asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	5f17490f4d	Fix ast_writestream leaks Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	54460c74e4	func_jitterbuffer: fix frame leaks. Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ ........ Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d51169cd36	audiohooks: Clean references to formats Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a256324fcf	manager: Unsubscribe from acl_change_sub at shutdown. ASTERISK-24453 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	294ff83152	main/bridge: Destroy features struct on off nominal path during bridge impart When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	221dcb1335	main/manager: Fix typo in AMI event documentation of "OriginateResponse" The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni ........ Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	e979d0d5c1	Documentation: Improve documentation for ExtensionStatus AMI events Review: https://reviewboard.asterisk.org/r/4085/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	70f1c82ac2	translage.c: Fix regression when generating translation path strings. Fix the AMI Status action read and write translation path strings from growing for each channel in the status event list by reseting the ast string given to ast_translate_path_to_str() to fill in the given translation path. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0e911663e3	AST-2014-011: Fix POODLE security issues There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all encryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) ........ Merged revisions 425987 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fa94bc815b	AMI: Add missing VarSet events when a channel inherits variables. There should be AMI VarSet events when channel variables are inherited by an outgoing channel. Also local;2 should generate VarSet events when it gets all of its channel variables from channel local;1. ASTERISK-24415 #close Reported by: Richard Mudgett Patches: jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/ ........ Merged revisions 425782 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fdcec1ef40	main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size' When refactoring CDRs to use the configuration framework, a 'whoops' was introduced where the CDR batch size was used when rescheduling a batch, as opposed to the time duration. This patch corrects that obvious mistake. ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged revisions 425735 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b8f505294a	config: Fix inf loop using ast_category_browse and ast_variable_retrieve Fix infinite loop when calling ast_variable_retrieve inside an ast_category_browse loop when there is more than 1 category with the same name. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4089/ ........ Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	54a5b8798f	config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG the /main/config config_basic_ops test was causing a SEGV while doing an ast_category_delete in an ast_category_browse loop. Apparently this never worked but was also never tested. I removed the test, added 2 notes to config.h indicating that it's not supported and added a few lines of code to ast_category_delete to prevent the SEGV should someone attempt it in the future. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4078/ ........ Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2d07283dc5	Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute Tasks that were marked for pending deletion in the scheduler would be moved to the cache for later reuse, but after being recycled the deleted mark wouldn't be removed resulting in fresh tasks being deleted without reason... and immediately moved back into the cache where they could be reused again. This could cause horrendous things to happen in just about anything that used a scheduler. ASTERISK-24321 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4071/ ........ Merged revisions 425503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9a7fb8ee6c	stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2) Masquerades into and out of channels that are involved in a dial operation don't create the expected dial end event. The missing dial end event goes against the model for things like CDRs and generating Dial end manager actions and such. There are four cases: 1) A channel masquerades into the caller channel. The case happens when performing a blonde transfer using the channel driver's protocol. 2) A channel masquerades into a callee channel. The case happens when performing a directed call pickup. 3) The caller channel masquerades out of dial. The case happens when using the Bridge application on the caller channel. 4) A callee channel masquerades out of dial. The case happens when using the Bridge application on a peer channel. As it turned out, all four cases need to be handled instead of just the first one. ASTERISK-24237 Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ ........ Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8bb06d592a	manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	743ad19699	bridge: During a smart bridge operation provide a more complete bridge to the old technology. When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	32624fb541	CallerID: Fix parsing regression This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d1c9621852	res_phoneprov: Refactor phoneprov to allow pluggable config providers This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	376ec31b17	cdr.c: Make turning on CDR debug a one step process instead of two. Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b32d8b5317	Stasis: Relegate log message to dev-mode This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	9e08180304	Indexer: Format message types may not exist In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424833 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	57a5e2ebee	Stasis: Only log errors for non-declined types When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	673c2febba	data: Properly access formats in capabilities structure when adding codecs. Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5c607f9735	message: Don't close an AMI connection on SendMessage action error If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6a8cb946eb	features.c: Fix lingering channel ref while Bridge() application is active. Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c092e49344	sdp_srtp: Add new lines to some WARNING messages ........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9e3b5be182	Release AMI connections on shutdown. ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cff192429b	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6a844be566	chan_pjsip: Fix deadlock when masquerading PJSIP channels. Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b67094624d	sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function When you call ast_sorcery_create() you don't necessarily know which wizard is going to be invoked. If it happens to be a wizard like 'config' that doesn't have a 'create' virtual function you get a segfault in the sorcery_wizard_create callback. This patch catches the null function pointer, does an ast_assert, and logs an error. Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	1cb36afce3	Manager: Add missing fields and documentation for CoreShowChannels This corrects some issues introduced in the responses to the CoreShowChannels AMI command as well as adding documentation for the responses. The command in Asterisk 12 was missing the following fields: Duration, Application, ApplicationData, and BridgedChannel and BridgedUniqueID (replaced with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	00207158e1	threadpool.c: Minor cleanup fixes. * Fix threadpool_alloc() prototype. * Add missing off-nominal NULL check of pool in threadpool_alloc(). * searializer_create() does not need to create the object with a lock as the lock is not used. ........ Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9d1c0348f2	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423660 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d85e59a23b	main/channel: Unlock channel in off-nominal path In r423414 (13) / r423415 (trunk), an API call that determines if a format capability structure is empty was added. This returns true if the format capability structure is completely empty or "none". A check for this was added in channel.c's set_format call. Unfortunately, when this check was true, it returned from the function while still holding the channel lock. This caused the CDR unit tests - which have a tendency to create channels with no formats - to deadlock. Whoops. This patch unlocks the channel on the off-nominal path. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	d1b1e911bf	Stasis_channels: Resolve unfinished Dials when doing masquerades Masquerades into channels that are in the dialing state don't end their dial and this goes against the model for things like CDRs and generating Dial end manager actions and such. ASTERISK-24237 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	fade256307	PJSIP: Prevent T38 framehook being put on wrong channel This change gives framehooks a reverse-direction masquerade callback in addition to chan_fixup_cb similar to the callback added to datastores to handle the same situation. The new callback provides the same parameters as the fixup callback, but is called on the new channel's framehooks before moving framehooks from the old channel to the new channel. This gives the framehooks an oppurtunity to decide whether they should remain on the new channel or be removed. This new callback is used to prevent the PJSIP T.38 framehook from remaining on a masqueraded channel if the new channel is not also a PJSIP channel. This was causing a crash when a local channel was masqueraded into a PJSIP channel and the framehook was executed on the local channel since the channel's tech private data was not structured as expected. Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7ae706b2d	utils: Create ast_strsep function that ignores separators inside quotes This function acts like strsep with three exceptions... * The separator is a single character instead of a string. * Separators inside quotes are treated literally instead of like separators. * You can elect to have leading and trailing whitespace and quotes stripped from the result and have '\' sequences unescaped. Like strsep, ast_strsep maintains no internal state and you can call it recursively using different separators on the same storage. Also like strsep, for consistent results, consecutive separators are not collapsed so you may get an empty string as a valid result. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8b0352ffae	astobj2.c/refcounter.py: Fix to deal with invalid object refs. * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	23a375be5f	Add API call to determine if format capability structure is "empty". Empty here means that there are no formats in the format_cap structure or the only format in it is the "none" format. I've added calls to check the emptiness of a format_cap in a few places in order to short-circuit operations that would otherwise be pointless as well as to prevent some assertions from being triggered in cases where channels with no formats are used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	0a2e6a1c7e	config: bug: Fix SEGV in ast_category_insert when matching category isn't found If you call ast_category_insert with a match category that doesn't exist, the list traverse runs out of 'next' categories and you get a SEGV. This patch adds check for the end-of-list condition and changes the signature to return an int for success/failure indication instead of a void. The only consumer of this function is manager and it was also changed to use the return value. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3993/ ........ Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423277 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423278 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	79b702f308	Voicemail: get correct duration when copying file to vm Changes made during format improvements resulted in the recording to voicemail option 'm' of the MixMonitor app writing a zero length duration in the msgXXXX.txt file. This change introduces a new function ast_ratestream(), which provides the sample rate of the format associated with the stream, and updates the app_voicemail function for ast_app_copy_recording_to_vm to calculate the right duration. Review: https://reviewboard.asterisk.org/r/3996/ ASTERISK-24328 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2e3f45e8e6	Realtime: Fix a bug that caused realtime destroy command to crash Also has could affect with anything that goes through ast_destroy_realtime. If a CLI user used the command 'realtime destroy <family>' with only a single column/value pair, Asterisk would crash when trying to create a variable list from a NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged revisions 422984 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	0f2bd8d855	Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. ast_play_and_record_full() has a parameter called "acceptdtmf" that is a string of acceptable DTMF digits that may be pressed by a caller to end and accept the recording. ARI uses this function in order to perform recording, and it provides options for what is passed as acceptdtmf to ast_play_and_record_full(). By default, ARI passes an empty string, with the intention that no DTMF can be used to end the recording. The problem is that ast_play_and_record_full() attempts to be "helpful" by setting "#" as the acceptdtmf if an empty string or NULL pointer has been passed in. With ARI, this results in unexpected behavior occurring if you have attempted to intercept "#" yourself in order to perform some other manipulation of the live recording. This change removes the "helpful" behavior by no longer accepting "#" as a default acceptdtmf if none is specified by the caller of ast_play_and_record_full(). This makes the ARI scenario work as expected. The other callers of ast_play_and_record_full() are app_voicemail and app_minivm, and in both cases, they pass an explicit "#" to ast_play_and_record_full() as acceptdtmf, so they are unaffected by this change. ........ Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	43c4529f15	config: bug: fix truncation of included config files on permissions error ast_config_text_file_save() currently truncates include files as they are processed. If a subsequent include file or the main config file has a permissions error that prevents writing, earlier include files are left truncated resulting in a frantic search for backups. This patch causes ast_config_text_file_save to check for write access on all files before it truncates any of them. Will be applied 1.8 > trunk. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3986/ ........ Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422903 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422904 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	128d187f38	main/cdr: Copy over location information during a fork When a CDR is forked, a new CDR is created and appended to the CDR chain for the Party A. The forked CDR starts life off as a clone of the last non-finalized for the particular Party A. In the past, merely copying over the snapshots for Party A/Party B would be sufficient. However, as the CDRs now contain cached information from Party A - specifically application/data, context, and extension - we need to copy that over during a fork as well. Huzzah for unit tests catching this when the context/extension were derived from a cached value on the CDR instead of on Party A. ........ Merged revisions 422769 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8302bc7f0a	main/rtp_engine: Format NTP timestamps as unsigned ints On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as opposed to long ints. When the RTP engine formats these as strings, it was previously formatting them as signed integers, which can result in some odd negative timestamp values (particularly on 32-bit systems). This patch formats the values as unsigned long integers. ........ Merged revisions 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0fbd9947e2	main/cdrs: Preserve context/extension when executing a Macro or GoSub The context/extension in a CDR is generally considered the destination of a call. When looking at a 2-party call CDR, users will typically be presented with the following: context exten channel dest_channel app data default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial actually takes place in a Macro, the current behaviour in 12 will result in the following CDR: context exten channel dest_channel app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The same is true of a GoSub: context exten channel dest_channel app data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This generally makes the context/exten fields less than useful. It isn't hard to preserve these values in the CDR state machine; however, we need to have something that informs us when a channel is executing a subroutine. Prior to this patch, there isn't anything that does this. This patch solves this problem by adding a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a Macro or a GoSub. The CDR engine looks for this value when updating a Party A snapshot; if the flag is present, we don't override the context/exten on the main CDR object. In a funny quirk, executing a hangup handler must not abide by this logic, as the endbeforehexten logic assumes that the user wants to see data that occurs in hangup logic, which includes those subroutines. Since those execute outside of a typical Dial operation (and will typically have their own dedicated CDR anyway), this is unlikely to cause any heartburn. Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis ........ Merged revisions 422718 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ffffc0efd8	main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios This patch fixes an issue where CDRs would get stuck generating an infinite number of CDRs, eventually crashing Asterisk (and consuming a lot of memory along the way). When a channel enters into a multi-party bridge, the CDR engine creates mappings of each participant to each other participant, picking the 'A' party as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob, Charlie, Denise), we would have something like: Alice => Bob Alice => Charlie Alice => Denise Bob => Charlie Bob => Denise Charlie => Denise This works fine when participants enter the bridge a single time. When a participant leaves a bridge, the CDRs for that channel are transitioned to a finalized state. The bug occurs if Bob rejoins. When the CDR engine creates mappings between the channels, it walks through all the participants currently in the bridge, and realizes that no one in the bridge can create a CDR with the channel (Bob). As such it creates a new CDR for the candidate and appends it to that candidate's chain. Unfortunately, on this particular code path, it doesn't stop traversing the candidate's chain. Since we just added ourselves to the chain, this causes the loop to keep going, constantly adding new CDRs. This patch makes it so the engine bails when it creates a CDR match in this case. Review: https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat ASTERISK-24208 Reported by: Frankie Chin ........ Merged revisions 422715 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c56aa2d8f6	Dial API: Add a dial option to indicate the dialed channel will replace dialer Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes. Review: https://reviewboard.asterisk.org/r/3968/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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