asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	fbfce63b30	func_cdr: Allow empty value for CDR dialplan function. A regression was introduced in 12 where passing an empty value to the CDR dialplan function was not longer allowed. This change returns to the behavior of 11 where it is permitted. ASTERISK-26173 Change-Id: I3f148203b54ec088007e29e30005a5de122e51c5	9 years ago
Kevin Harwell	3a5683d66c	res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures When a call gets put on hold RTP is temporarily stopped and Asterisk was setting the remote RTCP address to NULL. Then when RTCP data was received from the remote endpoint, Asterisk would be missing this information when publishing the rtcp_message stasis event. Consequently, message subscribers (in this case res_hep_rtcp) trying to parse the "from" field output the following error: "ast_sockaddr_split_hostport: Port missing in (null)" This patch makes it so the remote RTCP address is no longer set to NULL when stopping RTP. There was only one place that appeared to check if the remote RTCP address was NULL as a way to tell if RTCP was running. This patch added an additional check on the RTCP schedid for that case to make sure RTCP was truly not running. ASTERISK-26860 #close Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b	9 years ago
Jenkins2	24e5ae7e0a	Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections" into 14	9 years ago
Sean Bright	7f5ce428ed	cleanup: Change severity of fread short-read warning Many sound files don't have a full frame's worth of data at EOF, so the warning messages were a bit too noisy. So we demote them to debug messages. Change-Id: I6b617467d687658adca39170a81797a11cc766f6	9 years ago
Jenkins2	956f462431	Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook." into 14	9 years ago
Thierry Magnien	5cee143d9c	channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections For outgoing TCP connections, Asterisk uses the first IP address of the interface instead of the IP address we asked him to bind to. ASTERISK-26922 #close Reported-by: Ksenia Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb	9 years ago
Richard Mudgett	d54ae43cc8	res_pjsip_t38.c: Fix deadlock in T.38 framehook. A deadlock can happen between a channel lock and a pjsip session media container lock. One thread is processing a reINVITE's SDP and walking through the session's media container when it waits for the channel lock to put the determined format capabilities onto the channel. The other thread is writing a frame to the channel and processing the T.38 frame hook. The T.38 frame hook then waits for the pjsip session's media container lock. The two threads are now deadlocked. * Made the T.38 frame hook release the channel lock before searching the session's media container. This fix has been done to several other frame hooks to fix deadlocks. ASTERISK-26974 #close Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186	9 years ago
George Joseph	d9549bc475	res_pjsip_outbound_authenticator_digest: Add context to log messages There was no context info in this module's log messages so it was impossible to toubleshoot. Added endpoint or host to all messages and added the realms in the challenge for the "No auth credentials for any realm" message. Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b	9 years ago
Jenkins2	a5e79f71ee	Merge "frame: Better handle interpolated frames." into 14	9 years ago
Jenkins2	d2a135f121	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate" into 14	9 years ago
Jenkins2	41c0ff26e1	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string" into 14	9 years ago
Jenkins2	416319852d	Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK" into 14	9 years ago
Jenkins2	3d82216b32	Merge "vector: defaults and indexes" into 14	9 years ago
Joshua Colp	efe84aede9	Merge "cleanup: Fix fread() and fwrite() error handling" into 14	9 years ago
Jenkins2	35c15c589d	Merge "pjproject_bundled: Add --disable-libwebrtc to configure" into 14	9 years ago
George Joseph	cd4824ca69	res_pjsip_session: Add cleanup to ast_sip_session_terminate If you use ast_request to create a PJSIP channel but then hang it up without causing a transaction to be sent, the session will never be destroyed. This is due ot the fact that it's pjproject that triggers the session cleanup when the transaction ends. app_chanisavail was doing this to get more granular channel state and it's also possible for this to happen via ARI. * ast_sip_session_terminate was modified to explicitly call the cleanup tasks and unreference session if the invite state is NULL AND invite_tsx is NULL (meaning we never sent a transaction). * chan_pjsip/hangup was modified to bump session before it calls ast_sip_session_terminate to insure that session stays valid while it does its own cleanup. * Added test events to session_destructor for a future testsuite test. ASTERISK-26908 #close Reported-by: Richard Mudgett Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9	9 years ago
Jenkins2	31ede0f9d3	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation." into 14	9 years ago
Kevin Harwell	6722ec6a6a	res_pjsip/res_pjsip_callerid: NULL check on caller id name string It's possible for a name in a party id structure to be marked as valid, but the name string itself be NULL (for instance this is possible to do by using the dialplan CALLERID function). There were a couple of places where the name was validated, but the string itself was not checked before passing it to functions like 'strlen'. This of course caused a crashed. This patch adds in a NULL check before attempting to pass it into a function that is not NULL tolerant. ASTERISK-25823 #close Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a	9 years ago
Kevin Harwell	00001443dc	vector: defaults and indexes Added an pre-defined integer vector declaration. This makes integer vectors easier to declare and pass around. Also, added the ability to default a vector up to a given size with a default value. Lastly, added functionality that returns the "nth" index of a matching value. Also, updated a unit test to test these changes. Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5	9 years ago
Joshua Colp	4cdf937a2e	frame: Better handle interpolated frames. Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b	9 years ago
Jean Aunis	87a24362a2	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK Some equipments may send a re-INVITE containing an SDP in the final ACK request. If this happens in the context of direct media, the remote end should be updated with a re-INVITE. This patch queues an "update RTP peer" frame to trigger the re-INVITE, instead of the "source change" frame wich was used previously. ASTERISK-26951 Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6	9 years ago
George Joseph	6b15c7fc05	pjproject_bundled: Add --disable-libwebrtc to configure Without the disable, pjproject tries to build it's internal webrtc implementation which requires sse2. This fails on platforms without sse2. ASTERISK-26930 #close Reported-by: abelbeck Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410	9 years ago
Jenkins2	7b7d0571f7	Merge "alembic: Add table for 'resource_list' PJSIP RLS type." into 14	9 years ago
Jenkins2	3e791349af	Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions." into 14	9 years ago
George Joseph	89d10f40b7	Merge "res_hep: Add additional config initialization and validation" into 14	9 years ago
Sean Bright	4cb1458245	cleanup: Fix fread() and fwrite() error handling Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in the format modules. Neither of these functions will ever return a value less than 0, which we were checking for in some cases. I've introduced a fair amount of duplication in the format modules, but I plan to change how format modules work internally in a subsequent patch set, so this is simply a stop-gap. Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872	9 years ago
George Joseph	64f759c3e8	Merge "res_pjsip_session.c: Restructure ast_sip_session_alloc()" into 14	9 years ago
Joshua Colp	1174b566ab	alembic: Add table for 'resource_list' PJSIP RLS type. This change adds an Alembic migration which adds a ps_resource_list table that can contain resource_list RLS configuration objects. ASTERISK-26929 Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05	9 years ago
Sean Bright	03fb67f027	res_hep: Add additional config initialization and validation * Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor does not close fd 0 * Add logging output when the required option - capture_address - is not specified. * Remove a no longer relevant #define and correct related documentation * Pass appropriate flags to aco_option_register so that capture_address cannot be the empty string. ASTERISK-26953 #close Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076	9 years ago
Sean Bright	d757a70156	core: Use eventfd for alert pipes on Linux when possible The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	9 years ago
George Joseph	a40f2c8246	Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified" into 14	9 years ago
Richard Mudgett	85bdf33808	res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions. If ICE is enabled and a STUN server does not respond then we will block until we give up on the STUN response. This will take nine seconds. In the mean time the peer that sent the INVITE will send retransmissions. * Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out earlier to prevent these retransmissions. ASTERISK-26890 Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8	9 years ago
Richard Mudgett	47bc32173f	res_pjsip_session.c: Restructure ast_sip_session_alloc() * Restructure ast_sip_session_alloc() to need less cleanup on off nominal error paths. * Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid unnecessary ref manipulation to return a session. This is faster than calling a function. That function may do logging of the ref changes with REF_DEBUG enabled. Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a	9 years ago
George Joseph	f9b6e9e439	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes." into 14	9 years ago
George Joseph	d072e3211d	Merge "build: Update config.guess and config.sub" into 14	9 years ago
George Joseph	757491d315	Merge "make ari-stubs so doc periodic jobs can run" into 14	9 years ago
Sean Bright	6bbccb0dfd	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	9 years ago
George Joseph	5557efe507	Merge "res_stun_monitor: Don't fail to load if DNS resolution fails" into 14	9 years ago
Richard Mudgett	f5af1a1f74	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	9 years ago
Richard Mudgett	3adb547069	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	9 years ago
Sean Bright	b0b309d023	build: Update config.guess and config.sub Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172	9 years ago
Joshua Colp	7e507155b5	Merge "format_wav: Read 16khz wav samples properly" into 14	9 years ago
Joshua Colp	384d33f799	Merge "Revert "bridging: Ensure successful T.38 negotation"" into 14	9 years ago
George Joseph	04a30e28a3	Merge "format_pcm: Track actual header size of .au files" into 14	9 years ago
Sean Bright	e24bbf318c	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	9 years ago
George Joseph	26780e6da4	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: I5b47094b3480ed6a952efe7d45a77111d89b6eca	9 years ago
Sean Bright	07e6f5806a	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	9 years ago
Richard Mudgett	9464bc3f08	Revert "bridging: Ensure successful T.38 negotation" This reverts commit `443058f154`. Change-Id: Ifae215f971d2b235a2c2d61dd74c5dd8abad57ba	9 years ago
Sean Bright	22c07f5223	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	9 years ago
Sean Bright	ba30c8b692	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	9 years ago

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