29326 Commits (19a79ae12c73992508910d2c7cddc059e10bc48c)

 

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Author	SHA1	Message	Date
Richard Mudgett	16fdb11bc3	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	9 years ago
Joshua Colp	bab4885f1e	stream: Add media stream definition and API with unit tests. ... This change adds the media stream definition and API for accessing and using it. Unit tests have also been written which exercise aspects of the API. ASTERISK-26773 Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87	9 years ago
George Joseph	648d181d2f	configs/samples: Fix placement of 'identify' entry in sorcery.conf ... The entry for 'identify' was incorrectly placed in the res_pjsip section when it should be in res_pjsip_endpoint_identifier_ip. ASTERISK-26785 #close Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a	9 years ago
Mark Michelson	46147a8f30	Revert "Update qualifies when AOR configuration changes." ... This reverts commit 6492e91392. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e	9 years ago
zuul	c72b4c98e8	Merge "srv: Fix crash when ast_srv_lookup is used and 0 records are returned."	9 years ago
Joshua Colp	5422ec140c	srv: Fix crash when ast_srv_lookup is used and 0 records are returned. ... When performing an SRV lookup using the ast_srv_lookup function it did not properly handle the situation where 0 records are returned. If this happened it would wrongly assume that at least one record was present. This change fixes the code so it will exit early if an error occurs or if 0 records are returned. ASTERISK-26772 patches: srv_lookup.patch submitted by nappsoft (license 6822) Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351	9 years ago
Joshua Colp	b79cc62057	res_stasis_device_state: Protect the adding/removing of subscriptions. ... The adding and removing of device state subscriptions did not protect fully against simultaneous manipulation. In particular the subscribe case allowed a small window where two subscriptions could be added for the same device state instead of just one. This change makes the code hold the subscriptions lock for the entirety of each operation to ensure that two are not occurring at the same time. ASTERISK-26770 Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b	9 years ago
Richard Mudgett	b47cf1a7d6	res_pjsip: Fix some off nominal tdata leaks. ... Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d	9 years ago
Sebastien Duthil	7b280e7ccf	res_ari: fix memory leak for channelvars ... In ari.conf, when setting the option channelvars, every Stasis channel snapshot would create a list of variable/value that would not be freed when the snapshot is freed, resulting in a often-recurring memory leak. ASTERISK-26767 #close Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d	9 years ago
Joshua Colp	e63635b578	Merge "Update qualifies when AOR configuration changes."	9 years ago
Joshua Colp	9e05379f65	Merge "channel.c: Fix unbalanced read queue deadlocking local channels."	9 years ago
Joshua Colp	69fba55838	Merge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)"	9 years ago
Tzafrir Cohen	c6c7f17206	libasteriskssl: do nothing with OpenSSL >= 1.1 ... OpenSSL 1.1 requires no explicit initialization. The hacks in the library are not needed. They also happen to fail running Asterisk. Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100	9 years ago
Tzafrir Cohen	bc041ca14a	tcptls: use TLS_client_method with OpenSSL 1.1 ... OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous version-specific methods (such as TLSv1_client_method(). Other than being simpler to use and more correct (gain support for TLS newer that TLS1, in our case), the older ones produce a deprecation warning that fails the build in dev-mode. Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07	9 years ago
Tzafrir Cohen	2c8d0764de	openssl 1.1 support: use OPENSSL_VERSION_NUMBER ... Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect the openssl 1.1 API. Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458	9 years ago
George Joseph	0478c1f76a	Merge "Add reload options to CLI/AMI stale object commands."	9 years ago
Joshua Colp	5ca45117df	Merge "Frame deferral: Revert API refactoring."	9 years ago
zuul	2b62015e78	Merge "res_odbc: Remove deprecated settings from sample configuration file"	9 years ago
Richard Mudgett	50029f585e	channel.c: Fix unbalanced read queue deadlocking local channels. ... Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6	9 years ago
Richard Mudgett	97c308471d	res_agi: Prevent an AGI from eating frames it should not. (Re-do) ... A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43	9 years ago
Richard Mudgett	72e3fc5845	Frame deferral: Revert API refactoring. ... There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496	9 years ago
zuul	3c558b4b4a	Merge "audiohooks: Muting a hook can mute underlying frames"	9 years ago
Sean Bright	4c51ad158d	res_odbc: Remove deprecated settings from sample configuration file ... ASTERISK-26704 #close Reported by: Anthony Messina Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f	9 years ago
zuul	5504dd3cdd	Merge "res_pjsip: Handle invocation of callback on outgoing request when error occurs."	9 years ago
Richard Mudgett	7d9b50a7b2	res_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap. ... ASTERISK-26765 Change-Id: I27eb97df7f8d7e624b0b9a61c0fcee4718c86d8d	9 years ago
Sean Bright	2849b726b6	audiohooks: Muting a hook can mute underlying frames ... If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed	9 years ago
Joshua Colp	97d58add32	Merge "res_rtp_asterisk: Swap byte-order when sending signed linear"	9 years ago
Mark Michelson	bbed75c3ba	Update qualifies when AOR configuration changes. ... Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121	9 years ago
Joshua Colp	aeea634bc0	res_pjsip: Handle invocation of callback on outgoing request when error occurs. ... There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91	9 years ago
Sean Bright	7a16524a83	res_rtp_asterisk: Swap byte-order when sending signed linear ... Before Asterisk 13, signed linear was converted into network byte order by a smoother before being sent over the network. We restore this behavior by forcing the creation of a smoother when slinear is in use and setting the appropriate flags so that the byte order conversion is always done. ASTERISK-24858 #close Reported-by: Frankie Chin Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9	9 years ago
George Joseph	e252aff9ad	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts ... Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c	9 years ago
zuul	b814343538	Merge "make_build_h: handle backslashes in external strings"	9 years ago
zuul	ada0032305	Merge "app_queue: Fix queues randomly disappearing on reload"	9 years ago
zuul	6cbe894828	Merge "debug_utilities: Add ast_logescalator"	9 years ago
zuul	940c395110	Merge "libastssl/pj: libastssl/pj should have an so_version"	9 years ago
George Joseph	ef4deb8ecd	debug_utilities: Add ast_logescalator ... The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	9 years ago
zuul	3eabae43c4	Merge "tests: use datadir for sound files"	9 years ago
Torrey Searle	178b90af02	libastssl/pj: libastssl/pj should have an so_version ... Issue introduced in b59956a87. In the non-darwin case libastssl/pj should be versioned. This causes the symbol file for this lib to not be generated. Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c (cherry picked from commit 54b027916a)	9 years ago
George Joseph	6f645a6d4e	Merge "media: Add experimental support for RTCP feedback."	9 years ago
zuul	08bc42201d	Merge "res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving."	9 years ago
kkm	138cd8d019	make_build_h: handle backslashes in external strings ... LikewiseOpen creates user names with a backslash in them. A gentle massage with sed(1) allows such strings to be inserted into build.h properly quoted. I am also adding the same for host name and other strings used in the script that are more or less user-controlled. ASTERISK-26754 Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae	9 years ago
kkm	8270d2436d	app_queue: Fix queues randomly disappearing on reload ... With 500+ queues and a reload every minute, a random queue disappears upon reload. The cause is mususe of the 'dead' flag. Namely, all queues were marked dead up front, and then "resurrected" by dropping this flag for those found in the configuration. But a queue marked dead can be removed also when control leaves the app entry point on a PBX thread. With this change, the queue is marked only not found, and at the end of reload only the queues that are still not found are actually marked as dead, so the dead flag is never reset, and set only on positively dead queues. ASTERISK-26755 Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf	9 years ago
zuul	10631bb209	Merge "PJPROJECT logging: Fix detection of max supported log level."	9 years ago
George Joseph	0ad6d2b3cf	Merge "ari: Implement 'debug all' and request/response logging"	9 years ago
George Joseph	071aa069e6	Merge "res_musiconhold.c: Fix format ref leak when parsing MOH config class."	9 years ago
George Joseph	2484c3ee39	Merge "frame.c: Fix off-nominal format ref leaks."	9 years ago
George Joseph	ac5528a028	Merge "test_voicemail_api: order of params to VERIFY macros"	9 years ago
zuul	d7e0e4f43f	Merge "Add notes about embedded ast_frame structs holding a format ref."	9 years ago
Joshua Colp	7fa3de7ae9	res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving. ... This change adds a missing unreference of the hostname when resolving and also cleans up the iterator. ASTERISK-26735 Change-Id: Ic012ebaf3d89e714eec340b7b0c5e63c66af857a	9 years ago
George Joseph	96dbf54e97	Merge "T.140: Fix format ref and memory leaks."	9 years ago