asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	522593f901	Add the missing part of r400140 When the patch to add retry-on-forbidden-response was committed, part of the patch for chan_sip was not committed which caused the feature to be entirely nonfunctional. This corrects the code in question. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874 ........ Merged revisions 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405081 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	34c595beb9	res_pjsip_acl: Fix another case of assuming a contact will always contain a URI. ........ Merged revisions 405034 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f89ab79862	res_pjsip_nat: Don't assume a Contact header will always contain a URI. If the 'rewrite_contact' option was enabled and a Contact header was received which contained a '' a crash would occur. This change makes the res_pjsip_nat module ignore the Contact header if it contains only a ''. (closes issue ASTERISK-23101) Reported by: Matt Jordan ........ Merged revisions 405019 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e95e66cede	app_voicemail: Explicitly set defaultenabled=yes ........ Merged revisions 405006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	61f215250e	External MWI AMI support. The external MWI AMI interface provides a thin wrapper around the core external MWI resource. The resource adds the following AMI actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404954 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9fa171e547	External MWI core support. * The core external MWI resource provides for MWI message counts persistence using sorcery. With sorcery, the user is able to configure which sorcery wizzard backend to use if the default astdb is not desired. * The core external MWI resoruce provides some debugging CLI commands enabled by defining MWI_DEBUG_CLI. The debugging CLI commands are: "mwi delete all", "mwi delete like <regex>", "mwi delete mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new> [<old>]]". (closes issue AFS-43) Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged revisions 404952 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	986c9e897d	res_pjsip_outbound_registration: Don't assume that a registration client will always exist. ........ Merged revisions 404935 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	11f18e4724	res_pjsip_outbound_registration: Create registration client in pj thread. Depending on which threading was loading the outbound registration it was possible for the registration client to be allocated outside of a pj thread. This change moves the creation inside the synchronous task where it is guaranteed it will occur in a pj thread. Reported by: Rob Thomas ........ Merged revisions 404923 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	3eee3f21ad	asterisk.c: suppress live_dangerously warning on rasterisk Even since the fixes of AST-2013-007, Asterisk prints the following warning on startup if the user decided to live dangerously: Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This message is intended for the logs and interactive startup. No need for it to appear on a remote console. This commit removes it from there. (closes issue ASTERISK-23084) Review: https://reviewboard.asterisk.org/r/3101/ ........ Merged revisions 404861 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404888 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404911 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	2f62394fb5	cel_pgsql: module not correctly reloading Upon reload the module unconditionally "unloaded" the module (freeing memory and setting pointers to NULL) and then when attempting a "load" if the config file had not changed then nothing would be reinitialized. By moving the "unload" to occur conditionally (reload only) after an attempted configuration load, but before module "loading" alleviates the issue. The module now loads/unloads/reloads correctly. (closes issue ASTERISK-22871) Reported by: Matteo ........ Merged revisions 404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404858 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404859 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f8b55f16d2	res_pjsip_logger: Add the ASTERISK_FILE_VERSION macro Registering yourself with the Asterisk core is the nice thing to do, even when you're a logging module. ........ Merged revisions 404855 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c6df713da7	res_pjsip_authenticator_digest: Fix md5 hash buffer An md5 hash is 32 bytes long. The char buffer must be at least 33 bytes to avoid clobbering of the stack. This patch also fixes a potential clobbering in test_utils.c. Thanks to Andrew Nagy for reporting and testing this out in #asterisk-dev Reported by: Andrew Nagy Tested by: Andrew Nagy ........ Merged revisions 404843 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	bd4be333fb	manager: UserEvent including action on output AMI action UserEvent event response would include the action header in its keyvalue pairs list. Adjusted the start of the header loop to skip over the action part. (closes issue ASTERISK-22899) Reported by: outtolunc Patches: svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license 5198) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	1ee4870a81	chan_dahdi: dahdi show channels slices PRI channel dnid on output dahdi show channels output slices the callerid (which is dnid copied over on PRI channels). If the channel naming structures look like: 'DAHDI/i1/1408409XXXX-6' then the output slices 1408409XXXX down to 1408409XXX. This patch just opens it up to 15 chars so you can see the whole thing. (closes issue ASTERISK-22918) Reported by: outtolunc Patches: svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198) ........ Merged revisions 404784 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404785 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404786 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c2caaecc45	test_stasis.c: Fix ref leak in normal execution path. ........ Merged revisions 404764 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	0db94b70b9	app_meetme: compiler warning Fixed a compiler warning (errors in 'dev-mode') given by gcc version 4.8.1. The one in app_meetme involved the 'sizeof-pointer-memaccess' (see: http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it would no longer issue a warning and can compile again in 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/ ........ Merged revisions 404742 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404773 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0ae89e7b7e	res_pjsip: Ensure more URI validation happens in pj threads. ........ Merged revisions 404737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f0f23655c9	res_pjsip_outbound_registration: Ensure URI validation happens in a pjlib thread. This change moves outbound registration URI validation into the task executed within a pjlib thread. Reported by: Andrew Nagy ........ Merged revisions 404725 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404726 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
Kevin Harwell	821ab51381	res_pjsip: add 'set_var' support on endpoints Added a new 'set_var' option for ast_sip_endpoint(s). For each variable specified that variable gets set upon creation of a pjsip channel involving the endpoint. (closes issue ASTERISK-22868) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/3095/ ........ Merged revisions 404663 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f720a9ac89	chan_pjsip: Handle hanging up before calling. Channel creation in Asterisk is broken up into two steps: requesting and calling. In some cases a channel may be requested but never called. This happens in the ChanIsAvail dialplan application for determining if something is reachable or not. The PJSIP channel driver did not take this situation into account and attempted to end a session that was never called out on. The code now checks the session state to determine if the session has been called out on and if not terminates it instead of ending it. (closes issue ASTERISK-23074) Reported by: Kilburn ........ Merged revisions 404652 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e583dca9dd	res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match' field. Hostnames specified in the 'match' field will be resolved and all addresses returned. Each address will be added to the endpoint identifier for the matching process. Reported by: Rob Thomas ........ Merged revisions 404613 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	bb30b22419	cel_pgsql: deadlock on unload and core_event_dispatcher A deadlock can happen between a thread unloading or reloading the cel_pgsql module and the core_event_dispatcher taskprocessor thread. Description of what is happening: Thread 1 (for example, a netconsole thread): a "module reload cel_pgsql" is launched the thread enter the "my_unload_module" function (cel_pgsql.c) the thread acquire the write lock on psql_columns the thread enter the "ast_event_unsubscribe" function (event.c) the thread try to acquire the write lock on ast_event_subs[sub->type] Thread 2 (core_event_dispatcher taskprocessor thread): the taskprocessor pop a CEL event the thread enter the "handle_event" function (event.c) the thread acquire the read lock on ast_event_subs[sub->type] the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events the thread try to acquire a read lock on psql_columns (closes issue ASTERISK-22854) Reported by: Etienne Lessard Patches: cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license 6394) ........ Merged revisions 404603 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404604 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404605 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	135b9d3562	res_pjsip_outbound_registration: Add validation for 'server_uri' and 'client_uri'. When applying configuration for outbound registrations the 'server_uri' and 'client_uri' fields were not validated. The code will now confirm that they exist and that they contain parseable SIP URIs. Reported by: Andrew Nagy ........ Merged revisions 404592 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	20d02c1288	channels.c: core show channeltypes slicing 'core show channeltypes' type column is being sliced, resulting in incomplete type names. (closes issue ASTERISK-22919) Reported by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded by outtolunc (license 5198) ........ Merged revisions 404579 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404581 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404582 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	33ec6f9d03	Added note to UPGRADE.txt about the default value of live_dangerously changing ........ Merged revisions 404568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a952abb9da	http: Properly reject requests with Transfer-Encoding set Asterisk does not support any of the transfer encodings specified in HTTP/1.1, other than the default "identity" encoding. According to RFC 2616: A server which receives an entity-body with a transfer-coding it does not understand SHOULD return 501 (Unimplemented), and close the connection. A server MUST NOT send transfer-codings to an HTTP/1.0 client. This patch adds the 501 Unimplemented response, instead of the hard work of actually implementing other recordings. This behavior is especially problematic for Node.js clients, which use chunked encoding by default. (closes issue ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/ ........ Merged revisions 404565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c3d5c41dae	res_pjsip_pubsub: Ensure dialog manipulation happens on proper thread. When destroying a subscription we remove the serializer from its dialog and decrease its reference count. Depending on which thread dropped the subscription reference count to 0 it was possible for this to occur in a thread where it is not possible. (closes issue ASTERISK-22952) Reported by: Matt Jordan ........ Merged revisions 404553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	82eb03b915	chan_dahdi: enable ignore_failed_channels by default If ignore_failed_channels is set to "true" for a channel, the channel will continue to be configured even if configuring it has failed. This allows Asterisk to start before all the DAHDI initialization is done and thus not force the starting order dahdi -> asterisk. Review: https://reviewboard.asterisk.org/r/3063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5a15803a1b	res_pjsip/pjsip_cli: fix compilation error caused by passing ast_free When wanting to pass *free as a function pointer, ast_free_ptr has to be used instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled. ........ Merged revisions 404531 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	40a7f68e4b	ari: Remove support for specifying channel vars during origination. When we added support for specifying channel variables for an origination, we didn't consider how that would interact with another feature, namely specifying request parameters in a JSON request body. The method of specifying channel variables (as a flat JSON object passed in the JSON body) interferes with parsing parameters out of the request body. Unfortunately, fixing this would be a backward incompatible change. In the interest of keeping the API sane and keeping our release schedule, we're dropping the feature for specifying channel variables in the origination request. We will bring the feature back soon, as a backward compatible addition to the API. (closes issue ASTERISK-23051) Review: https://reviewboard.asterisk.org/r/3088 ........ Merged revisions 404509 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	bd0ef25080	Remove automerge properties ........ Merged revisions 404488 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b172d369c4	res_pjsip: Add PJSIP CLI commands Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	a0c288bb23	say.c: correct time for polish In ast_say_date_with_format_pl(), change ast_say_number() to use tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported by: Robert Mordec Review: https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch uploaded by veilen (license 6555) ........ Merged revisions 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404457 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404458 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	344cdab3a7	Fix issue where PJSIP blind transferer dialog may not complete as planned. When transferring to a dialplan extension that will not place any outbound calls, the only control frames that the PJSIP REFER framehook will receive are inconsequential (such as unhold or srcchange). As such, we shouldn't allow for the reception of those types of frames prevent us from signaling to the transferring party that the transfer has completed successfully once voice frames are read. Thanks to Jonathan Rose for pointing this out. ........ Merged revisions 404439 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7ddfe12aff	res_stasis_device_state: Set resource type for subscriptions to deviceState The documentation for ARI already specifies that the device state resource when used for subscribing for events is "deviceState", not "device_state". The code, however, used "device_state"; although this was inconsistent as well in doxygen comments in resource_applications. Because the actual resource being subscribed to is /deviceStates/{device}/, it makes sense for the resource type specifier to be deviceState. Note that the key value in the events is still "device_state". ........ Merged revisions 404437 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	72c282cc66	ao2_iterator: Mini-audit of the ao2_iterator loops in the new code files. * Fixed several places where ao2_iterator_destroy() was not called. * Fixed several iterator loop object variable reference problems. * Fixed res_parking AMI actions returning non-zero. Only the AMI logoff action can return non-zero. Review: https://reviewboard.asterisk.org/r/3087/ ........ Merged revisions 404434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c2fd2ac823	manager: bump version to 2.0.0 AMI has received substantial updates over the past year. Not only has the syntax been vastly improved and made consistent (which entails many event changes), but the underlying things that those events convey have changed substantially as well. After some conversation in #asterisk-dev, it was agreed that this is a good time to jump to 2. At the same time, since ARI will most likely use semantic versioning, we might as well use that for AMI as well. That also affords us greater meaning for the AMI version. ........ Merged revisions 404421 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9e4f80a4f6	Whitespace fixes. ........ Merged revisions 404419 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	06b577f7dc	Documentation: Updates for info about NAT-related settings and fixes for pjsip.conf.sample Added another NAT example to pjsip.conf.sample. We had a few mentions of NAT configuration throughout the sample, but I added another for a little bit more clarity. Additionally many pjsip options were affected by the change to snake case, so I fixed any instances of those options in pjsip.conf. I regenerated the config option list (at the bottom of the file) from a new xml config doc dump, so all the snake case changes should be reflected there, as well as any other changes to those options. (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/ ........ Merged revisions 404405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404406 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	efd7c40712	security_events: log events with descriptive names This patch updates the log messages to include descriptive names for event types. This is an improvement over having only cryptic type numbers. (closes issue ASTERISK-22909) Reported by: outtolunc Review: https://reviewboard.asterisk.org/r/3081/ Patches: svn_security_events.c.names.diff.txt uploaded by outtolunc (license 5198) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	dbead14c3b	Put notice in CHANGES as well as UPGRADE.txt. ........ Merged revisions 404375 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	433c6f010f	res_pjsip: Ignore 401/407 responses for transactions and dialogs we don't know about. Under normal conditions it is unlikely we will ever receive a response for a transaction or dialog we don't know about but if any are received ignore them. ........ Merged revisions 404371 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	8402cd4cd9	res_pjsip_session: Fix SDP negotiation when resending an INVITE with authentication. The process for resending an INVITE with authentication involves restarting the UAC session. We were incorrectly passing in that a new offer is being sent, causing the SDP negotiation to get into a (technically speaking) funky state. ........ Merged revisions 404369 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1b91ee6c4b	Fix a deadlock that occurred due to a conflict of masquerades. For the explanation, here is a copy-paste of the review board explanation: Initially, it was discovered that performing an attended transfer of a multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread started a masquerade and reached the point where it was calling the fixup() callback on the "original" channel. For chan_pjsip, this involves pushing a synchronous task to the session's serializer. The problem was that a task ahead of the fixup task was also attempting to perform a channel masquerade. However, since masquerades are designed in a way to only allow for one to occur at a time, the task ahead of the fixup could not continue until the masquerade already in progress had completed. And of course, the masquerade in progress could not complete until the task ahead of the fixup task had completed. Deadlock. The initial fix was to change the fixup task to be asynchronous. While this prevented the deadlock from occurring, it had the frightful side effect of potentially allowing for tasks in the session's serializer to operate on a zombie channel. Taking a step back from this particular deadlock, it became clear that the problem was not really this one particular issue but that masquerades themselves needed to be addressed. A PJSIP attended transfer operation calls ast_channel_move(), which attempts to both set up and execute a masquerade. The problem was that after it had set up the masquerade, the PBX thread had swooped in and tried to actually perform the masquerade. Looking at changes that had been made to Asterisk 12, it became clear that there never is any time now that anyone ever wants to set up a masquerade and allow for the channel thread to actually perform the masquerade. Everyone always is calling ast_channel_move(), performs the masquerade itself before returning. In this patch, I have removed all blocks of code from channel.c that will attempt to perform a masquerade if ast_channel_masq() returns true. Now, there is no distinction between setting up a masquerade and performing the masquerade. It is one operation. The only remaining checks for ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not want to interrupt a masquerade by hanging up the channel. Instead, now ast_hangup() will wait for a masquerade to complete before moving forward with its operation. The ast_channel_move() function has been modified to basically in-line the logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has been killed off for real. ast_channel_move() now has a lock associated with it that is used to prevent any simultaneous moves from occurring at once. This means there is no need to make sure that ast_channel_masq() or ast_channel_masqr() are already set on a channel when ast_channel_move() is called. It also means the channel container lock is not pulling double duty by both keeping the container locked and preventing multiple masquerades from occurring simultaneously. The ast_do_masquerade() function has been renamed to do_channel_masquerade() and is now internal to channel.c. The function now takes explicit arguments of which channels are involved in the masquerade instead of a single channel. While it probably is possible to do some further refactoring of this method, I feel that I would be treading dangerously. Instead, all I did was change some comments that no longer are true after this changeset. The other more minor change introduced in this patch is to res_pjsip.c to make ast_sip_push_task_synchronous() run the task in-place if we are already a SIP servant thread. This is related to this patch because even when we isolate the channel masquerade to only running in the SIP servant thread, we would still deadlock when the fixup() callback is reached since we would essentially be waiting forever for ourselves to finish before actually running the fixup. This makes it so the fixup is run without having to push a task into a serializer at all. (closes issue ASTERISK-22936) Reported by Jonathan Rose Review: https://reviewboard.asterisk.org/r/3069 ........ Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3ccd5dee18	udptl: Dead code elimination. ast_udptl_bridge was not used. Removing dead code starting with ast_udptl_bridge() eliminated the code in this change. Note: This code has actually been dead since Asterisk v1.4 when it was first put in. Review: https://reviewboard.asterisk.org/r/3079/ ........ Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	c061f634da	res_fax.c: crash on framehook with no dsp in fax detect In fax_detect_framehook() a null pointer reference can occur where a voice frame is processed but no dsp is attached to the fax detection structure. The code block that rejects frames that detection cannot be processed on is checking for dsp but falls through when it should instead return, as this change implements. (closes issue ASTERISK-22942) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged revisions 404351 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404352 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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
Scott Griepentrog	2882c5f9f1	astdb: crash in sqlite3 during shutdown When Asterisk is shut down, the astdb_atexit() function releases (finalize) the previously initiated (prepared) SQL statements in sqlite3. Another thread making a subsequent request can cause a crash in sqlite3. This patch eliminates that issue by resetting the statement pointer after it is released/cleared. The sqlite3 code detects the null pointer, and aborts the operation cleanly. (closes issue AST-1265) Reported by: Alexander Hömig (closes issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged revisions 404344 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404345 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	eb235ad05f	channel: Add a missing ast_channel_unlock when allocating a Surrogate channel. ........ Merged revisions 404332 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

1 2 3 4 5 ...

24996 Commits (522593f901baff3f1ba334ecaad9eefd6a9c9316) All Branches Search

24996 Commits (522593f901baff3f1ba334ecaad9eefd6a9c9316)

All Branches