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Author	SHA1	Message	Date
Jenkins2	3ee963585a	Merge "res_pjsip: Fix leak of persistent endpoint references." into 14	8 years ago
Richard Mudgett	d1b63e2323	cdr.c: Replace redundant check with an ast_assert() ... The only caller of cdr_object_fn_table.process_party_b() explicitly does the check before calling. Change-Id: Ib0c53cdf5048227842846e0df9d2c19117c45618	8 years ago
Richard Mudgett	e43959cda7	cdr.c: Replace inlined code with ao2_t_replace() ... Change-Id: I9f424f5282ca7d833592f958d95f1b2bafb549b0	8 years ago
Richard Mudgett	3c6b20bd34	cdr.c: Use current ao2 flag names ... Change-Id: Ib59d7d2f2a4a822754628f2c48a308d6791a6e6e	8 years ago
Richard Mudgett	f9f9a5412f	cdr.h: Fix doxygen comments. ... * Also some misc formatting in cdr.c. Change-Id: Ied89a28802a662c37c43326a1aafdce596e0df4a	8 years ago
Richard Mudgett	b145619594	res_pjsip_registrar.c: Update remove_existing AOR contact handling. ... When "rewrite_contact" is enabled, the "max_contacts" count option can block re-registrations because the source port from the endpoint can be random. When the re-registration is blocked, the endpoint may give up re-registering and require manual intervention. * The "remove_existing" option now allows a registration to succeed by displacing any existing contacts that now exceed the "max_contacts" count. Any removed contacts are the next to expire. The behaviour change is beneficial when "rewrite_contact" is enabled and "max_contacts" is greater than one. The removed contact is likely the old contact created by "rewrite_contact" that the device is refreshing. ASTERISK-27192 Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b	8 years ago
Corey Farrell	cf474bf57a	res_pjsip: Fix issues that prevented shutdown of modules. ... res_pjsip and res_pjsip_session had circular references, preventing both modules from shutting down. * Move session supplement registration to res_pjsip. * Use create internal functions for use by pjsip_message_filter.c. ASTERISK-27306 Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b	8 years ago
Sean Bright	4bdc3540fe	res_config_sqlite: Don't enable SQLite CDRs when running 'make samples' ... Change-Id: I65a5190b2732b2246d67472db70dd37db64ddad4	8 years ago
hajekd	5b4fc0699a	res/res_ari.c Fix: Memory leaks in ARI when using Content-Type: application/json ... ASTERISK-27305 Reported by: David Hajek Tested by: David Hajek Change-Id: Ife3e289062e6cf7d0e7d342dbf79ed96feff441e	8 years ago
Alexander Traud	20fd595890	tcptls: Do not re-bind to wildcard on client creation. ... Since ASTERISK-26922, this issue affected only those chan_sip which were * enabled for dual-stack (bindaddr=::), and * enabled for TCP (tcpenable=yes) and/or TLS (tlsenable=yes), and * tried to register and/or invite a IPv4-only service, * via TCP and/or TLS. Now, ast_tcptls_client_create does not re-bind to [::] anymore. ASTERISK-27324 #close Change-Id: I4b242837bdeb1ec7130dc82505c6180a946fd9b5	8 years ago
Corey Farrell	cacf938d38	res_pjsip: Fix leak of persistent endpoint references. ... Do not manually call sip_endpoint_apply_handler from load_all_endpoints. This is not necessary and causes memory leaks. Additionally reinitialize persistent->aors when we reuse a persistent object with a new endpoint. ASTERISK-27306 Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb	8 years ago
Jenkins2	09932a13ef	Merge "vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED." into 14	8 years ago
Jenkins2	b4ed8c4e4f	Merge "res_pjsip: Fix leak of fake_auth references." into 14	8 years ago
Corey Farrell	e71f6d4dfa	vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. ... Use temporary variable to prevent multiple evaluations of elem argument. This resolves a memory leak in res_pjproject startup. ASTERISK-27317 #close Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d	8 years ago
Jenkins2	acb53a2295	Merge "main/strings: Fix uninitialized value." into 14	8 years ago
Corey Farrell	3e2cb5a9e9	res_pjsip: Fix leak of fake_auth references. ... pjsip_distributor leaks references to fake_auth when the default realm has not changed. ASTERISK-27306 Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202	8 years ago
Corey Farrell	3421b1de27	main/strings: Fix uninitialized value. ... ast_strings_match uses sscanf and checks for non-zero return to verify a token was parsed. This is incorrect as sscanf returns EOF (-1) for errors. ASTERISK-27318 #close Change-Id: Ifcece92605f58116eff24c5a0a3b0ee08b3c87b1	8 years ago
Daniel Tryba	e8ed162941	res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy ... Currently privacy requests are only granted if the Privacy header value is exactly "id" (defined in RFC 3325). It ignores any other possible value (or a combination there of). This patch reverses the logic from testing for "id" to grant privacy, to testing for "none" and granting privacy for any other value. "none" must not be used in combination with any other value (RFC 3323 section 4.2). ASTERISK-27284 #close Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56	8 years ago
Joshua Colp	0991cb0a8f	Merge "app_queue.c: Fix announcements when announce-to-first-user not enabled." into 14	8 years ago
krells	44d443d87e	res_calendar_icalendar: Filter out occurrences superceded by another VEVENT ... When we are loading the calendars, we call libical's icalcomponent_foreach_recurrence method for each VEVENT component that we have in our calendar. That method has no knowledge concerning the existence of the other VEVENT components and will feed our callback with all ocurrences matching the requested time span. The occurrences generated by icalcomponent_foreach_recurrence while expanding a recurring VEVENT's RRULE and RDATE properties can be superceded by an other VEVENT sharing the same UID. I use an external iterator (in libical terminology) to avoid messing with the internal ones from the calling function, and search for VEVENTS which could supersede the current occurrence. The event which can invalidate this occurence needs to have: - the same UID as our recurrent component (comp) - a RECURRENCE-ID property, which represents the start time of this occurrence If one component is found, just clean and return. ASTERISK-27296 #close Reported by: Benoît Dereck-Tricot Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8	8 years ago
Jenkins2	752d55f719	Merge "heap.c: No need to calloc heap pointer array." into 14	8 years ago
Jenkins2	4fdc39bcfb	Merge "logger: Bring back ability to turn debug on by source file" into 14	8 years ago
Richard Mudgett	699b6a70d5	app_queue.c: Fix announcements when announce-to-first-user not enabled. ... The previous patch for ASTERISK-27216 made it so you wouldn't get any position or periodic announcements unless you had announce-to-first-user enabled. The announce-to-first-user feature was added by ASTERISK_21782 as a result of the patch which introduced the redundant announcements that ASTERISK-27216 removes. * By noting that the makeannouncement variable is used to suppresses the first user announcement, we set its initial value to the announce-to-first-user enable setting. ASTERISK-27216 Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a	8 years ago
Richard Mudgett	37935c79f6	heap.c: No need to calloc heap pointer array. ... Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68	8 years ago
Jenkins2	88ba803110	Merge "pjsip_message_filter: Fix regression causing bad contact address" into 14	8 years ago
George Joseph	00fc98d330	logger: Bring back ability to turn debug on by source file ... Somewhere along the way we lost the ability to debug individual source files. For modules, this wasn't a big deal but all the source files in ./main are in the one "core" module so debugging individual core capabilities was almost impossible. * Added a test to DEBUG_ATLEAST that also checks __FILE__ instead of just module name. Any source file will work even if it's in a module subdirectory. Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e	8 years ago
Joshua Colp	821927b077	Merge "pjproject: Patch to correct STUN FINGERPRINT usage" into 14	8 years ago
Jenkins2	77558d80f3	Merge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential." into 14	8 years ago
Richard Mudgett	de99052acb	res_pjsip_outbound_publish.c: Fix misplaced parenthesis. ... The pjsip_publishc_init() call was referenced with a misplaced parentheses. As a result, outbound publication messages went out with an expiration of 1 second. ASTERISK-27298 Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08	8 years ago
George Joseph	3c5c31f14c	pjsip_message_filter: Fix regression causing bad contact address ... The "res_pjsip: Filter out non SIP(S) requests" commit moved the filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER in order to filter out incoming bad uri schemes as early as possible. Since the change affected outgoing messages as well and the TRANSPORT layer is the last to be run on outgoing messages, we were overwriting the setting of external_signaling_address (which is set earlier by res_pjsip_nat) with an internal address. * pjsip_message_filter now registers itself as a pjproject module twice. Once in the TSX layer for the outgoing messages (as it was originally), then a second time in the TRANSPORT layer for the incoming messages to catch the invalid uri schemes. ASTERISK-27295 Reported by: Sean Bright Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c	8 years ago
Richard Mudgett	738da78786	res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential. ... The bridge_p2p_rtp_write() has potential reentrancy problems. * Accessing the bridged RTP members must be done with the instance1 lock held. The DTMF and asymmetric codec checks must be split to be done with the correct RTP instance struct locked. i.e., They must be done when working on the appropriate side of the point to point bridge. * Forcing the RTP mark bit was referencing the wrong side of the point to point bridge. The set mark bit is used everywhere else to set the mark bit when sending not receiving. The patches for ASTERISK_26745 and ASTERISK_27158 did not take into account that not everything carried by RTP uses a codec. The telephony DTMF events are not exchanged with a codec. As a result when RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is enabled, the DTMF digits would always get passed to the core even though the local native RTP bridge is active, and the DTMF digits would go out using the wrong SSRC id. * Add protection for non-format payload types like DTMF when updating the lastrxformat and lasttxformat. Also protect against non-format payload types when checking for asymmetric codecs. ASTERISK-27292 Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186	8 years ago
Jenkins2	9bd204f64a	Merge "channel.c: Fix invalid reference in conditionaled out code." into 14	8 years ago
Jenkins2	9fe0803c8b	Merge "app_queue: Only do announcement logic between ringing cycles" into 14	8 years ago
Sean Bright	95eae41b37	pjproject: Patch to correct STUN FINGERPRINT usage ... Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6	8 years ago
Richard Mudgett	787fd96709	channel.c: Fix invalid reference in conditionaled out code. ... ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c	8 years ago
George Joseph	31cbc1166c	build: A few gcc 7 error fixes ... Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec	8 years ago
StefanEng86	b197d851d8	app_queue: Only do announcement logic between ringing cycles ... This patch reverts the change by patch 2263 from old reviewboard. Note that reverting that 2263-patch still preserves the behaviour that the commit log of the 2263-patch claimed to add. The reason for this is: The function wait_for_answer is only called from try_calling which in turn is only called from the main for loop in queue_exec, and earlier in that loop we already check the things that's removed by this patch. There's no need to check those things twice each loop iteration, and I think the proper place to check it is before each ringing cycle. By checking it in wait_for_answer, you allow the issue explained in the jira - that the head caller hears announcements while the agents' sip phones are actively ringing. Reported-by: Stefan Engström Tested-by: Stefan Engström ASTERISK-27216 #close Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0	8 years ago
Sean Bright	875568c0f9	res_pjsip: Use ast_sip_is_content_type() where appropriate ... Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d	8 years ago
Joshua Colp	ddc216f12e	Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1" into 14	8 years ago
Jenkins2	040487b90e	Merge "res_srtp: lower log level of auth failures" into 14	8 years ago
Rodrigo Ramírez Norambuena	339676b77d	res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1 ... In PostgreSQL 9.1 the backslash are string literals and not the escape of characters. In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the support for old version of Postgresql than 9.1 was dropped. The sentence before make was "ESCAPE '\'" but in version before than 9.1 need it to be as follow "ESCAPE '\\'". ASTERISK-27283 Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949	8 years ago
Joshua Colp	8ee7ed020f	Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI" into 14	8 years ago
Jenkins2	e5f911519d	Merge "bridge : Fix one-way direct-media when early bridging with native_rtp" into 14	8 years ago
Jean Aunis	d884871ca7	bridge : Fix one-way direct-media when early bridging with native_rtp ... When two channels were early bridged in a native_rtp bridge, the RTP description on one side was not updated when the other side answered. This patch forbids non-answered channels to enter a native_rtp bridge, and triggers a bridge reconfiguration when an ANSWER frame is received. ASTERISK-27257 Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df	8 years ago
George Joseph	c97eb95193	res_pjsip_pubsub: Check for Content-Type header in rx_notify_request ... pubsub_on_rx_notify_request wasn't checking for a null Content-Type header before checking that it was application/simple-message-summary. ASTERISK-27279 Reported by: Ross Beer Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52	8 years ago
David J. Pryke	060cea2cca	chan_sip: Expose read-only access to the full SIP INVITE Request-URI ... Provide a way to get the contents of the the Request URI from the initial SIP INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}") ASTERISK-27278 Reported by: David J. Pryke Tested by: David J. Pryke Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e	8 years ago
Joshua Colp	3274c13b12	Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 14	8 years ago
Joshua Colp	864e76374c	Merge "res_calendar: On reload, update all configuration" into 14	8 years ago
Joshua Colp	05bb57c56f	Merge "cdr_mysql.c: Apply cdrzone to start and answer" into 14	8 years ago
Alexander Traud	393d813789	tcptls: Fixed a white space error. ... ASTERISK-26606 Change-Id: I81a7268ef7ba012d4d80d44c70b6276d48e397fa	8 years ago