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2015-09-21res/res_stasis_device_state: Allow for subscribing to 'all' device stateMatt Jordan
This patch adds support for subscribing to all device state changes. This is done either by subscribing to an empty device, e.g., 'eventSource=deviceState:', or by the WebSocket connection specifying that it wants all state in the system. ASTERISK-24870 Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32
2015-09-21ARI: Add events for Contact and Peer Status changesMatt Jordan
This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
2015-09-21Merge "main/config_options: Check for existance of internal object before ↵Matt Jordan
derefing"
2015-09-19Merge "app_queue: AgentComplete event has wrong reason"Matt Jordan
2015-09-19Merge "app_queue: Crash when transferring"Matt Jordan
2015-09-19Merge "CHAOS: res_pjsip_diversion avoid crash if allocation fails"Matt Jordan
2015-09-19Merge "dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF"Matt Jordan
2015-09-19Merge "chan_sip: Fix From header truncation for extremely long CALLERID(name)."Joshua Colp
2015-09-19Merge "CHAOS: avoid crash if string create fails"Joshua Colp
2015-09-18CHAOS: avoid crash if string create failsScott Griepentrog
Validate string buffer allocation before using them. ASTERISK-25323 Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0
2015-09-18dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUFRodrigo Ramírez Norambuena
Remove repeated code on macro of assigned buffer to SQL vars Change-Id: Icb19ad013124498e172ea1d0b29ccd0ed17deef0
2015-09-18chan_sip: Fix From header truncation for extremely long CALLERID(name).Walter Doekes
The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260
2015-09-17CHAOS: res_pjsip_diversion avoid crash if allocation failsRichard Mudgett
Validate ast_malloc buffer returned before using it in set_redirecting_value(). ASTERISK-25323 Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253
2015-09-17app_queue: AgentComplete event has wrong reasonKevin Harwell
When a queued caller transfers an agent to another extension sometimes the raised AgentComplete event has a reason of "caller" and sometimes "transfer". Since a transfer has taken place this should always be transfer. This occurs because sometimes the stasis hangup event arrives before the transfer event thus writing a different reason out. With this patch, when a hangup event is received during a transfer it will check to see if the channel that is hanging up is part of a transfer. If so it will return and let the subsequently received transfer event handler take care of the cleanup. ASTERISK-25399 #close Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d
2015-09-17PJSIP: avoid crash when getting rtp peerScott Griepentrog
Although unlikely, if the tech private is returned as a NULL, chan_pjsip_get_rtp_peer() would crash. ASTERISK-25323 Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a
2015-09-17app_queue: Crash when transferringKevin Harwell
During some transfer scenarios involving queues Asterisk would sometimes crash when trying to obtain a channel snapshot (could happen on caller or member channels). This occurred because the underlying channel had already disappeared when trying to obtain the latest snapshot. This patch adds a reference to both the member and caller channels that extends to the lifetime of the queue'd call, thus making sure the channels will always exist when retrieving the latest snapshots. ASTERISK-25185 #close Reported by: Etienne Lessard Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128
2015-09-17res_pjsip_pubsub: Eliminate race during initial NOTIFY.Mark Michelson
There is a slim chance of a race condition occurring where two threads can both attempt to manipulate the same area. Thread A can be handling an incoming initial SUBSCRIBE request. Thread A lets the specific subscription handler know that the subscription has been established. At this point, Thread B may detect a state change on the subscribed resource and queue up a notification task on Thread C, the subscription serializer thread. Now Thread A attempts to generate the initial NOTIFY request to send to the subscriber at the same time that Thread C attempts to generate a state change NOTIFY request to send to the subscriber. The result is that Threads A and C can step on the same memory area, resulting in a crash. The crash has been observed as happening when attempting to allocate more space to hold the body for the NOTIFY. The solution presented here is to queue the subscription establishment and initial NOTIFY generation onto the subscription serializer thread (Thread C in the above scenario). This way, there is no way that a state change notification can occur before the initial NOTIFY is sent, and if there is a quick succession of NOTIFYs, we can guarantee that the two NOTIFY requests will be sent in succession. Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815
2015-09-15scheduler: Use queue for allocating sched IDs.Mark Michelson
It has been observed that on long-running busy systems, a scheduler context can eventually hit INT_MAX for its assigned IDs and end up overflowing into a very low negative number. When this occurs, this can result in odd behaviors, because a negative return is interpreted by callers as being a failure. However, the item actually was successfully scheduled. The result may be that a freed item remains in the scheduler, resulting in a crash at some point in the future. The scheduler can overflow because every time that an item is added to the scheduler, a counter is bumped and that counter's current value is assigned as the new item's ID. This patch introduces a new method for assigning scheduler IDs. Instead of assigning from a counter, a queue of available IDs is maintained. When assigning a new ID, an ID is pulled from the queue. When a scheduler item is released, its ID is pushed back onto the queue. This way, IDs may be reused when they become available, and the growth of ID numbers is directly related to concurrent activity within a scheduler context rather than the uptime of the system. Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2
2015-09-11main/config_options: Check for existance of internal object before derefingMatt Jordan
Asterisk can load and register an object type while still having an invalid sorcery mapping. This can cause an issue when a creation call is invoked. For example, mis-configuring PJSIP's endpoint identifier by IP address mapping in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a subsequent ARI invocation to create the object will cause a crash, as the internal type may not be registered as sorcery expects. Merely checking for a NULL pointer here solves the issue. Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac
2015-09-11Merge "res/res_pjsip_nat: Ignore REGISTER requests when looking for a ↵Matt Jordan
Record-Route"
2015-09-11Merge "chan_sip.c: Validation on module reload"Matt Jordan
2015-09-10chan_sip.c: Validation on module reloadRodrigo Ramírez Norambuena
Change validation on reload module because now used the cli function for reload. The sip_reload() function never fail and ever return NULL for this reason on reload() now use the call the sip_reload() and return AST_MODULE_LOAD_SUCCESS. This problem is dectected on reload by PUT method on ARI, getting always 404 http code when the module is reloaded. ASTERISK-25325 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb
2015-09-10res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.Richard Mudgett
Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093
2015-09-10res_pjsip_pubsub.c: Add some notification comments.Richard Mudgett
Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20
2015-09-10res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.Richard Mudgett
We should not try to send a SIP response message because we may be restoring a persistent subscription where we are not responding to a SIP request. Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec
2015-09-10res_pjsip_pubsub.c: Fix off-nominal memory leak.Richard Mudgett
Fix off-nominal visited vector leak in build_resource_tree(). Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c
2015-09-10res_pjsip_pubsub.c: Fix one byte buffer overrun error.Richard Mudgett
ast_sip_pubsub_register_body_generator() did not account for the null terminator set by sprintf() in the allocated output buffer. Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2
2015-09-10res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().Richard Mudgett
Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3
2015-09-10res_pjsip_pubsub.c: Add missing error return in load_module().Richard Mudgett
Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc
2015-09-10res_pjsip/location.c: Use the builtin ao2_callback() match function instead.Richard Mudgett
Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f
2015-09-10res_pjsip: Copy default_from_user to avoid crash.Mark Michelson
The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c
2015-09-10res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-RouteMatt Jordan
We will only rewrite the Contact header if there is no Record-Route header in the received request. If a malfunctioning proxy places a Record-Route header into a REGISTER request, we will decide that we shouldn't update the IP/port in the Contact header, and we will end up storing a contact with an AoR that contains the NAT'd IP address. While it is nice to have the proxy *not* send a Record-Route in a REGISTER request, it's also a good idea to not process the header in a non-dialog message. This patch updates the code to explicitly ignore the Record-Route header in REGISTER requests. ASTERISK-25387 #close Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f
2015-09-10Merge "ParkAndAnnounce: Add variable inheritance"Joshua Colp
2015-09-09Merge "chan_ooh323: Add ProgressIndicator IE with inband info available"Matt Jordan
2015-09-09Merge "pjsip: avoid possible crash req_caps allocation failure"Joshua Colp
2015-09-09chan_ooh323: Add ProgressIndicator IE with inband info availableAlexander Anikin
Add ProgressIndicator IE with inband info present to Progress and Alerting Q.931 message ASTERISK-25227 #close Reported by: Alexandr Dranchuk Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203
2015-09-09pjsip: avoid possible crash req_caps allocation failureScott Griepentrog
Make certain that the pjsip session has not failed to allocate the format capabilities structure, which can otherwise cause a crash when referenced. ASTERISK-25323 Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750
2015-09-09Merge "res_pjsip: Use hash for contact object identity instead of Contact URI."Joshua Colp
2015-09-08Merge "res_rtp_asterisk: Add more ICE debugging"Matt Jordan
2015-09-08Merge "Core/General: Add #ifdef needed on FreeBSD."Joshua Colp
2015-09-08res_rtp_asterisk: Add more ICE debuggingDavid M. Lee
In working through a recent ICE negotiation bug, I found the debug logging in res_rtp_asterisk to be lacking. This patch adds a number of debug and warning statements that were helpful. Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80
2015-09-08Merge "res/res_pjsip: Purge contacts when an AoR is deleted"Joshua Colp
2015-09-08res_pjsip: Use hash for contact object identity instead of Contact URI.Joshua Colp
In the wild it is possible for Contact URIs to be quite long as parameters can exist on them. This can present a problem when storing them in the AstDB as the URI is used as part of the object name and there is a fixed length limit for the AstDB. This will cause the contact to not get stored. This change uses the MD5 hash of the Contact URI as part of the object name instead. This has a fixed length which is guaranteed to not exceed the AstDB length limit. ASTERISK-25295 #close Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02
2015-09-07chan_ooh323: call ast_rtp_instance_stop on ooh323_destroyAlexander Anikin
Call ast_rtp_instance_stop on ooh323_destroy to free resources allocated by rtp instance ASTERISK-25299 #close Report by: Alexandr Dranchuk Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f
2015-09-07res/res_pjsip: Purge contacts when an AoR is deletedMatt Jordan
When an AoR is deleted by an external mechanism, such as through ARI, we currently do not remove dynamic contacts that were created for that AoR as a result of a received REGISTER request. As a result, re-creating the AoR will cause the dynamic contact to be interpreted as a persistent contact, leading to some rather strange state being created for the contacts/endpoints. This patch adds a sorcery observer for the 'aor' object. When a delete is issued on the underlying sorcery object, the observer is called, and all contacts created and persisted in sorcery for that AoR are also removed. Note that we don't want to perform this action when an AO2 object that is an AoR is destroyed, as the AoR can still exist in the backing storage (and we would thus be removing valid contacts from an AoR that still "exists".) ASTERISK-25381 #close Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328
2015-09-05Merge "endpoint snapshot: avoid second cleanup on alloc failure"Matt Jordan
2015-09-05Merge "channels/pjsip/dialplan_functions: Add an option for extracting the ↵Matt Jordan
SIP call-id"
2015-09-05Merge "res_pjsip: Change default from user value."Joshua Colp
2015-09-05Merge "Fix when remote candidates exceed PJ_ICE_MAX_CAND"Joshua Colp
2015-09-05channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-idMatt Jordan
This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
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