asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	5e9230b528	uuid: Enable UUID in Solaris 11. ASTERISK-27933 Reported by: bautsche Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49	8 years ago
George Joseph	498e775a4a	Merge changes from topic 'ASTERISK-27625' into 13 * changes: channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. channel.c: Fix usage of CHECK_BLOCKING() autoservice: Don't start channel autoservice if the thread is a user interface.	8 years ago
George Joseph	aedbf5ba58	Merge "ARI POST DTMF: Make not compete with channel's media thread." into 13	8 years ago
George Joseph	4228dc7b4f	Merge "AMI PlayDTMF Action: Make not compete with channel's media thread." into 13	8 years ago
Richard Mudgett	f878de44af	channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. * Removed an unnecessary call to ast_channel_blocker_set() in __ast_read(). ASTERISK-27625 Change-Id: I342168b999984666fb869cd519fe779583a73834	8 years ago
Richard Mudgett	99f439dc01	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	8 years ago
Richard Mudgett	6a1626c265	AMI PlayDTMF Action: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	8 years ago
Richard Mudgett	1abcc41fff	channel.c: Fix usage of CHECK_BLOCKING() The CHECK_BLOCKING() macro is used to indicate if a channel's handling thread is about to do a blocking operation (poll, read, or write) of media. A few operations such as ast_queue_frame(), soft hangup, and masquerades use the indication to wake up the blocked thread to reevaluate what is going on. ASTERISK-27625 Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d	8 years ago
Richard Mudgett	5d34ca5b33	autoservice: Don't start channel autoservice if the thread is a user interface. Executing dialplan functions from either AMI or ARI by getting a variable could place the channel into autoservice. However, these user interface threads do not handle the channel's media so we wind up with two threads attempting to handle the media. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49	8 years ago
Richard Mudgett	7b30158147	Fix some doxygen and curly placement. Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8	8 years ago
Alexander Traud	ef2386fcd7	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	8 years ago
Alexander Traud	8c78337479	tcptls.h: Repair ./configure --with-ssl=PATH. asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those inclusions got replaced by forward declarations. As side effect, the inclusions got completed. ASTERISK-27878 Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7	8 years ago
Alexander Traud	b6234f9577	tcptls: Repair ./configure --with-ssl=PATH. SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH. ASTERISK-27865 Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71	8 years ago
Joshua Colp	8926bc20fd	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code." into 13	8 years ago
Joshua Colp	ac9d6b0523	Merge "pjsip: Rewrite OPTIONS support with new eyes." into 13	8 years ago
Alexander Traud	9fe4f99cba	rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48	8 years ago
George Joseph	42abc9c430	Merge "BuildSystem: Add DragonFly BSD." into 13	8 years ago
Joshua Colp	bea52b3706	pjsip: Rewrite OPTIONS support with new eyes. The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	8 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	8 years ago
Alexander Traud	7b219311eb	BuildSystem: Add DragonFly BSD. ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	8 years ago
Jenkins2	57aca68bbf	Merge "utils: Add ast_assert_return" into 13	8 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
George Joseph	39c51394c8	utils: Add ast_assert_return Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the following... If the assert passes... NoOp If the assert fails and AST_DEVMODE is defined, execute ast_assert() then, if DO_CRASH isn't set, return from the calling function with the supplied value. If the assert fails and AST_DEVMODE is not defined, return from the calling function with the supplied value. The macro will execute a return without a value if one isn't suppled. Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e	8 years ago
Jenkins2	ad0ba520b5	Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks." into 13	8 years ago
Richard Mudgett	b92ebdba5f	pjsip_scheduler.c: Add ability to trace scheduled tasks. When a scheduled task is created you can pass in the AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to be logged. Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b	8 years ago
Richard Mudgett	12aa25b2e1	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	8 years ago
Richard Mudgett	dfdc9a2575	pjsip_scheduler.c: Fix some corner cases. * Fix the periodic interval wander because it may take significant time between the sched thread queueing the task in the serializer and the serializer actually executing the task. The time it takes to actually execute the task was already taken into account. * Pass a schtd ref to the serializer when we queue a scheduled task on the serializer. We don't want it going away on us while it is in the serializer queue. * Skip the scheduled task if the task was canceled between queueing the task to the serializer and the serializer actually executing the task. * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed task_id and added next_periodic. * Hold a ref to the passed in serializer so the serializer cannot go away on the scheduled task. ASTERISK_26806 Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24	8 years ago
Jenkins2	dfd0529abc	Merge "pjsip_scheduler.c: Fix ao2 usage errors." into 13	8 years ago
Richard Mudgett	c4f02c975b	pjsip_scheduler.c: Fix ao2 usage errors. * Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the 'tasks' container being accessed without a lock in a multi-threaded environment. A recipe for crashes. * Removed needlessly obtaining schtd object references. If the caller providing you a pointer to an object doesn't have a valid reference then you cannot safely get one from it. * Getting a ref to 'tasks' when you aren't copying the pointer into another location is useless. The 'tasks' container pointer is global. * Removed many unnecessary uses of RAII_VAR. * Make ast_sip_schedule_task() name parameter const. ASTERISK_26806 Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db	8 years ago
Richard Mudgett	72b16ee400	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	8 years ago
Jenkins2	dcac292ae3	Merge "BuildSystem: Add support for building RADIUS with radcli." into 13	8 years ago
Richard Mudgett	e94f8e4a24	res_pjsip: Update authenticate_qualify documentation. Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4	8 years ago
Alexander Traud	83353997f4	BuildSystem: Add support for building RADIUS with radcli. Radcli is yet another RADIUS client library, generally compatible with freeradius and radiusclient-ng. This commit adds autoconf option for detecting it as well and changes cdr_radius and cel_radius to use its header file in that case. ASTERISK-26540 Reported by: Tzafrir Cohen Change-Id: Icc056d476b7acf481309219e9abdca416866c6ec	8 years ago
Corey Farrell	5908c6753b	core: Create main/options.c. This creates a separate source to 'own' symbols related to options.h and paths.h. This significantly reduces the number of exports created by main/asterisk.o. This change is required to eventually be able to link unmodified Asterisk sources to utilities and/or stand-alone tests. ASTERISK~26245 Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380	8 years ago
Jenkins2	7d8445d576	Merge "core: Remove dead symbols from asterisk.exports.in." into 13	8 years ago
Jenkins2	b39c727848	Merge "channel.c: Allow generic plc then channel formats are equal" into 13	8 years ago
Corey Farrell	6f304697b0	core: Remove dead symbols from asterisk.exports.in. * dahdi_chan_name * dahdi_chan_name_len * dahdi_chan_mode * __manager_event * dialed_interface_info Added comment about __progname and environ being needed for FreeBSD to prevent accidental removal in the future. Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5	8 years ago
George Joseph	373e7e3fb0	channel.c: Allow generic plc then channel formats are equal If the two formats on a channel are equal, we don't transcode and since the generic plc needs slin to work, it doesn't get invoked. * A new configuration option "genericplc_on_equal_codecs" was added to the "plc" section of codecs.conf to allow generic packet loss concealment even if no transcoding was originally needed. Transcoding via SLIN is forced in this case. ASTERISK-27743 Change-Id: I0577026a179dea34232e63123254b4e0508378f4	8 years ago
Alexander Traud	0f634c1446	BuildSystem: Remove unused dependency on libltdl. Asterisk does not need the development package of libltdl, because it does not use any symbol of -lltdl directly. Instead, it uses the runtime package via the shared library -lodbc. On the supported platforms, that shared library declares its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have failed. ASTERISK-27745 Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba	9 years ago
Jenkins2	a243fed64f	Merge "core: Remove incorrect usage of attribute_malloc." into 13	9 years ago
Corey Farrell	dc738b145f	core: Remove incorrect usage of attribute_malloc. GCC documentation states that when __attribute__((malloc)) is used it should not return storage which contains any valid pointers. It specifically mentions that realloc functions should not have the malloc attribute, but this also means that complex initializers which could contain initialized pointers should not use this attribute. Change-Id: If507f33ffb3ca3b83b702196eb0e8215d27fc7d2	9 years ago
Alexander Traud	54874eb44d	BuildSystem: Enable PortAudio in NetBSD. In NetBSD, PortAudio 1 is still the default version. PortAudio 2 can be installed side by side but gets placed in a 'portaudio2' subdirectory. To find PortAudio 2 even in a subdirectory, the tool pkg-config is queried via AST_PKG_CONFIG_CHECK. For those platforms, which do not list PowerAudio 2 via pkg-config, the previous check remains and is executed thereafter. ASTERISK-27721 Change-Id: I4175500126909ad1b181fff8e11bb4a3a6ae4fa9	9 years ago
Jenkins2	91193807c8	Merge "BuildSystem: Detect whether uselocale(.) is available." into 13	9 years ago
Alexander Traud	98e8e849da	BuildSystem: Detect whether uselocale(.) is available. ASTERISK-27712 Reported by: Joerg Sonnenberger, D'Arcy Cain Change-Id: Idf1c9d43617a3e13028b95b313415903d80ef807	9 years ago
Richard Mudgett	104468ad3a	pjproject: Add cache_pools debugging option. The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	9 years ago
Kevin Harwell	ffb15b2bc7	AMI: Bumping AMI non-breaking number for Asterisk 13.20.0 release A few changes were made to AMI: * Fixed "(null):" header in AMI AsyncAGIEnd event * A mute header was added to the ConfbridgeJoin AMI event * ConfbridgeList action's ConfbridgeList events now output all the standard channel snapshot headers Change-Id: I94a82a44b02c91becae08d254e9a56abba5697cf	9 years ago
Alexander Traud	4a5221cd43	BuildSystem: Remove chan_h323 leftovers. ASTERISK-27670 Change-Id: I07a8ef8bbd6001e25711fa1bff152eb6c9efa729	9 years ago
Jenkins2	62d491527e	Merge "app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs." into 13	9 years ago
George Joseph	7debdd285c	res_pjsip_pubsub: Prune subs with reliable transports at startup In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	9 years ago
Richard Mudgett	4a337b1a76	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	9 years ago

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3640 Commits (ef788f4d6473f7708e60cdbc90c8cdbefdb7dc91)