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Author	SHA1	Message	Date
Jenkins2	6aa11c4b31	Merge "AST-2017-003: Handle zero-length body parts correctly." into 13	8 years ago
Mark Michelson	919ccdb9ac	AST-2017-002: Ensure transaction key buffer is large enough. ... ASTERISK-26938 #close Change-Id: I266490792fd8896a23be7cb92f316b7e69356413	8 years ago
Mark Michelson	49c032abef	AST-2017-003: Handle zero-length body parts correctly. ... ASTERISK-26939 #close Change-Id: I7ea235ab39833a187db4e078f0788bd0af0a24fd	8 years ago
George Joseph	1cc18d4025	AST-2017-004: chan_skinny: Add EOF check in skinny_session ... The while(1) loop in skinny_session wasn't checking for EOF so a packet that was longer than a header but still truncated would spin the while loop infinitely. Not only does this permanently tie up a thread and drive a core to 100% utilization, the call of ast_log() in such a tight loop eats all available process memory. Added poll with timeout to top of read loop ASTERISK-26940 #close Reported-by: Sandro Gauci Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898	8 years ago
Sean Bright	c107ab4c04	res_hep_rtcp: Add support level to module info ... Change-Id: I5661478f9cf12d431f730e42be79323b62831e92	8 years ago
Ivan Poddubny	cfeae52c0f	app_queue: Fix duplicate queue_log entries for EXITEMPTY and ABANDON ... There are 2 places in app_queue.c that log EXITEMPTY event: one in wait_our_turn, and another one in queue_exec in the loop trying to call an agent after wait_our_turn. In most cases it leads to logging EXITEMPTY twice. ABANDON is also logged on two places, and in the rare case when an agent and caller hang up simultaneously it's also possible to get duplicates in queue_log. This commit changes wait_our_turn to return -1 ("the caller should exit the queue") instead of 0 ("the caller's turn has arrived") in case of leaving when empty, so queue_exec skips the agent calling loop. Also, leave_queue is now executed only once in this case, because 2nd time is just a noop when the queue entry has already been removed. Also, it sets qe->handled to -1 to indicate that the call was not answered by an agent, but the necessary handling has already been done in order to avoid logging an extra ABANDON entry. ASTERISK-25665 #close Reported by: Ove Aursand Change-Id: I4578dd383bf2ac41589cf167865e8aaebcd4c11e	8 years ago
Jenkins2	722ec0671e	Merge "res_pjsip_session.c: Process initial INVITE sooner. (key exists)" into 13	8 years ago
Rodrigo Ramírez Norambuena	5da91c65be	Fix spelling queues.conf.sample file ... Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee	8 years ago
Joshua Colp	1618203964	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	8 years ago
Richard Mudgett	b67363006f	res_pjsip_session.c: Process initial INVITE sooner. (key exists) ... Retransmissions of an initial INVITE could be queued in the serializer before we have processed the first INVITE message. If the first INVITE message doesn't get completely processed before the retransmissions are seen then we could try to setup the same call from the retransmissions. A symptom of this is seeing a (key exists) message associated with an INVITE. An earlier change attempted to address this kind of problem by calculating a distributor serializer to use for unassociated messages. Part of that change also made incoming calls keep using that distributor serializer. (ASTERISK-26088) However, some leftover code was still deferring the INVITE processing to the session's serializer even though we were already in that serializer. This not only is unnecessary but would cause the same call resetup problem. * Removed the code to defer processing the initial INVITE to the session's serializer because we are already running in that serializer. ASTERISK-26998 #close Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6	8 years ago
Jenkins2	6383d9214a	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages." into 13	8 years ago
Jenkins2	3cfbb8b481	Merge "logger: Added logger_queue_limit to the configuration options." into 13	8 years ago
Jenkins2	ddbc68b68a	Merge "tcptls: Improve error messages for TLS connections." into 13	8 years ago
Alexei Gradinari	6af2dd34af	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	8 years ago
Jenkins2	a546e16cdb	Merge "Prevent Undefined Capath Crash" into 13	8 years ago
Jenkins2	a2c0d8c25d	Merge "cel_odbc: Fix timestamp processing for microseconds" into 13	8 years ago
Joshua Colp	6fba0a41f0	tcptls: Improve error messages for TLS connections. ... This change uses the functions provided by OpenSSL to query and better construct error messages for situations where the connection encounters a problem. ASTERISK-26606 Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b	8 years ago
Joshua Elson	8ec6e19c86	Prevent Undefined Capath Crash ... It is possible to initialize a valid config without a capath or cafile definition. This will cause a crash on a reload. This fix ensures capath is always allocated. ASTERISK-26983 #close Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12	8 years ago
George Joseph	d6325373ac	cel_odbc: Fix timestamp processing for microseconds ... When a column is of type timestamp, the fraction part of the event field's seconds was frequently parsed incorrectly especially if there were leading zeros. For instance "2017-05-23 23:55:03.023" would be parsed into an int as "23" then when the timestamp was formatted again to be inserted into the database column it'd be "2017-05-23 23:55:03.23" which is now 230 milliseconds instead of 23 milliseconds. "03.000001" would be transformed to "03.1", etc. * If the event field is 'eventtime' and the db column is timestamp, then existing processing has already correctly formatted the timestamp so now we simply use it rather than parsing it and re-printing it. This is the most common use case anyway. * If the event field is other than 'eventtime' and the db column is timestamp, we now parse the seconds, including the fractional part into a double rather than 2 ints. This preserves the magnitude and precision of the fractional part. When we print it, we now print it as a "%09.6lf" which correctly represents the input. To be honest, why we parse the string timestamp into components, test the components, then print the components back into a string timestamp is beyond me. We should use parse it, test it, then if it passes, use the original string representation in the database call. Maybe someone thought that some implementations wouldn't take a partial timestamp string like "2017-05-06" and decided to always produce a full timestamp string even if an abbreviated one was supplied. Anyway, I'm leaving it as it is. ASTERISK-25032 #close Reported-by: Etienne Lessard Change-Id: Id407e6221f79a5c1120e1a70bc7e893bbcaf1938	8 years ago
Joshua Colp	10a49ab362	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	8 years ago
Joshua Colp	371213217c	Merge "func_cdr: Allow empty value for CDR dialplan function." into 13	8 years ago
George Joseph	7d4a22bf2e	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	8 years ago
Vitezslav Novy	1bcce442d0	chan_sip: Change sip_get_codec() to return correct codec list ... Return cahnnel nativeformats to fix bridge technology selection process. Same approach as in pjsip module. ASTERISK-26143 Reported-by: Henning Holtschneider Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48	8 years ago
Joshua Colp	e3fa458440	Merge "netsock2.c: Made get/set addr port avoid potential uninitialized memory." into 13	8 years ago
Joshua Colp	4d0bc3e5fc	Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI." into 13	8 years ago
Richard Mudgett	614eda785d	netsock2.c: Made get/set addr port avoid potential uninitialized memory. ... Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647	8 years ago
Jenkins2	250fee82aa	Merge "app_confbridge: Fix reference to cfg in menu_template_handler" into 13	8 years ago
Joshua Colp	c3ed63cb2c	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	8 years ago
George Joseph	bed6c0d04b	app_confbridge: Fix reference to cfg in menu_template_handler ... menu_template_handler wasn't properly accounting for the fact that it might be called both during a load/reload (which isn't really valid but not prevented) and by a dialplan function. In both cases it was attempting to use the "pending" config which wasn't valid in the latter case. aco_process_config is also partly to blame because it wasn't properly cleaning "pending" up when a reload was done and no changes were made. Both of these contributed to a crash if CONFBRIDGE(menu,template) was called in a dialplan after a reload. * aco_process_config now sets info->internal->pending to NULL after it unrefs it although this isn't strictly necessary in the context of this fix. * menu_template_handler now uses the "current" config and silently ignores any attempt to be called as a result of someone uses the "template" parameter in the conf file. Luckily there's no other place in the codebase where aco_pending_config is used outside of aco_process_config. ASTERISK-25506 #close Reported-by: Frederic LE FOLL Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7	8 years ago
Joshua Colp	ce1985b099	Merge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures" into 13	8 years ago
Joshua Colp	7ffd80cc04	bridge: Fix returning to dialplan when executing Bridge() from AMI. ... When using the Bridge AMI action on the same channel multiple times it was possible for the channel to return to the wrong location in the dialplan if the other party hung up. This happened because the priority of the channel was not preserved across each action invocation and it would fail to move on to the next priority in other cases. This change makes it so that the priority of a channel is preserved when taking control of it from another thread and it is incremented as appropriate such that the priority reflects where the channel should next be executed in the dialplan, not where it may or may not currently be. The Bridge AMI action was also changed to ensure that it too starts the channels at the next location in the dialplan. ASTERISK-24529 Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a	8 years ago
Kevin Harwell	bbe90d6aed	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	8 years ago
Jenkins2	1d44c838d8	Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections" into 13	8 years ago
Sean Bright	526a0081a0	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	8 years ago
Jenkins2	08dc126b38	Merge "rtp_engine.c: Fix deadlock potential copying RTP payload maps." into 13	8 years ago
Jenkins2	20cc154dd3	Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook." into 13	8 years ago
Thierry Magnien	23db04ed93	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	8 years ago
Richard Mudgett	02234e920c	rtp_engine.c: Fix deadlock potential copying RTP payload maps. ... There is a theoretical potential to deadlock in ast_rtp_codecs_payloads_copy() because it locks two different ast_rtp_codecs locks. It is theoretical because the callers of the function are either copying between a local ast_rtp_codecs struct and a RTP instance of the ast_rtp_codecs struct. Or they are copying between the caller and callee channel RTP instances before initiating the call to the callee. Neither of these situations could actually result in a deadlock because there cannot be another thread involved at the time. * Add deadlock avoidance code to ast_rtp_codecs_payloads_copy() since it locks two ast_rtp_codecs locks to perform a copy. This only affects v13 since this deadlock avoidance code is already in newer branches. Change-Id: I1aa0b168f94049bd59bbd74a85bd1e78718f09e5	8 years ago
Richard Mudgett	9d5df48968	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	8 years ago
George Joseph	623832b94e	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	8 years ago
Jenkins2	bf7cf10d15	Merge "frame: Better handle interpolated frames." into 13	8 years ago
Jenkins2	9bb683242c	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate" into 13	8 years ago
Jenkins2	dc7166e59f	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string" into 13	8 years ago
Jenkins2	a439faec57	Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK" into 13	8 years ago
Jenkins2	e3df529db5	Merge "vector: defaults and indexes" into 13	8 years ago
Joshua Colp	5e3e309c82	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation." into 13	8 years ago
Jenkins2	b49e41390d	Merge "cleanup: Fix fread() and fwrite() error handling" into 13	8 years ago
Jenkins2	cf19589001	Merge "pjproject_bundled: Add --disable-libwebrtc to configure" into 13	8 years ago
George Joseph	c5b9ed20fd	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	8 years ago
Kevin Harwell	c853cfdc7c	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	8 years ago