summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2017-10-09cdr.h: Fix doxygen comments.Richard Mudgett
* Also some misc formatting in cdr.c. Change-Id: Ied89a28802a662c37c43326a1aafdce596e0df4a
2017-10-09res_pjsip_registrar.c: Update remove_existing AOR contact handling.Richard Mudgett
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
2017-10-09res_pjsip: Fix issues that prevented shutdown of modules.Corey Farrell
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
2017-10-09res_config_sqlite: Don't enable SQLite CDRs when running 'make samples'Sean Bright
Change-Id: I65a5190b2732b2246d67472db70dd37db64ddad4
2017-10-09res/res_ari.c Fix: Memory leaks in ARI when using Content-Type: application/jsonhajekd
ASTERISK-27305 Reported by: David Hajek Tested by: David Hajek Change-Id: Ife3e289062e6cf7d0e7d342dbf79ed96feff441e
2017-10-08tcptls: Do not re-bind to wildcard on client creation.Alexander Traud
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
2017-10-06res_pjsip: Fix leak of persistent endpoint references.Corey Farrell
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
2017-10-06Merge "vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED." into 14Jenkins2
2017-10-06Merge "res_pjsip: Fix leak of fake_auth references." into 14Jenkins2
2017-10-06vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.Corey Farrell
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
2017-10-06Merge "main/strings: Fix uninitialized value." into 14Jenkins2
2017-10-06res_pjsip: Fix leak of fake_auth references.Corey Farrell
pjsip_distributor leaks references to fake_auth when the default realm has not changed. ASTERISK-27306 Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202
2017-10-05main/strings: Fix uninitialized value.Corey Farrell
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
2017-10-05res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacyDaniel Tryba
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
2017-10-04Merge "app_queue.c: Fix announcements when announce-to-first-user not ↵Joshua Colp
enabled." into 14
2017-10-04res_calendar_icalendar: Filter out occurrences superceded by another VEVENTkrells
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
2017-10-03Merge "heap.c: No need to calloc heap pointer array." into 14Jenkins2
2017-10-03Merge "logger: Bring back ability to turn debug on by source file" into 14Jenkins2
2017-09-28app_queue.c: Fix announcements when announce-to-first-user not enabled.Richard Mudgett
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
2017-09-28heap.c: No need to calloc heap pointer array.Richard Mudgett
Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68
2017-09-28Merge "pjsip_message_filter: Fix regression causing bad contact address" into 14Jenkins2
2017-09-28logger: Bring back ability to turn debug on by source fileGeorge Joseph
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
2017-09-28Merge "pjproject: Patch to correct STUN FINGERPRINT usage" into 14Joshua Colp
2017-09-28Merge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential." ↵Jenkins2
into 14
2017-09-27res_pjsip_outbound_publish.c: Fix misplaced parenthesis.Richard Mudgett
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
2017-09-26pjsip_message_filter: Fix regression causing bad contact addressGeorge Joseph
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
2017-09-26res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential.Richard Mudgett
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
2017-09-26Merge "channel.c: Fix invalid reference in conditionaled out code." into 14Jenkins2
2017-09-26Merge "app_queue: Only do announcement logic between ringing cycles" into 14Jenkins2
2017-09-25pjproject: Patch to correct STUN FINGERPRINT usageSean Bright
Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6
2017-09-25channel.c: Fix invalid reference in conditionaled out code.Richard Mudgett
ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c
2017-09-25build: A few gcc 7 error fixesGeorge Joseph
Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec
2017-09-25app_queue: Only do announcement logic between ringing cyclesStefanEng86
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
2017-09-22res_pjsip: Use ast_sip_is_content_type() where appropriateSean Bright
Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d
2017-09-22Merge "res_config_pgsql: Fix removed support to previous for versions ↵Joshua Colp
PostgreSQL 9.1" into 14
2017-09-21Merge "res_srtp: lower log level of auth failures" into 14Jenkins2
2017-09-21res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1Rodrigo Ramírez Norambuena
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
2017-09-21Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI" ↵Joshua Colp
into 14
2017-09-21Merge "bridge : Fix one-way direct-media when early bridging with ↵Jenkins2
native_rtp" into 14
2017-09-20bridge : Fix one-way direct-media when early bridging with native_rtpJean Aunis
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
2017-09-19res_pjsip_pubsub: Check for Content-Type header in rx_notify_requestGeorge Joseph
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
2017-09-19chan_sip: Expose read-only access to the full SIP INVITE Request-URIDavid J. Pryke
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
2017-09-19Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 14Joshua Colp
2017-09-19Merge "res_calendar: On reload, update all configuration" into 14Joshua Colp
2017-09-19Merge "cdr_mysql.c: Apply cdrzone to start and answer" into 14Joshua Colp
2017-09-18tcptls: Fixed a white space error.Alexander Traud
ASTERISK-26606 Change-Id: I81a7268ef7ba012d4d80d44c70b6276d48e397fa
2017-09-18res_srtp: lower log level of auth failuresAlexander Traud
Previously, sRTP authentication failures were reported on log level WARNING. When such failures happen, each RT(C)P packet is affected, spamming the log. Now, those failures are reported at log level VERBOSE 2. Furthermore, the amount is further reduced (previously all two seconds, now all three seconds). Additionally, the new log entry informs whether media (RTP) or statistics (RTCP) are affected. ASTERISK-16898 #close Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0
2017-09-15AST-2017-008: Improve RTP and RTCP packet processing.Richard Mudgett
Validate RTCP packets before processing them. * Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. * Fixed potentially reading garbage beyond the received RTCP record data. * Fixed rtp->themssrc only being set once when the remote could change the SSRC. We would effectively stop handling the RTCP statistic records. * Fixed rtp->themssrc to not treat a zero value as special by adding rtp->themssrc_valid to indicate if rtp->themssrc is available. ASTERISK-27274 Make strict RTP learning more flexible. Direct media can cause strict RTP to attempt to learn a remote address again before it has had a chance to learn the remote address the first time. Because of the rapid relearn requests, strict RTP could latch onto the first remote address and fail to latch onto the direct media remote address. As a result, you have one way audio until the call is placed on and off hold. The new algorithm learns remote addresses for a set time (1.5 seconds) before locking the remote address. In addition, we must see a configured number of remote packets from the same address in a row before switching. * Fixed strict RTP learning from always accepting the first new address packet as the new stream. * Fixed strict RTP to initialize the expected sequence number with the last received sequence number instead of the last transmitted sequence number. * Fixed the predicted next sequence number calculation in rtp_learning_rtp_seq_update() to handle overflow. ASTERISK-27252 Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
2017-09-15Merge "res_pjsip: Filter out non SIP(S) requests" into 14Jenkins2
2017-09-15res_calendar: On reload, update all configurationSean Bright
This changes the behavior of res_calendar to drop all existing calendars and re-create them whenever a reload is done. The Calendar API provides no way for configuration information to be pushed down to calendar 'techs' so updated settings would not take affect until a module unload/load was done or Asterisk was restarted. Asterisk 15+ already has a configuration option 'fetch_again_at_reload' that performs a similar function. Also fix a tiny memory leak in res_calendar_caldav while we're at it. ASTERISK-25524 #close Reported by: Jesper Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-19 15:54:09 +0000