for the peer_table to be NULL when we normally would not expect that to be the case, the solution in this patch is to simply skip over processing an extension's priorities if peer_table is NULL. Prior to this patch, the tests/pbx/callerid_match test would crash during module unload. With this patch, the test no longer crashes after running. ASTERISK-24774 #close Reported by: Corey Farrell Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40	10 years ago
Richard Mudgett	1269dd06bc	res_fax: Fix latent bug exposed by ASTERISK-24841 changes. ... Three fax related tests started failing as a result of changes made for ASTERISK-24841: tests/fax/pjsip/gateway_t38_g711 tests/fax/sip/gateway_mix1 tests/fax/sip/gateway_mix3 Historically, ast_channel_make_compatible() did nothing if the channels were already "compatible" even if they had a sub-optimal translation path already setup. With the changes from ASTERISK-24841 this is no longer true in order to allow the best translation paths to always be picked. In res_fax.c:fax_gateway_framehook() code manually setup the channels to go through slin and then called ast_channel_make_compatible(). With the previous version of ast_channel_make_compatible() this was always a no-operation. * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() that now undoes what was just setup when the framehook is attached. * Fixed locking around saving the channel formats in fax_gateway_framehook() to ensure that the formats that are saved are consistent. * Fix copy pasta errors in fax_gateway_framehook() that confuses read and write when dealing with saved channel formats. ASTERISK-24841 Reported by: Matt Jordan Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d	10 years ago
Corey Farrell	c1d44ff043	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	8435a0cdff	Merge "Detect potential forwarding loops based on count."	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	bb347fa594	Merge topic 'ASTERISK-24863' ... * changes: res_pjsip: Add global option to limit the maximum time for initial qualifies pjsip_options: Add qualify_timeout processing and eventing res_pjsip: Refactor endpt_send_request to include transaction timeout	10 years ago
Joshua Colp	fd823252e3	Merge "res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced"	10 years ago
Kevin Harwell	56a2baa21d	bridge.c: NULL app causes crash during attended transfer ... Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: https://gerrit.asterisk.org/#/c/133/ Change-Id: I5134b84c4524906d8148817719d76ffb306488ac	10 years ago
George Joseph	c6ed681638	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
Scott Griepentrog	664d3263e4	res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced ... This change makes the send_notify of the sub_tree not happen when the sub_tree has been deleted due to the notify call failing, which avoids a crash. ASTERISK-24970 #close Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf	10 years ago
George Joseph	51886c68dc	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	70c3759c40	Merge "res_pjsip: Add external PJSIP resolver implementation using core DNS API."	10 years ago
George Joseph	ab6382cafd	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
Matt Jordan	ab38353892	Merge "Build System: Enable use of ~/.asterisk.makeopts and /etc/asterisk.makeopts."	10 years ago
George Joseph	043c38f6de	More .gitignore updates ... Added .pyc and .sha1 to the top-level .gitignore. Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Corey Farrell	abf10a1d4c	Build System: Enable use of ~/.asterisk.makeopts and /etc/asterisk.makeopts. ... The Makefile claims that you can set default menuselect options by creating ~/.asterisk.makeopts or /etc/asterisk.makeopts, but they are never read. The rule for menuselect.makeopts is only allowed to run if the active target is 'menuselect', but the menuselect target doesn't depend on menuselect.makeopts. A dot (wildcard character) was added so the rule will be active for the targets that cause it to run: nmenuselect, cmenuselect, and gmenuselect. ASTERISK-13271 #close Reported by: John Nemeth Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2	10 years ago
Joshua Colp	a3cec44a0a	res_pjsip: Add external PJSIP resolver implementation using core DNS API. ... This change adds the following: 1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked. 2. Unit tests for the query set implementation. 3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups. For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A, with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit transport has been provided. Configured transports on the system are taken into account to eliminate resolved addresses which have no hope of completing. ASTERISK-24947 #close Reported by: Joshua Colp Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e	10 years ago
Matt Jordan	60d1911482	Merge "cel_pgsql: Fix name string for log on unable allocate memory."	10 years ago
Rodrigo Ramírez Norambuena	33a319ae73	cel_pgsql: Fix name string for log on unable allocate memory. ... The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and LENGTHEN_BUF2. ASTERISK-24965 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744	10 years ago
Corey Farrell	f89481e39c	test_astobj2_weaken: Fix source file registration. ... Update test_astobj2_weaken to use the new AST_REGISTER_FILE macro. Change-Id: Ieedadf16610f2e042f393e0501a36447cd07f83d	10 years ago
Matt Jordan	83b0dcadaa	Merge "Build System: Create Makefile macro MOD_ADD_SOURCE."	10 years ago
Matt Jordan	7687188a2d	Merge "astobj2: Add support for weakproxy objects."	10 years ago
Matt Jordan	941512e588	Merge ".gitignore updates for master/13"	10 years ago
Corey Farrell	62508d6891	Build System: Create Makefile macro MOD_ADD_SOURCE. ... This new macro allows a single line to add all additional sources to a module. This helps prevent modules from missing steps, and makes future changes easier since they can be made in a single place. ASTERISK-24960 #close Reported by: Corey Farrell Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b	10 years ago
Matt Jordan	9f5ba58332	Merge "cdr_pgsql: Fix CLI "cdr show pgsql status" command."	10 years ago
Rodrigo Ramírez Norambuena	23a180cade	cdr_pgsql: Fix CLI "cdr show pgsql status" command. ... The command always showed the usage information. * Fix the error in command validation for CLI_SHOWUSAGE. ASTERISK-24959 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5	10 years ago
George Joseph	bf46ef35ca	.gitignore updates for master/13 ... Added products of ./bootstrap Added nmenuselect and gmenuselect to menuselect/ Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35	10 years ago
Corey Farrell	62e95065d6	AMI: Fix improper handling of lines that are exactly 1025 bytes long. ... When AMI receives a line that is 1025 bytes long, it sends two error messages. Copy the last byte in the buffer to the first postiion, set the length to 1. ASTERISK-20524 #close Reported by: David M. Lee Change-Id: Ifda403e2713b59582c715229814fd64a0733c5ea	10 years ago
Corey Farrell	cb6bf3094e	astobj2: Add support for weakproxy objects. ... This implements "weak" references. The weakproxy object is a real ao2 with normal reference counting of its own. When a weakproxy is pointed to a normal object they hold references to each other. The normal object is automatically freed when a single reference remains (the weakproxy). The weakproxy also supports subscriptions that will notify callbacks when it does not point to any real object. ASTERISK-24936 #close Reported by: Corey Farrell Change-Id: Ib9f73c02262488d314d9d9d62f58165b9ec43c67	10 years ago
David M. Lee	a573b77f78	Fixing extconf compile ... During the mass code deletion for clang support, a stray backslash was left behind that was causing utils to fail to compile. Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1	10 years ago
Matt Jordan	974d44fd89	Merge "build_tools/make_version: Update version parsing for Git migration"	10 years ago
Joshua Colp	755563f0f3	Merge "git migration: Refactor the ASTERISK_FILE_VERSION macro"	10 years ago
Matt Jordan	3f9aa29945	build_tools/make_version: Update version parsing for Git migration ... External systems - such as the Asterisk Test Suite - require knowledge of the upstream branch. Unfortunately, after moving to Git, the Asterisk version currently consists of only a 'GIT" prefix followed by an object blob, e.g., GIT-as08d7. This makes it difficult for such systems to know what features are available in a particular check out of Asterisk. This patch fixes this by hardcoding the branch in a variable in the make_version script. Since the mainline branches are not changed often - typically only once a year - this is a reasonable approach to solving the problem, and is more reliable than parsing the output of 'git branch -vv'. Branches that track off of an upstream primary branch will then get the benefit of knowing which mainline branch they are currently based off of. ASTERISK-24954 #close Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799	10 years ago
Corey Farrell	fbc8ddfe63	Optional API: Fix handling of sources that are both provider and user. ... OPTIONAL_API has conditionals to define AST_OPTIONAL_API and AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined. Unfortunately this is inside the include protection block, so only the first status of AST_API_MODULE is respected. For example res_monitor is an optional API provider, but uses func_periodic_hook. This makes func_periodic_hook non-optional to res_monitor. This changes optional_api.h so that AST_OPTIONAL_API and AST_OPTIONAL_API_ATTR is redefined every time the header is included. ASTERISK-17608 #close Reported by: Warren Selby Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679	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
Matt Jordan	1174ef588d	Merge "main/editline: Add .gitignore."	10 years ago
Corey Farrell	5d34bce635	main/editline: Add .gitignore. ... This patch adds a .gitignore for main/editline to ignore all build results. Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d	10 years ago