28942 Commits (b9a4ab8c8c00c8d53584d6f7e31729b5027c8dd6)

 

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Author	SHA1	Message	Date
George Joseph	12affa7b6e	Merge "make ari-stubs so doc periodic jobs can run" into 13	8 years ago
Sean Bright	98e38daf82	pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified ... Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE is passed to these functions, the calling thread will be blocked until the newly created channel has been hung up. After this patch, we run the dial on the current thread rather than spawning a new one. The only in-tree code that passes AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced thread usage if you are using .call files. Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913	8 years ago
Richard Mudgett	55f452884f	res_rtp_asterisk.c: Fix crash in RTCP DTLS operation. ... Occasionally a crash happens when processing the RTCP DTLS timeout handler. The RTCP DTLS timeout timer could be left running if we have not completed the DTLS handshake before we place the call on hold or we attempt direct media. * Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP. * Made some sanity tweaks to ast_rtp_prop_set() when switching from standard RTCP mode to RTCP multiplexed mode. ASTERISK-26692 #close Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0	8 years ago
Richard Mudgett	f856cfbb51	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	8 years ago
Sean Bright	dafcd97a77	build: Update config.guess and config.sub ... Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172	8 years ago
Joshua Colp	f9de4b5de9	Merge "format_wav: Read 16khz wav samples properly" into 13	8 years ago
Joshua Colp	69bebd8f1a	Merge "format_ogg_vorbis: Clear ogg/vorbis data structures on close" into 13	8 years ago
Joshua Colp	441a75daea	Merge "Revert "bridging: Ensure successful T.38 negotation"" into 13	8 years ago
Sean Bright	9bbfa6fda1	format_wav: Read 16khz wav samples properly ... When opening a PCM wave file for reading, we aren't tracking the frequency of the opened file, so we treat 16khz files as 8khz and do half reads. This patch also cleans up some of the data types and an unnecessarily complex `if` expression. ASTERISK-26613 #close Reported by: Vitaly K Change-Id: I05f8b263058dc573ea8ffe0c62e7964506e11815	8 years ago
George Joseph	4ccaffe644	make ari-stubs so doc periodic jobs can run ... The periodic doc job does a make ari-stubs and checks that there are no changes before generating the docs. Since I changed the mustache template (and the generated code directly) recently and forgot to regenerate the stubs, the doc job thinks they're out of date. Change-Id: Ibd4bc649556615ff714d44534c45b6c2f6aa449d	8 years ago
Sean Bright	90c630aaa1	format_ogg_vorbis: Clear ogg/vorbis data structures on close ... On filestream close, we need to clear out the ogg & vorbis data structures to prevent a memory leak. ASTERISK-26169 #close Reported by: Ivan Myalkin Change-Id: Iee94c5a5d5bdafbf8b181c5c064d15d90ace8274	8 years ago
Richard Mudgett	9084c85cb1	Revert "bridging: Ensure successful T.38 negotation" ... This reverts commit 3e7c396a51. Change-Id: I61d49d563babff788bb557345729b200d116bd88	8 years ago
Sean Bright	357d1fbdcc	res_stun_monitor: Don't fail to load if DNS resolution fails ... res_stun_monitor will fail to load if DNS resolution of the STUN server fails. Instead, we continue without the STUN server being resolved and we will re-attempt the resolution on the STUN refresh interval. ASTERISK-21856 #close Reported by: Jeremy Kister Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254	8 years ago
Sean Bright	ac15ebc379	format_pcm: Track actual header size of .au files ... Sun's Au file format has a minimum data offset 24 bytes, but this offset is encoded in each .au file. Instead of assuming the minimum, read the actual value and store it for later use. ASTERISK-20984 #close Reported by: Roman S. Patches: asterisk-1.8.20.0-au-clicks-2.diff (license #6474) patch uploaded by Roman S. Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c	8 years ago
Joshua Colp	5e2a8ef94a	Merge "modules: change module LOAD_FAILUREs to LOAD_DECLINES" into 13	8 years ago
zuul	a22de46a06	Merge "strings.h: Avoid overflows in the string hash functions" into 13	8 years ago
Joshua Colp	ec051345f2	Merge "bridging: Ensure successful T.38 negotation" into 13	8 years ago
George Joseph	f882ca2572	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25	8 years ago
zuul	2e9f186c17	Merge "res_rtp_asterisk.c: Add stun_blacklist option" into 13	8 years ago
Joshua Colp	fdc63152c3	Merge changes from topics 'ASTERISK-26890', 'ASTERISK-26851' into 13 ... * changes: stun.c: Fix ast_stun_request() erratic timeout. sorcery.c: Speed up ast_sorcery_retrieve_by_id() res_pjsip: Fix pointer use after unref. res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member.	8 years ago
Richard Mudgett	cd80af508e	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	8 years ago
Richard Mudgett	f8219a2e12	stun.c: Fix ast_stun_request() erratic timeout. ... If ast_stun_request() receives packets other than a STUN response then we could conceivably never exit if we continue to receive packets with less than three seconds between them. * Fix poll timeout to keep track of the time when we sent the STUN request. We will now send a STUN request every three seconds regardless of how many other packets we receive while waiting for a response until we have completed three STUN request transmission cycles. Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266	8 years ago
Richard Mudgett	19b82a8644	sorcery.c: Speed up ast_sorcery_retrieve_by_id() ... Return early if ast_sorcery_retrieve_by_id() is not passed an id to find. Also eliminated the RAII_VAR() usage in the function. Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218	8 years ago
Richard Mudgett	aecf19e7d2	res_pjsip: Fix pointer use after unref. ... Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1	8 years ago
Richard Mudgett	304f652cda	res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member. ... * create_rtp(): Eliminate use of deprecated transport struct member. That member and several others in the transport structure were deprecated because of an infinite loop created when using realtime configuration. See 2451d4e455 ASTERISK-26851 Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc	8 years ago
Richard Mudgett	bb8cd2add7	tcptls.c: Cleanup TCP/TLS listener thread on abnormal exit. ... Temporarily running out of file descriptors should not terminate the listener thread. Otherwise, when there becomes more file descriptors available, nothing is listening. * Added EMFILE exception to abnormal thread exit. * Added an abnormal TCP/TLS listener exit error message. * Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not appear dead if something tries to connect to the socket. ASTERISK-26903 #close Change-Id: I10f2f784065136277f271159f0925927194581b5	8 years ago
Torrey Searle	d8967ff2c0	strings.h: Avoid overflows in the string hash functions ... On 2's compliment machines abs(INT_MIN) behavior is undefined and results in a negative value still being returnd. This results in negative hash codes that can result in crashes. ASTERISK-26528 #close Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b	8 years ago
Walter Doekes	bbbd262ec0	samples: Undo removal of include from canonicalize-app-names commit. ... This include was accidentally removed in changeset Ia79aea64de89531362e993e34230c2044a70aa93. My bad. Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082	8 years ago
zuul	e8514129a6	Merge "pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete()." into 13	8 years ago
zuul	b82f58d2bf	Merge "pjsip: Add Alembic for PUBLISH support." into 13	8 years ago
zuul	293f9d39b7	Merge "samples: Canonicalize app names in extensions.conf.sample." into 13	8 years ago
Joshua Colp	b3f4a6365e	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	8 years ago
Alexander Traud	e0e5a337fd	pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). ... When the Asterisk channel driver res_pjsip offers SIP-over-TLS, sometimes, not reproducible, Asterisk crashed in pj_ssl_sock_get_info() because a NULL pointer was read. This change avoids this crash. ASTERISK-26927 #close Change-Id: I24a6011b44d1426d159742ff4421cf806a52938b	8 years ago
Joshua Colp	09ec2dbbf3	Merge "chan_sip: Session Timers required but refused wrongly." into 13	8 years ago
Joshua Colp	0b00df55fc	Merge "pjproject_bundled: Add 3 upstream patches" into 13	8 years ago
Torrey Searle	3e7c396a51	bridging: Ensure successful T.38 negotation ... When a T.38 happens immediatly after call establishment, the control frame can be lost because the other leg is not yet in the bridge. This patch detects this case an makes sure T.38 negotation happens when the 2nd leg is being made compatible with the negotating first leg ASTERISK-26923 #close Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94	8 years ago
Joshua Colp	4b504544f6	Merge "sample_config: Add samples for pubsub to pjsip.conf.sample" into 13	8 years ago
Joshua Colp	1a35801994	Merge "Unused realtime MOH classes not purged on 'moh reload'" into 13	8 years ago
Joshua Colp	f80b7f7014	Merge "res_pjsip_session: Allow BYE to be sent on disconnected session." into 13	8 years ago
George Joseph	4e6e069491	pjproject_bundled: Add 3 upstream patches ... 0035-r5572-svn-backport-dialog-transaction-deadlock.patch 0036-r5573-svn-backport-ua-pjsua-transaction-deadlock.patch 0037-r5576-svn-backport-session-timer-crash.patch Also removed the progress bar from wget download to stdout. ASTERISK-26905 #close Reported-by: Ross Beer Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256	8 years ago
George Joseph	d2a33cdedc	sample_config: Add samples for pubsub to pjsip.conf.sample ... Added: * outbound-publish * resource_list * inbound-publication * asterisk-publication Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a	8 years ago
Walter Doekes	ab9d2fc86d	samples: Canonicalize app names in extensions.conf.sample. ... This takes care of warnings by ossobv/asterisklint. Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93	8 years ago
Richard Mudgett	6906765381	res_pjsip_sdp_rtp.c: Don't alter global addr variable. ... * create_rtp(): Fix unexpected alteration of global address_rtp if a transport is bound to an address. * create_rtp(): Fix use of uninitialized memory if the endpoint RTP media address is invalid or the transport has an invalid address. ASTERISK-26851 Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7	8 years ago
Corey Farrell	68bde0f07d	CDR: Protect from data overflow in ast_cdr_setuserfield. ... ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could result in a buffer overrun when called from chan_sip or func_cdr. This patch adds a maximum bytes written to the field by using ast_copy_string instead. ASTERISK-26897 #close patches: 0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted by Corey Farrell (license #5909) Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c	8 years ago
Daniel Journo	70e5a2655d	Unused realtime MOH classes not purged on 'moh reload' ... Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf hasn't changed. ASTERISK-25974 #close Change-Id: I42c78ea76528473a656f204595956c9eedcf3246	8 years ago
Richard Mudgett	27b556778d	res_pjsip: Fix transport ref leak. ... We were leaking a transport ref in multihomed_on_rx_message() which resulted in the FRACK about excessive ref counts. ASTERISK-26916 #close Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f	8 years ago
Alexander Traud	94bd529f9e	chan_sip: Session Timers required but refused wrongly. ... SIP user-agents indicate which protocol extensions are allowed in headers like Supported and Required. Such protocol extensions are Session Timers (RFC 4028) for example. Session Timers are supported since Mantis-10665. Since ASTERISK-21721, not only the first but multiple Supported/Required headers in a message are parsed. In that change, an existing variable was re-used within a newly added do-loop. Currently, at the end of that loop, that variable is an empty string always. Previously, that variable was used within log output. However, the log output was not changed. ASTERISK-26915 #close Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990	8 years ago
zuul	365db26535	Merge "Forward declare 'struct ast_json' in asterisk.h" into 13	8 years ago
Joshua Colp	bca9685d39	res_pjsip_session: Allow BYE to be sent on disconnected session. ... It is perfectly acceptable for a BYE to be sent on a disconnected session. This occurs when we respond to a challenge to the BYE for authentication credentials. ASTERISK-26363 Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045	8 years ago
George Joseph	01c1e60a4f	Merge "build: Fix deb build issues with fakeroot" into 13	8 years ago