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Author	SHA1	Message	Date
Scott Griepentrog	537df26f9c	Channel alert pipe: improve diagnostic error return ... When a frame is queued on a channel, any failure in ast_channel_alert_write is logged along with errno. This change improves the diagnostic message through aligning the errno value with actual failure cases. ASTERISK-25224 Reported by: Andrey Biglari Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b	10 years ago
Joshua Colp	c12ace3ab3	Merge "res_sorcery_realtime: Fix leak of sorcery object type."	10 years ago
Matt Jordan	3cd53bd3bc	Merge "channel: Remove ignore of answer on non-outgoing channels."	10 years ago
Mark Michelson	58d18324f0	res_sorcery_realtime: Fix leak of sorcery object type. ... This prevents a leak of a sorcery object type when realtime sorcery objects are retrieved by fields or when multiple objects are retrieved. The extent of this leak is that sorcery object types would be leaked. These are allocated whenever an object type is registered with sorcery, meaning that on module shutdown, these objects would be leaked. This could be problematic if many reloads were performed, but it is not as severe as if every sorcery object retrieved from realtime were being leaked. ASTERISK-25165 #close Reported by Corey Farrell Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8	10 years ago
Matt Jordan	598a5f0d15	Merge "res_pjsip_nat: Adjust when contact should be rewritten."	10 years ago
Matt Jordan	d66c059fbc	Merge "res/res_corosync: Always decline module load, instead of failing"	10 years ago
Matt Jordan	80d97290bb	res/res_corosync: Always decline module load, instead of failing ... Returns a 'failure' from the module load routine indicates to Asterisk that it should abort loading completely. This is rarely - in fact, really, never - a good option. Aborting load of Asterisk from a dynamic module implies that the core, and the rest of the dynamic modules, don't matter: we should abandon all processing. res_corosync is really not that important. This patch updates the module such that, if it fails to load, it politely declines (emitting ERROR messages along the way), and allows Asterisk to continue to function. Note that this issue was keeping Asterisk unit tests from running on certain build agents. Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f	10 years ago
Matt Jordan	892cc5625f	main/pbx: Resolve case sensitivity regression in PBX hints ... When 8297136f was merged for ASTERISK-25040, a regression was introduced surrounding the case sensitivity of device names within hints. Previously, device names - such as 'sip/foo' - were compared in a case insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After that patch, only the case sensitive name would match, i.e., 'SIP/foo'. As a result, some dialplan hints stopped working. This patch re-introduces case insensitive matching for device names in hints. ASTERISK-25040 ASTERISK-25202 #close Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c (cherry picked from commit 96bbcf495a)	10 years ago
Mark Michelson	e18b22a806	res_pjsip_nat: Adjust when contact should be rewritten. ... A previous change made the contact only get rewritten if the dialog's route set was not marked frozen. Unfortunately, while the intent of this is correct, the dialog's route set actually gets marked as frozen earlier than expected, especially for UAS dialogs. Instead, the idea is that the contact needs to not be rewritten if there is a pre-existing route set on the dialog. This is now accomplished by checking the dialog's route set list instead of checking if the route set is frozen. Doing this causes some broken tests to begin passing again. ASTERISK-25196 Reported by Mark Michelson Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e	10 years ago
Richard Mudgett	99b1aa6d26	res_pjsip_outbound_registration.c: Add a serializer shutdown group. ... The client_state objects contain a serializer used to send the outbound REGISTER messages. Once all those message transactions are complete then the module can shutdown. ASTERISK-24907 #close Reported by: Kevin Harwell Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547	10 years ago
Matt Jordan	c0194b55b5	Merge "threadpool, res_pjsip: Add serializer group shutdown API calls."	10 years ago
Matt Jordan	8c1161a268	Merge "res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs"	10 years ago
Matt Jordan	4208fe6ba7	Merge "res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API"	10 years ago
Matt Jordan	d568177d7d	Merge "res_pjsip_refer: Prevent sending duplicate headers."	10 years ago
Matt Jordan	771b9bb442	Merge "sorcery: Add ast_sorcery_object_unregister() API call."	10 years ago
Matt Jordan	8a9628dce5	Merge "res_pjsip_outbound_registration.c: Reorder load_module() and unload_module()."	10 years ago
Matt Jordan	19fcb928b2	Merge "AMI: Add Linkedid to the standard channel snapshot AMI event headers."	10 years ago
Matt Jordan	05dbfedb43	Merge "res_pjsip_nat: Rewrite route set when required."	10 years ago
Mark Michelson	f536e9b59c	res_pjsip_refer: Prevent sending duplicate headers. ... res_pjsip_refer will attempt to add Referred-By or Replaces headers to outbound INVITEs at times. If the INVITE gets challenged for authentication, then we will resend the INVITE. Prior to this patch, the Referred-By or Replaces header would be re-added to the outbound INVITE, resulting in duplicated headers. ASTERISK-25204 #close Reported by Mark Michelson Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d	10 years ago
Richard Mudgett	c2d48a2a28	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ... ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	10 years ago
Mark Michelson	700606a659	res_pjsip_nat: Rewrite route set when required. ... When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f	10 years ago
Richard Mudgett	af4ae3095e	threadpool, res_pjsip: Add serializer group shutdown API calls. ... A module trying to unload needs to wait for all serializers it creates and uses to complete processing before unloading. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059	10 years ago
Richard Mudgett	4c133d81cd	res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs ... * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f	10 years ago
Richard Mudgett	dc63377c60	res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API ... The sorcery pjsip 'registration' config object needs to be destroyed on module unload. Otherwise, a reload of res_pjsip could try to use callbacks for a previously unloaded instance of the module provided by ast_sorcery_object_register() or one of the variants. Also, if res_pjsip_outbound_registration were subsequently reloaded, the sorcery config field objects would be registered in sorcery twice. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697	10 years ago
Richard Mudgett	9ec8a0f3cc	sorcery: Add ast_sorcery_object_unregister() API call. ... Find and unlink the specified sorcery object type to complement ast_sorcery_object_register(). Without this function you cannot completely unload individual modules that use sorcery for configuration. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88	10 years ago
Richard Mudgett	77ff7325a2	res_pjsip_outbound_registration.c: Reorder load_module() and unload_module(). ... It is best if the loading code creates and initializes the module's infrastructure before letting the system know of its existence. The unloading code needs to reverse the actions of the loading code and in the reverse order. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4	10 years ago
Joshua Colp	8d6cf667dc	channel: Remove ignore of answer on non-outgoing channels. ... Due to the way that channels can now be moved around inside of Asterisk it is possible for the outgoing flag of a channel to get cleared before it has been answered. This results in the bridge not receiving notification that the outgoing leg has been answered. This most easily exhibits itself with DTMF based blond transfers. Since the answer of the outgoing leg is ignored the other party continues to receive both a locally generated ringing and the media stream of the outgoing leg upon its answer. This results in no media being heard. This change removes the ignore of the answer and allows it to pass through. ASTERISK-25171 #close Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e	10 years ago
Mark Michelson	3f1fe83633	Merge "res_pjsip_mwi: Set up unsolicited MWI upon registration."	10 years ago
Joshua Colp	6fffef78c3	Merge "test.c: Add unit test registration checks for summary and description."	10 years ago
Joshua Colp	52aa9536a1	Merge "Unit tests: Fix more unit test description strings."	10 years ago
Joshua Colp	7f30be6f5e	Merge "Unit tests: Fix unit test description strings."	10 years ago
Joshua Colp	2315a08297	Merge "DNS unit tests: Fix extraneous description string commas."	10 years ago
Richard Mudgett	daaa551c92	test.c: Add unit test registration checks for summary and description. ... Added checks when a unit test is registered to see that the summary and description strings do not end with a new-line '\n' for consistency. The check generates a warning message and will cause the /main/test/registrations unit test to fail. * Updated struct ast_test_info member doxygen comments. Change-Id: I295909b6bc013ed9b6882e85c05287082497534d	10 years ago
Richard Mudgett	71a4d1a033	Unit tests: Fix more unit test description strings. ... Analyzing the code shows that the unit test summary and description strings should not end with a new-line character. Where these strings are used in the code a new-line is provided for output. Change-Id: I2f4f37988ec363c8d1c5077a2fc8ca841c5cd30c	10 years ago
Richard Mudgett	9c6d72e30d	Unit tests: Fix unit test description strings. ... Analyzing the code shows that the unit test summary and description strings should not end with a new-line character. Where these strings are used in the code a new-line is provided for output. Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba	10 years ago
Richard Mudgett	a0c2d2089d	DNS unit tests: Fix extraneous description string commas. ... Change-Id: Icf5f13c8e1c2c92a4473bb573ed2dd856ce1b64e	10 years ago
Joshua Colp	3b2b004d69	app_dial: Hold reference to calling channel formats when dialing outbound. ... Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db	10 years ago
Richard Mudgett	af66b0f3f7	res_pjsip_outbound_registration.c: Add missing line endings to CLI commands ... Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7	10 years ago
Richard Mudgett	3f0708e5fe	res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage. ... Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e	10 years ago
Richard Mudgett	9ceb848242	res_pjsip_outbound_registration.c: Misc code cleanups. ... * Break some long lines. * Fix doxygen comment. Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305	10 years ago
Joshua Colp	39c79cd6fb	Merge "res_pjsip_outbound_registration.c: Fix whitespace conflict potential."	10 years ago
Kevin Harwell	44c3c392e3	bridge.c: Hangup attended transfer target if bridged ... After completing an attended transfer the transfer target channel was not being hung up after leaving the bridge. Added an explicit softhangup to hangup said channel, but only if it was previously bridged. ASTERISK-24782 #close Reported by: John Bigelow Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada	10 years ago
Joshua Colp	7846f73432	res_pjsip_mwi: Set up unsolicited MWI upon registration. ... The res_pjsip_mwi previously required a reload to set up the proper subscriptions to allow unsolicited MWI to work. This change makes it so the act of registering will also cause this to occur. This is particularly useful if realtime is involved as no reload needs to occur within Asterisk to cause the MWI information to get sent. ASTERISK-25180 #close Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252	10 years ago
Richard Mudgett	096b27d9d2	res_pjsip_outbound_registration.c: Fix whitespace conflict potential. ... Change-Id: I82e6e388e3688aebe0783f16c9e0800a747584b5	10 years ago
Alexander Traud	1ad9a6b6b6	chan_sip: Reload peer without its old capabilities. ... On reload, previously allowed codecs were not removed. Therefore, it was not possible to remove codecs while Asterisk was running. Furthermore, newly added codecs got appended behind the previous codecs. Therefore, it was not possible to add a codec with a priority of #1. This change removes the old capabilities before the current ones are added. ASTERISK-25182 #close Reported by: Alexander Traud patches: asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802	10 years ago
Joshua Colp	5caefc98a1	chan_sip: Destroy peers without holding peers container lock. ... Due to the use of stasis_unsubscribe_and_join in the peer destructor it is possible for a deadlock to occur when an event callback is occurring at the same time. This happens because the peer may be destroyed while holding the peers container lock. If this occurs the event callback will never be able to acquire the container lock and the unsubscribe will never complete. This change makes it so the peers that have been removed from the peers container are not destroyed with the container lock held. ASTERISK-25163 #close Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33	10 years ago
Matt Jordan	bd77ace25a	Merge "Resolve race conditions involving Stasis bridges."	10 years ago
Mark Michelson	d7a1e84a1e	Resolve race conditions involving Stasis bridges. ... This resolves two observed race conditions. First, a bit of background on what the Stasis application does: 1a Creates a stasis_app_control structure. This structure is linked into a global container and can be looked up using a channel's unique ID. 2a Puts the channel in an event loop. The event loop can exit either because the stasis_app_control structure has been marked done, or because of some other factor, such as a hangup. In the event loop, the stasis_app_control determines if any specific ARI commands need to be run on the channel and will run them from this thread. 3a Checks if the channel is bridged. If the channel is bridged, then ast_bridge_depart() is called since channels that are added to Stasis bridges are always imparted as departable. 4a Unlink the stasis_app_control from the container. When an ARI command is received by Asterisk, the following occurs 1b A thread is spawned to handle the HTTP request 2b The stasis_app_control(s) that corresponds to the channel(s) in the request is/are retrieved. If the stasis_app_control cannot be retrieved, then it is assumed that the channel in question has exited the Stasis app or perhaps was never in Stasis in the first place. 3b A command is queued onto the stasis_app_control, and the channel's event loop thread is signaled to run the command. 4b While most ARI commands do nothing further, some, such as adding or removing channels from a bridge, will block until the command they issued has been completed by the channel's event loop. The first race condition that is solved by this patch involves a crash that can occur due to faulty detection of the channel's bridged status in step 3a. What can happen is that in step 2a, the event loop may run the ast_bridge_impart() function to asynchronously place the channel into a bridge, then immediately exit the event loop because the channel has hung up. In step 3a, we would detect that the channel was not bridged and would not call ast_bridge_depart(). The reason that the channel did not appear to be bridged was that the depart_thread that is spawned by ast_bridge_impart() had not yet started. That is the thread where the channel is marked as being bridged. Since we did not call ast_bridge_depart(), the Stasis application would exit, and then the channel would be destroyed Then the depart_thread would start up and try to manipulate the destroyed channel, causing a crash. The fix for this is to switch from using ast_channel_is_bridged() to checking the NULLity of ast_channel_internal_bridge_channel() to determine if ast_bridge_depart() needs to be called. The channel's internal bridge_channel is set when ast_bridge_impart() is called and is NULLed by the call to ast_bridge_depart(). If the channel's internal bridge_channel is non-NULL, then the channel must have been imparted into the bridge and needs to be departed, even if the actual bridging operation has not yet started. By departing the channel when necessary, the thread that is running the Stasis application will block until the bridge gives the okay that the depart_thread has exited. The second race condition that is solved by this patch involves a leak of HTTP handler threads. The problem was that step 2b would successfully retrieve a stasis_app_control structure. Then step 2a would exit the channel from the event loop due to a hangup. Steps 3a and 4a would execute, and then finally steps 3b and 4b would. The problem is that at step 4b, when attempting to add a channel to a bridge, the thread would block forever since the channel would never execute the queued command since it was finished with the event loop. This meant that the HTTP handling thread would be leaked, along with any references that thread may have owned (in my case, I was seeing bridges leaked). The fix for this is to hone in better on when the channel has exited the event loop. The stasis_app_control structure has an is_done field that is now set at each point where the channel may exit the event loop. If step 2b retrieves a valid stasis_app_control structure but the control is marked as done, then the attempted operation exits immediately since there will be nothing to service the attempted command. ASTERISK-25091 #close Reported by Ilya Trikoz Change-Id: If66265b73b4c9f8f58599124d777fedc54576628	10 years ago
Joshua Colp	9668a1acb5	res_sorcery_memory_cache: Remove 'prefetch' option. ... To prevent confusion I am removing the prefetch option until such time as it is implemented. All other functionality, however, has been implemented. ASTERISK-25067 Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895	10 years ago
Matt Jordan	8c0b917032	Merge "Parking: Add documentation for AMI ParkedCallSwap event."	10 years ago