3049 Commits (ea6f32fa18d85e4cbe7f1affef2bdc03bd997aae)

Author	SHA1	Message	Date
George Joseph	370c82acca	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@425713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6fc3f0e413	PJSIP: Enforce module load dependencies ... This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@425690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	ca474c25e4	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@425525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d0566ed1c6	res_phoneprov: Create accessor for ast_phoneprov_std_variable_lookup ... Based on feedback from Richard, I created an accessor for res_phoneprov/ast_phoneprov_std_variable_lookup and added load priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@425480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	282bb06177	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@425383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	1d06ead110	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@424963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4869e033fb	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@424471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	8a8a95c067	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@423503 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	db6b5c24cd	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@423476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	8d10e6db68	res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies. ... res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE arrives. * It checks that there is a subscription handler for the Event * It checks that there are body generators for the types in the Accept header The problem is, there's nothing that ensures that these two things will actually mesh with each other. For instance, Asterisk will accept a subscription to MWI that accepts pidf+xml bodies. That doesn't make sense. With this commit, we add some type information to the mix. Subscription handlers state they generate data of type X, and body generators state that they consume data of type X. This way, Asterisk doesn't end up in some hilariously mismatched situation like the one in the previous paragraph. ASTERISK-24136 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3877 Review: https://reviewboard.asterisk.org/r/3878 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@423344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b436f387cc	res_pjsip: ami: Fix error in AMI output when an endpoint has no transport ... When no transport is associated to an endpoint, the AMI output for PJSIPShowEndpoint indicates an error instead of silently ignoring the missing transport. This patch causes the error to appear only if a transport was specified on the endpoint and the transport doesn't exist. It also fixes an issue with counting the objects that were actually found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3998/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@423282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a163335e88	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@423278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	06e083f9dd	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@422718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	11ad4ba3e1	Resolve race condition where channels enter dialplan application before media has been negotiated. ... Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@422536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	bd96feb6ff	CallerID: Fix parsing of malformed callerid ... This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@422114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	940e45f9cb	ARI: Fix a crash caused by hanging during playback to a channel in a bridge ... ASTERISK-24147 #close Reported by: Edvin Vidmar Review: https://reviewboard.asterisk.org/r/3908/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@421879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4abda9fb74	Stasis: Add information to blind transfer event ... When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@421537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	63987831aa	Bridges: Fix feature interruption/unintended kick caused by external actions ... If a manager or CLI user attached a mixmonitor to a call running a dynamic bridge feature while in a bridge, the feature would be interrupted and the channel would be forcibly kicked out of the bridge (usually ending the call during a simple 1 to 1 call). This would also occur during any similar action that could set the unbridge soft hangup flag, so the fix for this was to remove unbridge from the soft hangup flags and make it a separate thing all together. ASTERISK-24027 #close Reported by: mjordan Review: https://reviewboard.asterisk.org/r/3900/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bf6ff88a20	AMI/ARI: Update version to 2.5.0/1.5.0 respectively ... This is to support the backwards compatible changes made in the next version of Asterisk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	18375d3f57	Stasis: Allow internal channels directly into bridges ... The patch to catch channels being shoehorned into Stasis() via external mechanisms also happens to catch Announcer and Recorder channels because they aren't known to be stasis-controlled channels in the usual sense. This marks those channels as Stasis()-internal channels and allows them directly into bridges. Review: https://reviewboard.asterisk.org/r/3903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	23c7dff74e	Revert previous patch since it had some unreviewed content in it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	ae880e67dd	Ensure bridges actually exist when trying to determine the bridged peer. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e0a94cc164	Stasis: Convey transfer information to applications ... This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a48e144492	ARI: Add channel technology agnostic out of call text messaging ... This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@420089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	60a67b96f5	ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech ... This patch serves two purposes: (1) It fixes some bugs with endpoint subscriptions not reporting all of the channel events (2) It serves as the preliminary work needed for ASTERISK-23692, which allows for sending/receiving arbitrary out of call text messages through ARI in a technology agnostic fashion. The messaging functionality described on ASTERISK-23692 requires two things: (1) The ability to send/receive messages associated with an endpoint. This is relatively straight forwards with the endpoint core in Asterisk now. (2) The ability to send/receive messages associated with a technology and an arbitrary technology defined URI. This is less straight forward, as endpoints are formed from a tech + resource pair. We don't have a mechanism to note that a technology that *may* have endpoints exists. This patch provides such a mechanism, and fixes a few bugs along the way. The first major bug this patch fixes is the forwarding of channel messages to their respective endpoints. Prior to this patch, there were two problems: (1) Channel caching messages weren't forwarded. Thus, the endpoints missed most of the interesting bits (such as channel creation, destruction, state changes, etc.) (2) Channels weren't associated with their endpoint until after creation. This resulted in endpoints missing the channel creation message, which limited the usefulness of the subscription in the first place (a major use case being 'tell me when this endpoint has a channel'). Unfortunately, this meant another parameter to ast_channel_alloc. Since not all channel technologies support an ast_endpoint, this patch makes such a call optional and opts for a new function, ast_channel_alloc_with_endpoint. When endpoints are created, they will implicitly create a technology endpoint for their technology (if one does not already exist). A technology endpoint is special in that it has no state, cannot have channels created for it, cannot be created explicitly, and cannot be destroyed except on shutdown. It does, however, have all messages from other endpoints in its technology forwarded to it. Combined with the bug fixes, we now have Stasis messages being properly forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar), where bar has a single channel associated with it and foo has two channels associated with it. The messages would be forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the applications resource, can: - subscribe to endpoint:PJSIP/foo and get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - subscribe to endpoint:PJSIP and get notifications for channels PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, it never has events itself. It merely provides an aggregation point for all other endpoints in its technology (which in turn aggregate all channel messages associated with that endpoint). This patch also adds endpoints to res_xmpp and chan_motif, because the actual messaging work will need it (messaging without XMPP is just sad). Review: https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@419196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	29ca8a873b	ari: Add a copy operation for stored recordings ... This patch adds a new operation for stored recordings, copy. It takes an existing stored recording and makes a copy of it in the same directory or a relative directory under the stored recording directory. /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name} This is particularly useful for voicemail-esque applications, which may need to copy or move recordings around a directory structure. Review: https://reviewboard.asterisk.org/r/3768/ ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam Galarneau git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@419021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9ef593f706	Channels: Masquerades to automatically move frame/audio hooks ... Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4895ddef28	res_pjsip: Support setting a default accountcode on endpoints ... Most channel drivers let you specify a default accountcode to be set on channels associated with a particular peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip did not support such a setting. This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'. When a channel is created that is associated with an endpoint with this value set, the channel will automatically have its accountcode property set to the value configured for the endpoint. Review: https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	33e25f535f	logger.h: Extract DEBUG_ATLEAST() to complement VERBOSITY_ATLEAST(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	565d8eb17c	astobj2: tweak ao2_replace to do nothing when it would be a NoOp ... This change causes ao2_replace to do nothing when src == dst. This avoids REF_DEBUG logging when we're not actually doing anything. Review: https://reviewboard.asterisk.org/r/3743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7e22dfbd8b	include/asterisk/xmpp.h: Convert indentation to tabs ... This is a whitespace only change. ........ Merged revisions 418323 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	787e6a8d58	ARI: Make mixing bridges propagate linkedids and accountcodes. ... * Create a Stasis bridge sub-class to propagate linkedids and accountcodes. * Fixed the basic bridge sub-class to update peeraccount codes when the number of channels in the bridge drops back down to two parties. * Refactored ast_bridge_channel_update_accountcodes() to handle channels joining/leaving the bridge. * Fixed the basic bridge sub-class to not call the base bridge class pull method twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3720/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	36719cba20	manager/ARI: Update version to 2.4.0/1.4.0; Update UPGRADE.txt ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6e2d4a5106	res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence. ... This module implements dialog-info+xml for the purposes of presence. This means that phones such as Grandstreams can now subscribe to receive presence information for an extension. ASTERISK-21443 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3705/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ecaad9e608	ARI/res_stasis: Subscribe to both Local channel halves when originating to app ... This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ ASTERISK-23939 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	15061dadb9	HTTP: Add persistent connection support. ... Persistent HTTP connection support is needed due to the increased usage of the Asterisk core HTTP transport and the frequency at which REST API calls are going to be issued. * Add http.conf session_keep_alive option to enable persistent connections. * Parse and discard optional chunked body extension information and trailing request headers. * Increased the maximum application/json and application/x-www-form-urlencoded body size allowed to 4k. The previous 1k was kind of small. * Removed a couple inlined versions of ast_http_manid_from_vars() by calling the function. manager.c:generic_http_callback() and res_http_post.c:http_post_callback() * Add missing va_end() in ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	387fa1df51	Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 ... ........ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip and chan_pjsip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ Review: https://reviewboard.asterisk.org/r/3686/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a36c7c2d81	res_pjsip: Add ActionID to events created as a result of PJSIP AMI actions ... A number of various PJSIP AMI actions were failing to parse out and place the ActionID into their responses. This patch updates the various PJSIP actions such that the passed in ActionID is emitted on any event list complete events, as well as any intermediate events created as a result of the action. ASTERISK-23947 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3675/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a2f2cc2c72	res_http_websocket: Close websocket correctly and use careful fwrite ... When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	fbd533ef90	astobj2: Add an ao2_replace macro to astobj2.h ... This macro replaces one object reference with another cleaning up the original. param dst Pointer to the object that will be cleaned up. param src Pointer to the object replacing it. src's ref count is bumped if it's non-NULL. dst's ref count is decremented if it's non-NULL. src is assigned to dst, This patch was reviewed on IRC by coreyfarrell and mjordan. Tested by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	04ead6d18f	build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better. ... ast_ext_tool_check.m4 isn't handling cases where a path to a package is provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config tool (E.G. mysql_config) and the package has its own subdirectories in include or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include. Both cause configure to fail and there are others in the same boat. The problem is caused by logic in ast_ext_tool_check that overrides the result of the config tool's --cflags and --libs options if package_DIR is set. This patch prepends package_DIR (if specified) to the -L and -I results from the package's config tool instead of overriding them. A regenerated ./configure and include/asterisk/autoconfig.h.in are included but can be regenerated by running ./bootstrap.sh at any time. Tested by: George Joseph Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3550/ ........ Merged revisions 416929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416930 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	43e0a4ddea	pjsip cli: Change Identify to show CIDR notation instead of netmasks. ... * Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask instead of ast_sockaddr_stringify_addr. * Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead of ast_ha_join() for the CLI output. This is a CLI change only. AMI was not affected. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	74d550fc04	stasis: Reduce creation of channel snapshots to improve performance ... During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	73cf8b2cb8	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416067 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	44dd7898b2	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	197505339d	res_pjsip_pubsub: Persist subscriptions in sorcery so they are recreated on startup. ... This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default this uses the local astdb but it can also be configured to store within an outside database. When Asterisk is started these subscriptions are recreated if they have not expired. Notifications are sent to the devices which have subscribed and they are none the wiser that the system has restarted. Review: https://reviewboard.asterisk.org/r/3598/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c87dbe9922	bridges/bridge_native_rtp: Reconfigure bridge on removal of framehook ... This patch is a re-do of r414122. When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft hangup flags have a catastrophic effect on the pbx core if they leak out from the bridge layer: the channel gets hung up. With the number of threads involved in a blind transfer, and with the initial patch, it was likely that this would occur. This caused a large number of test failures This patch is nearly identical with the one proposed in r414122, save for the following changes: - We explicitly clear the UNBRIDGE flag when setting an after goto on a channel in a bridge - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it https://reviewboard.asterisk.org/r/3585/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7cccdb70d4	bridge.h: Remove redundant struct ast_bridge_channel forward declaration. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	9608543a7a	chan_sip: Fix order of variables specified in SIPNotify action ... Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415390 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	66f5c745ae	Split astobj2.c into more maintainable components. ... Split astobj2.c into the following files to improve maintainability. astobj2.c - object primitives, object primitive misc and initialization code. astobj2_private.h - internal object declarations needed by the containers. astobj2_container.c - generic conainer and container misc code. astobj2_container_hash.c - hash container specific code. astobj2_container_rbtree.c - rbtree container specific code. astobj2_container_private.h - generic container definitions and rtti prototypes. https://reviewboard.asterisk.org/r/3576/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@415317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago